chan_sip.c File Reference

Implementation of Session Initiation Protocol. More...

#include "asterisk.h"
#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <inttypes.h>
#include "asterisk/network.h"
#include "asterisk/paths.h"
#include "asterisk/lock.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock2.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/event.h"
#include "asterisk/cel.h"
#include "asterisk/data.h"
#include "asterisk/aoc.h"
#include "asterisk/message.h"
#include "sip/include/sip.h"
#include "sip/include/globals.h"
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
#include "sip/include/srtp.h"
#include "sip/include/sdp_crypto.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"
#include "sip/include/security_events.h"
#include "asterisk/sip_api.h"

Go to the source code of this file.

Data Structures
struct	ast_register_list
	The register list: Other SIP proxies we register with and receive calls from. More...
struct	ast_subscription_mwi_list
	The MWI subscription list. More...
struct	cfalias
	Structure for conversion between compressed SIP and "normal" SIP headers. More...
struct	cfsip_methods
	The core structure to setup dialogs. We parse incoming messages by using structure and then route the messages according to the type. More...
struct	cfsubscription_types
	Subscription types that we support. We support. More...
struct	domain_list
struct	epa_static_data_list
struct	event_state_compositor
	The Event State Compositors. More...
struct	invstate2stringtable
	Readable descriptions of device states. More...
struct	match_req_args
struct	show_peers_context
	Used in the sip_show_peers functions to pass parameters. More...
struct	sip_history_head
struct	sip_reasons
	Diversion header reasons. More...
struct	state_notify_data
Defines
#define	SIP_PEDANTIC_DECODE(str)
Variables
static int	authlimit = DEFAULT_AUTHLIMIT
static int	authtimeout = DEFAULT_AUTHTIMEOUT
static int	can_parse_xml
static unsigned int	chan_idx
static const char	config [] = "sip.conf"
static int	default_expiry = DEFAULT_DEFAULT_EXPIRY
static struct ast_jb_conf	default_jbconf
	Global jitterbuffer configuration - by default, jb is disabled.
static unsigned int	dumphistory
static int	global_authfailureevents
static unsigned int	global_autoframing
static int	global_callcounter
static unsigned int	global_cos_audio
static unsigned int	global_cos_sip
static unsigned int	global_cos_text
static unsigned int	global_cos_video
static int	global_dynamic_exclude_static = 0
static struct ast_jb_conf	global_jbconf
static int	global_match_auth_username
static int	global_max_se
static int	global_min_se
static int	global_prematuremediafilter
static int	global_qualify_gap
static int	global_qualify_peers
static int	global_qualifyfreq
static unsigned char	global_refer_addheaders
static int	global_reg_retry_403
static int	global_reg_timeout
static int	global_regattempts_max
static int	global_relaxdtmf
static int	global_rtpholdtimeout
static int	global_rtpkeepalive
static int	global_rtptimeout
static char	global_sdpowner [AST_MAX_EXTENSION]
static char	global_sdpsession [AST_MAX_EXTENSION]
static int	global_shrinkcallerid
static enum st_mode	global_st_mode
static enum st_refresher_param	global_st_refresher
static int	global_store_sip_cause
static int	global_t1
static int	global_t1min
static int	global_timer_b
static unsigned int	global_tos_audio
static unsigned int	global_tos_sip
static unsigned int	global_tos_text
static unsigned int	global_tos_video
static char	global_useragent [AST_MAX_EXTENSION]
static struct invstate2stringtable	invitestate2string []
static int	max_expiry = DEFAULT_MAX_EXPIRY
static int	max_subexpiry = DEFAULT_MAX_EXPIRY
static int	min_expiry = DEFAULT_MIN_EXPIRY
static int	min_subexpiry = DEFAULT_MIN_EXPIRY
static int	mwi_expiry = DEFAULT_MWI_EXPIRY
static const char	notify_config [] = "sip_notify.conf"
static unsigned int	recordhistory
static struct sip_settings	sip_cfg
static struct cfsip_methods	sip_methods []
static struct sip_reasons	sip_reason_table []
static struct cfsubscription_types	subscription_types []
static int	unauth_sessions = 0
DefaultSettings
Default setttings are used as a channel setting and as a default when configuring devices
static char	default_language [MAX_LANGUAGE]
static char	default_callerid [AST_MAX_EXTENSION]
static char	default_mwi_from [80]
static char	default_fromdomain [AST_MAX_EXTENSION]
static int	default_fromdomainport
static char	default_notifymime [AST_MAX_EXTENSION]
static char	default_vmexten [AST_MAX_EXTENSION]
static int	default_qualify
static int	default_keepalive
static char	default_mohinterpret [MAX_MUSICCLASS]
static char	default_mohsuggest [MAX_MUSICCLASS]
static char	default_parkinglot [AST_MAX_CONTEXT]
static char	default_engine [256]
static int	default_maxcallbitrate
static struct ast_codec_pref	default_prefs
static char	default_zone [MAX_TONEZONE_COUNTRY]
static unsigned int	default_transports
static unsigned int	default_primary_transport
Object counters @{
Bug: These counters are not handled in a thread-safe way ast_atomic_fetchadd_int() should be used to modify these values.
#define	sip_pvt_lock(x)   ao2_lock(x)
#define	sip_pvt_trylock(x)   ao2_trylock(x)
#define	sip_pvt_unlock(x)   ao2_unlock(x)
#define	BOGUS_PEER_MD5SECRET   "intentionally_invalid_md5_string"
	We can recognise the bogus peer by this invalid MD5 hash.
#define	SIP_TRANSPORT_STR_BUFSIZE   128
	Size of the SIP transport buffer.
#define	UNLINK(element, head, prev)
#define	append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)
	Append to SIP dialog history.
#define	check_request_transport(peer, tmpl)
	generic function for determining if a correct transport is being used to contact a peer
#define	CONTAINER_UNLINK(container, obj, tag)
	Unlink the given object from the container and return TRUE if it was in the container.
#define	FORMAT   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-25.25s %-15.15s %-15.15s \n"
#define	FORMAT2   "%-47.47s %9.9s %6.6s\n"
#define	FORMAT   "%-47.47s %-9.9s %-6.6s\n"
#define	FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"
#define	PEERS_FORMAT2   "%-25.25s %-39.39s %-3.3s %-10.10s %-10.10s %-3.3s %-8s %-11s %-32.32s %s\n"
#define	FORMAT   "%-40.40s %-20.20s %-16.16s\n"
#define	FORMAT2   "%-39.39s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"
#define	FORMAT   "%-39.39s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"
#define	FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"
#define	FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"
#define	FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"
#define	FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"
#define	FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"
#define	FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-10.10s %-10.10s\n"
#define	FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"
#define	DATA_EXPORT_SIP_PEER(MEMBER)
enum	message_integrity { MESSAGE_INVALID, MESSAGE_FRAGMENT, MESSAGE_FRAGMENT_COMPLETE, MESSAGE_COMPLETE }
	Indication of a TCP message's integrity. More...
enum	peer_unlink_flag_t { SIP_PEERS_MARKED, SIP_PEERS_ALL }
enum	match_req_res { SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, SIP_REQ_LOOP_DETECTED, SIP_REQ_FORKED }
static int	speerobjs = 0
static int	rpeerobjs = 0
static int	apeerobjs = 0
static int	regobjs = 0
static struct ast_flags	global_flags [3] = {{0}}
static int	global_t38_maxdatagram
static struct ast_event_sub *	network_change_event_subscription
static struct ast_event_sub *	acl_change_event_subscription
static int	network_change_event_sched_id = -1
static char	used_context [AST_MAX_CONTEXT]
static ast_mutex_t	netlock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 }
static ast_mutex_t	monlock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 }
	Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.
static ast_mutex_t	sip_reload_lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 }
static pthread_t	monitor_thread = AST_PTHREADT_NULL
	This is the thread for the monitor which checks for input on the channels which are not currently in use.
static int	sip_reloading = FALSE
static enum channelreloadreason	sip_reloadreason
struct ast_sched_context *	sched
static struct io_context *	io
static int *	sipsock_read_id
static struct domain_list	domain_list
static enum sip_debug_e	sipdebug
static int	sipdebug_text
	extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.
static struct _map_x_s	referstatusstrings []
static const int	HASH_PEER_SIZE = 563
static const int	HASH_DIALOG_SIZE = 563
struct {
enum ast_cc_service_type   service
const char *   service_string
}	sip_cc_service_map []
struct {
enum sip_cc_notify_state   state
const char *   state_string
}	sip_cc_notify_state_map []
struct epa_static_data_list	epa_static_data_list
static struct epa_static_data	cc_epa_static_data
static int	esc_etag_counter
static const int	DEFAULT_PUBLISH_EXPIRES = 3600
static struct sip_esc_publish_callbacks	cc_esc_publish_callbacks
static struct event_state_compositor	event_state_compositors []
static const int	ESC_MAX_BUCKETS = 37
struct ao2_container *	dialogs_needdestroy
struct ao2_container *	dialogs_rtpcheck
static struct ao2_container *	dialogs
static struct ao2_container *	threadt
	The table of TCP threads.
static struct ao2_container *	peers
	The peer list: Users, Peers and Friends.
static struct ao2_container *	peers_by_ip
static struct sip_peer *	bogus_peer
	A bogus peer, to be used when authentication should fail.
static struct ast_register_list	regl
static struct ast_subscription_mwi_list	submwil
static struct ast_threadstorage	ts_temp_pvt = { .once = PTHREAD_ONCE_INIT , .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , }
static struct ast_threadstorage	sip_transport_str_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_sip_transport_str_buf , .custom_init = NULL , }
static struct sip_auth_container *	authl = NULL
	Authentication container for realm authentication.
static ast_mutex_t	authl_lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 }
	Global authentication container protection while adjusting the references.
static int	sipsock = -1
	Main socket for UDP SIP communication.
struct ast_sockaddr	bindaddr
static struct ast_sockaddr	internip
	our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.
static struct ast_sockaddr	externaddr
	our external IP address/port for SIP sessions. externaddr.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:
static struct ast_sockaddr	media_address
static char	externhost [MAXHOSTNAMELEN]
static time_t	externexpire
static int	externrefresh = 10
static uint16_t	externtcpport
static uint16_t	externtlsport
static struct ast_ha *	localaddr
	List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.
static int	ourport_tcp
static int	ourport_tls
static struct ast_sockaddr	debugaddr
static struct ast_config *	notify_types = NULL
struct ast_channel_tech	sip_tech
	Definition of this channel for PBX channel registration.
struct ast_channel_tech	sip_tech_info
	This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.
static struct ast_cc_agent_callbacks	sip_cc_agent_callbacks
struct ao2_container *	sip_monitor_instances
static struct ast_cc_monitor_callbacks	sip_cc_monitor_callbacks
static struct ast_tls_config	sip_tls_cfg
	Working TLS connection configuration.
static struct ast_tls_config	default_tls_cfg
	Default TLS connection configuration.
static struct ast_tcptls_session_args	sip_tcp_desc
	The TCP server definition.
static struct ast_tcptls_session_args	sip_tls_desc
	The TCP/TLS server definition.
static struct ast_udptl_protocol	sip_udptl
	Interface structure with callbacks used to connect to UDPTL module.
static struct cfalias	aliases []
static struct ast_threadstorage	sip_content_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_sip_content_buf , .custom_init = NULL , }
static struct _map_x_s	regstatestrings []
static struct _map_x_s	stmodes []
	Report Peer status in character string.
static struct _map_x_s	strefresher_params []
static struct _map_x_s	strefreshers []
static struct _map_x_s	autopeermodes []
static struct _map_x_s	dtmfstr []
	mapping between dtmf flags and strings
static struct _map_x_s	insecurestr []
static struct _map_x_s	allowoverlapstr []
static struct _map_x_s	trust_id_outboundstr []
static struct _map_x_s	faxecmodes []
static struct ast_custom_function	sip_header_function
static struct ast_custom_function	checksipdomain_function
static struct ast_custom_function	sippeer_function
	Structure to declare a dialplan function: SIPPEER.
static struct ast_custom_function	sipchaninfo_function
	Structure to declare a dialplan function: SIPCHANINFO.
static struct ast_msg_tech	sip_msg_tech
static struct ast_rtp_glue	sip_rtp_glue
static char *	app_dtmfmode = "SIPDtmfMode"
static char *	app_sipaddheader = "SIPAddHeader"
static char *	app_sipremoveheader = "SIPRemoveHeader"
static struct ast_cli_entry	cli_sip []
	SIP Cli commands definition.
static struct ast_data_handler	peers_data_provider
static struct ast_data_entry	sip_data_providers []
static struct ast_sip_api_tech	chan_sip_api_provider
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .reload = reload, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .nonoptreq = "res_crypto,chan_local,res_http_websocket", }
static struct ast_module_info *	ast_module_info = &__mod_info
static enum ast_cc_service_type	service_string_to_service_type (const char *const service_string)
static int	sip_epa_register (const struct epa_static_data *static_data)
static void	sip_epa_unregister_all (void)
static void	cc_handle_publish_error (struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry)
static void	cc_epa_destructor (void *data)
static struct epa_static_data *	find_static_data (const char *const event_package)
static struct sip_epa_entry *	create_epa_entry (const char *const event_package, const char *const destination)
static int	cc_esc_publish_handler (struct sip_pvt *pvt, struct sip_request *req, struct event_state_compositor *esc, struct sip_esc_entry *esc_entry)
static void	esc_entry_destructor (void *obj)
static int	esc_hash_fn (const void *obj, const int flags)
static int	esc_cmp_fn (void *obj, void *arg, int flags)
static struct event_state_compositor *	get_esc (const char *const event_package)
static struct sip_esc_entry *	get_esc_entry (const char *entity_tag, struct event_state_compositor *esc)
static int	publish_expire (const void *data)
static void	create_new_sip_etag (struct sip_esc_entry *esc_entry, int is_linked)
static struct sip_esc_entry *	create_esc_entry (struct event_state_compositor *esc, struct sip_request *req, const int expires)
static int	initialize_escs (void)
static void	destroy_escs (void)
static int	temp_pvt_init (void *)
static void	temp_pvt_cleanup (void *)
static void	__init_ts_temp_pvt (void)
	A per-thread temporary pvt structure.
static void	__init_sip_transport_str_buf (void)
	A per-thread buffer for transport to string conversion.
static struct ast_channel *	sip_request_call (const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *dest, int *cause)
	PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.
static int	sip_devicestate (const char *data)
	Part of PBX channel interface.
static int	sip_sendtext (struct ast_channel *ast, const char *text)
static int	sip_call (struct ast_channel *ast, const char *dest, int timeout)
	Initiate SIP call from PBX used from the dial() application.
static int	sip_sendhtml (struct ast_channel *chan, int subclass, const char *data, int datalen)
	Send message with Access-URL header, if this is an HTML URL only!
static int	sip_hangup (struct ast_channel *ast)
	sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup
static int	sip_answer (struct ast_channel *ast)
	sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface
static struct ast_frame *	sip_read (struct ast_channel *ast)
	Read SIP RTP from channel.
static int	sip_write (struct ast_channel *ast, struct ast_frame *frame)
	Send frame to media channel (rtp)
static int	sip_indicate (struct ast_channel *ast, int condition, const void *data, size_t datalen)
	Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.
static int	sip_transfer (struct ast_channel *ast, const char *dest)
	Transfer SIP call.
static int	sip_fixup (struct ast_channel *oldchan, struct ast_channel *newchan)
	sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links
static int	sip_senddigit_begin (struct ast_channel *ast, char digit)
static int	sip_senddigit_end (struct ast_channel *ast, char digit, unsigned int duration)
	Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.
static int	sip_setoption (struct ast_channel *chan, int option, void *data, int datalen)
	Set an option on a SIP dialog.
static int	sip_queryoption (struct ast_channel *chan, int option, void *data, int *datalen)
	Query an option on a SIP dialog.
static const char *	sip_get_callid (struct ast_channel *chan)
	Deliver SIP call ID for the call.
static int	handle_request_do (struct sip_request *req, struct ast_sockaddr *addr)
	Handle incoming SIP message - request or response.
static int	sip_standard_port (enum sip_transport type, int port)
	Returns the port to use for this socket.
static int	sip_prepare_socket (struct sip_pvt *p)
static int	get_address_family_filter (unsigned int transport)
	Helper for dns resolution to filter by address family.
static int	sipsock_read (int *id, int fd, short events, void *ignore)
	Read data from SIP UDP socket.
static int	__sip_xmit (struct sip_pvt *p, struct ast_str *data)
static int	__sip_reliable_xmit (struct sip_pvt *p, uint32_t seqno, int resp, struct ast_str *data, int fatal, int sipmethod)
static void	add_cc_call_info_to_response (struct sip_pvt *p, struct sip_request *resp)
static int	__transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Base transmit response function.
static int	retrans_pkt (const void *data)
	Retransmit SIP message if no answer (Called from scheduler)
static int	transmit_response_using_temp (ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg)
	Transmit response, no retransmits, using a temporary pvt structure.
static int	transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, no retransmits.
static int	transmit_response_reliable (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.
static int	transmit_response_with_date (struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Add date before transmitting response.
static int	transmit_response_with_sdp (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid)
	Used for 200 OK and 183 early media.
static int	transmit_response_with_unsupported (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported)
	Transmit response, no retransmits.
static int	transmit_response_with_auth (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *nonce, enum xmittype reliable, const char *header, int stale)
	Respond with authorization request.
static int	transmit_provisional_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp)
static int	transmit_response_with_allow (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)
	Append Accept header, content length before transmitting response.
static void	transmit_fake_auth_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable)
	Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.
static int	transmit_request (struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch)
	Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry)
static int	transmit_request_with_auth (struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch)
	Transmit SIP request, auth added.
static int	transmit_publish (struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char *const explicit_uri)
static int	transmit_invite (struct sip_pvt *p, int sipmethod, int sdp, int init, const char *const explicit_uri)
	Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it.
static int	transmit_reinvite_with_sdp (struct sip_pvt *p, int t38version, int oldsdp)
	Transmit reinvite with SDP.
static int	transmit_info_with_aoc (struct sip_pvt *p, struct ast_aoc_decoded *decoded)
	Send SIP INFO advice of charge message.
static int	transmit_info_with_digit (struct sip_pvt *p, const char digit, unsigned int duration)
	Send SIP INFO dtmf message, see Cisco documentation on cisco.com.
static int	transmit_info_with_vidupdate (struct sip_pvt *p)
	Send SIP INFO with video update request.
static int	transmit_message (struct sip_pvt *p, int init, int auth)
	Transmit with SIP MESSAGE method.
static int	transmit_refer (struct sip_pvt *p, const char *dest)
	Transmit SIP REFER message (initiated by the transfer() dialplan application.
static int	transmit_notify_with_mwi (struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten)
	Notify user of messages waiting in voicemail (RFC3842)
static int	transmit_notify_with_sipfrag (struct sip_pvt *p, int cseq, char *message, int terminate)
	Notify a transferring party of the status of transfer (RFC3515)
static int	transmit_cc_notify (struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state)
static int	transmit_register (struct sip_registry *r, int sipmethod, const char *auth, const char *authheader)
	Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister())
static int	send_response (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
	Transmit response on SIP request.
static int	send_request (struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
static void	copy_request (struct sip_request *dst, const struct sip_request *src)
	copy SIP request (mostly used to save request for responses)
static void	receive_message (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Receive SIP MESSAGE method messages.
static void	parse_moved_contact (struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward)
	Parse 302 Moved temporalily response.
static int	sip_send_mwi_to_peer (struct sip_peer *peer, int cache_only)
	Send message waiting indication to alert peer that they've got voicemail.
static int	__sip_autodestruct (const void *data)
	Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().
static void *	registry_unref (struct sip_registry *reg, char *tag)
static int	update_call_counter (struct sip_pvt *fup, int event)
	update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.
static int	auto_congest (const void *arg)
	Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.
static struct sip_pvt *	find_call (struct sip_request *req, struct ast_sockaddr *addr, const int intended_method)
	find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_request_do
static void	free_old_route (struct sip_route *route)
	Remove route from route list.
static void	list_route (struct sip_route *route)
	List all routes - mostly for debugging.
static void	build_route (struct sip_pvt *p, struct sip_request *req, int backwards, int resp)
	Build route list from Record-Route header.
static enum check_auth_result	register_verify (struct sip_pvt *p, struct ast_sockaddr *addr, struct sip_request *req, const char *uri)
	Verify registration of user.
static struct sip_pvt *	get_sip_pvt_byid_locked (const char *callid, const char *totag, const char *fromtag)
	Lock dialog lock and find matching pvt lock.
static void	check_pendings (struct sip_pvt *p)
	Check pending actions on SIP call.
static void *	sip_park_thread (void *stuff)
	Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.
static int	sip_park (struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, uint32_t seqno, const char *park_exten, const char *park_context)
static void *	sip_pickup_thread (void *stuff)
	SIP pickup support function Starts in a new thread, then pickup the call.
static int	sip_pickup (struct ast_channel *chan)
	Pickup a call using the subsystem in features.c This is executed in a separate thread.
static int	sip_sipredirect (struct sip_pvt *p, const char *dest)
	Transfer call before connect with a 302 redirect.
static int	is_method_allowed (unsigned int *allowed_methods, enum sipmethod method)
	Check if method is allowed for a device or a dialog.
static void	try_suggested_sip_codec (struct sip_pvt *p)
	Try setting codec suggested by the SIP_CODEC channel variable.
static const char *	get_sdp_iterate (int *start, struct sip_request *req, const char *name)
	Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.
static char	get_sdp_line (int *start, int stop, struct sip_request *req, const char **value)
	Fetches the next valid SDP line between the 'start' line (inclusive) and the 'stop' line (exclusive). Returns the type ('a', 'c', ...) and matching line in reference 'start' is updated with the next line number.
static int	find_sdp (struct sip_request *req)
	Determine whether a SIP message contains an SDP in its body.
static int	process_sdp (struct sip_pvt *p, struct sip_request *req, int t38action)
	Process SIP SDP offer, select formats and activate media channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().
static int	process_sdp_o (const char *o, struct sip_pvt *p)
static int	process_sdp_c (const char *c, struct ast_sockaddr *addr)
static int	process_sdp_a_sendonly (const char *a, int *sendonly)
static int	process_sdp_a_ice (const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
static int	process_sdp_a_dtls (const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance)
static int	process_sdp_a_audio (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec)
static int	process_sdp_a_video (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)
static int	process_sdp_a_text (const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
static int	process_sdp_a_image (const char *a, struct sip_pvt *p)
static void	add_ice_to_sdp (struct ast_rtp_instance *instance, struct ast_str **a_buf)
	Add ICE attributes to SDP.
static void	add_dtls_to_sdp (struct ast_rtp_instance *instance, struct ast_str **a_buf)
	Add DTLS attributes to SDP.
static void	start_ice (struct ast_rtp_instance *instance)
	Start ICE negotiation on an RTP instance.
static void	add_codec_to_sdp (const struct sip_pvt *p, struct ast_format *format, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add codec offer to SDP offer/answer body in INVITE or 200 OK.
static void	add_noncodec_to_sdp (const struct sip_pvt *p, int format, struct ast_str **m_buf, struct ast_str **a_buf, int debug)
	Add RFC 2833 DTMF offer to SDP.
static enum sip_result	add_sdp (struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
	Add Session Description Protocol message.
static void	do_setnat (struct sip_pvt *p)
	Set nat mode on the various data sockets.
static void	stop_media_flows (struct sip_pvt *p)
	Immediately stop RTP, VRTP and UDPTL as applicable.
static int	reply_digest (struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len)
	reply to authentication for outbound registrations
static int	build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)
	Build reply digest.
static enum check_auth_result	check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, const char *uri, enum xmittype reliable)
	Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set)
static enum check_auth_result	check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr, struct sip_peer **authpeer)
	Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.
static int	check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr)
	Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.
static int	check_sip_domain (const char *domain, char *context, size_t len)
	check_sip_domain: Check if domain part of uri is local to our server
static int	add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)
	Add SIP domain to list of domains we are responsible for.
static void	clear_sip_domains (void)
	Clear our domain list (at reload)
static void	add_realm_authentication (struct sip_auth_container **credentials, const char *configuration, int lineno)
static struct sip_auth *	find_realm_authentication (struct sip_auth_container *credentials, const char *realm)
static int	check_rtp_timeout (struct sip_pvt *dialog, time_t t)
	helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked
static int	reload_config (enum channelreloadreason reason)
	Re-read SIP.conf config file.
static void	add_diversion (struct sip_request *req, struct sip_pvt *pvt)
	Add "Diversion" header to outgoing message.
static int	expire_register (const void *data)
	Expire registration of SIP peer.
static void *	do_monitor (void *data)
	The SIP monitoring thread.
static int	restart_monitor (void)
	Start the channel monitor thread.
static void	peer_mailboxes_to_str (struct ast_str **mailbox_str, struct sip_peer *peer)
	list peer mailboxes to CLI
static struct ast_variable *	copy_vars (struct ast_variable *src)
	duplicate a list of channel variables,
static int	dialog_find_multiple (void *obj, void *arg, int flags)
static struct ast_channel *	sip_pvt_lock_full (struct sip_pvt *pvt)
static int	sip_refer_alloc (struct sip_pvt *p)
	Allocate SIP refer structure.
static int	sip_notify_alloc (struct sip_pvt *p)
	Allocate SIP refer structure.
static void	ast_quiet_chan (struct ast_channel *chan)
	Turn off generator data XXX Does this function belong in the SIP channel?
static int	attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)
	Attempt transfer of SIP call This fix for attended transfers on a local PBX.
static int	do_magic_pickup (struct ast_channel *channel, const char *extension, const char *context)
static void	set_peer_nat (const struct sip_pvt *p, struct sip_peer *peer)
	Set the peers nat flags if they are using auto_* settings.
static void	check_for_nat (const struct ast_sockaddr *addr, struct sip_pvt *p)
	Check and see if the requesting UA is likely to be behind a NAT.
static int	extensionstate_update (const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static int	cb_extensionstate (char *context, char *exten, struct ast_state_cb_info *info, void *data)
	Callback for the devicestate notification (SUBSCRIBE) support subsystem.
static int	sip_poke_noanswer (const void *data)
	React to lack of answer to Qualify poke.
static int	sip_poke_peer (struct sip_peer *peer, int force)
	Check availability of peer, also keep NAT open.
static void	sip_poke_all_peers (void)
	Send a poke to all known peers.
static void	sip_peer_hold (struct sip_pvt *p, int hold)
	Change onhold state of a peer using a pvt structure.
static void	mwi_event_cb (const struct ast_event *event, void *userdata)
	Receive MWI events that we have subscribed to.
static void	network_change_event_cb (const struct ast_event *, void *)
static void	acl_change_event_cb (const struct ast_event *event, void *userdata)
static void	sip_keepalive_all_peers (void)
	Send a keepalive to all known peers.
static const char *	sip_nat_mode (const struct sip_pvt *p)
	Display SIP nat mode.
static char *	sip_show_inuse (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command to show calls within limits set by call_limit.
static char *	transfermode2str (enum transfermodes mode)
	Convert transfer mode to text string.
static int	peer_status (struct sip_peer *peer, char *status, int statuslen)
static char *	sip_show_sched (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	_sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Execute sip show peers command.
static struct sip_peer *	_sip_show_peers_one (int fd, struct mansession *s, struct show_peers_context *cont, struct sip_peer *peer)
	Emit informations for one peer during sip show peers command.
static char *	sip_show_peers (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Show Peers command.
static char *	sip_show_objects (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List all allocated SIP Objects (realtime or static)
static void	print_group (int fd, ast_group_t group, int crlf)
	Print call group and pickup group.
static void	print_named_groups (int fd, struct ast_namedgroups *group, int crlf)
	Print named call groups and pickup groups.
static const char *	dtmfmode2str (int mode)
	Convert DTMF mode to printable string.
static int	str2dtmfmode (const char *str)
	maps a string to dtmfmode, returns -1 on error
static const char *	insecure2str (int mode)
	Convert Insecure setting to printable string.
static const char *	allowoverlap2str (int mode)
	Convert AllowOverlap setting to printable string.
static void	cleanup_stale_contexts (char *new, char *old)
	Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.
static void	print_codec_to_cli (int fd, struct ast_codec_pref *pref)
	Print codec list from preference to CLI/manager.
static const char *	domain_mode_to_text (const enum domain_mode mode)
	Print domain mode to cli.
static char *	sip_show_domains (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI command to list local domains.
static char *	_sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Show one peer in detail (main function)
static char *	sip_show_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one peer in detail.
static char *	_sip_qualify_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])
	Send qualify message to peer from cli or manager. Mostly for debugging.
static char *	sip_qualify_peer (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Send an OPTIONS packet to a SIP peer.
static char *	sip_show_registry (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show SIP Registry (registrations with other SIP proxies.
static char *	sip_unregister (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Unregister (force expiration) a SIP peer in the registry via CLI.
static char *	sip_show_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	List global settings for the SIP channel.
static char *	sip_show_mwi (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static const char *	subscription_type2str (enum subscriptiontype subtype)
	Show subscription type in string format.
static struct cfsubscription_types *	find_subscription_type (enum subscriptiontype subtype)
	Find subscription type in array.
static char *	complete_sip_peer (const char *word, int state, int flags2)
	Do completion on peer name.
static char *	complete_sip_registered_peer (const char *word, int state, int flags2)
	Do completion on registered peer name.
static char *	complete_sip_show_history (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show history' CLI.
static char *	complete_sip_show_peer (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show peer' CLI.
static char *	complete_sip_unregister (const char *line, const char *word, int pos, int state)
	Support routine for 'sip unregister' CLI.
static char *	complete_sip_notify (const char *line, const char *word, int pos, int state)
	Support routine for 'sip notify' CLI.
static char *	sip_show_channel (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show details of one active dialog.
static char *	sip_show_channelstats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	SIP show channelstats CLI (main function)
static char *	sip_show_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show history details of one dialog.
static char *	sip_do_debug_ip (int fd, const char *arg)
	Enable SIP Debugging for a single IP.
static char *	sip_do_debug_peer (int fd, const char *arg)
	Turn on SIP debugging for a given peer.
static char *	sip_do_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Turn on SIP debugging (CLI command)
static char *	sip_cli_notify (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Cli command to send SIP notify to peer.
static char *	sip_set_history (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Enable/Disable SIP History logging (CLI)
static int	sip_dtmfmode (struct ast_channel *chan, const char *data)
	Set the DTMFmode for an outbound SIP call (application)
static int	sip_addheader (struct ast_channel *chan, const char *data)
	Add a SIP header to an outbound INVITE.
static int	sip_do_reload (enum channelreloadreason reason)
	Reload module.
static char *	sip_reload (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Force reload of module from cli.
static int	ast_sockaddr_resolve_first_af (struct ast_sockaddr *addr, const char *name, int flag, int family)
	Return the first entry from ast_sockaddr_resolve filtered by address family.
static int	ast_sockaddr_resolve_first (struct ast_sockaddr *addr, const char *name, int flag)
	Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.
static int	ast_sockaddr_resolve_first_transport (struct ast_sockaddr *addr, const char *name, int flag, unsigned int transport)
	Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.
static void	sip_dump_history (struct sip_pvt *dialog)
	Dump SIP history to debug log file at end of lifespan for SIP dialog.
static int	sip_debug_test_addr (const struct ast_sockaddr *addr)
	See if we pass debug IP filter.
static int	sip_debug_test_pvt (struct sip_pvt *p)
	Test PVT for debugging output.
static void	append_history_full (struct sip_pvt *p, const char *fmt,...)
	Append to SIP dialog history with arg list.
static struct sip_peer *	build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only)
	Build peer from configuration (file or realtime static/dynamic)
static void	sip_destroy_peer (struct sip_peer *peer)
	Destroy peer object from memory.
static void	sip_destroy_peer_fn (void *peer)
static void	set_peer_defaults (struct sip_peer *peer)
	Set peer defaults before configuring specific configurations.
static struct sip_peer *	temp_peer (const char *name)
	Create temporary peer (used in autocreatepeer mode)
static void	register_peer_exten (struct sip_peer *peer, int onoff)
	Automatically add peer extension to dial plan.
static int	sip_poke_peer_s (const void *data)
	Poke peer (send qualify to check if peer is alive and well)
static enum parse_register_result	parse_register_contact (struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
	Parse contact header and save registration (peer registration)
static void	reg_source_db (struct sip_peer *peer)
	Get registration details from Asterisk DB.
static void	destroy_association (struct sip_peer *peer)
	Remove registration data from realtime database or AST/DB when registration expires.
static void	set_insecure_flags (struct ast_flags *flags, const char *value, int lineno)
	Parse insecure= setting in sip.conf and set flags according to setting.
static int	handle_common_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v)
	Handle flag-type options common to configuration of devices - peers.
static void	set_socket_transport (struct sip_socket *socket, int transport)
static int	peer_ipcmp_cb_full (void *obj, void *arg, void *data, int flags)
static void	realtime_update_peer (const char *peername, struct ast_sockaddr *addr, const char *defaultuser, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms)
	Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.
static void	update_peer (struct sip_peer *p, int expire)
	Update peer data in database (if used)
static struct ast_variable *	get_insecure_variable_from_config (struct ast_config *config)
static const char *	get_name_from_variable (const struct ast_variable *var)
static struct sip_peer *	realtime_peer (const char *newpeername, struct ast_sockaddr *addr, char *callbackexten, int devstate_only, int which_objects)
	realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.
static char *	sip_prune_realtime (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Remove temporary realtime objects from memory (CLI)
static void	ast_sip_ouraddrfor (const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p)
	NAT fix - decide which IP address to use for Asterisk server?
static void	sip_registry_destroy (struct sip_registry *reg)
	Destroy registry object Objects created with the register= statement in static configuration.
static int	sip_register (const char *value, int lineno)
	create sip_registry object from register=> line in sip.conf and link into reg container
static const char *	regstate2str (enum sipregistrystate regstate)
	Convert registration state status to string.
static int	sip_reregister (const void *data)
	Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.
static int	__sip_do_register (struct sip_registry *r)
	Register with SIP proxy.
static int	sip_reg_timeout (const void *data)
	Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.
static void	sip_send_all_registers (void)
	Send all known registrations.
static int	sip_reinvite_retry (const void *data)
	Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.
static int	determine_firstline_parts (struct sip_request *req)
	Parse first line of incoming SIP request.
static const char *	gettag (const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize)
	Get tag from packet.
static int	find_sip_method (const char *msg)
	find_sip_method: Find SIP method from header
static unsigned int	parse_allowed_methods (struct sip_request *req)
	parse the Allow header to see what methods the endpoint we are communicating with allows.
static unsigned int	set_pvt_allowed_methods (struct sip_pvt *pvt, struct sip_request *req)
static int	parse_request (struct sip_request *req)
	Parse a SIP message.
static const char *	referstatus2str (enum referstatus rstatus)
	Convert transfer status to string.
static int	method_match (enum sipmethod id, const char *name)
	returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send
static void	parse_copy (struct sip_request *dst, const struct sip_request *src)
	Copy SIP request, parse it.
static void	parse_oli (struct sip_request *req, struct ast_channel *chan)
	Check for the presence of OLI tag(s) in the From header and set on the channel.
static const char *	find_alias (const char *name, const char *_default)
	Find compressed SIP alias.
static const char *	__get_header (const struct sip_request *req, const char *name, int *start)
static void	lws2sws (struct ast_str *data)
	Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.
static void	extract_uri (struct sip_pvt *p, struct sip_request *req)
	Check Contact: URI of SIP message.
static char *	remove_uri_parameters (char *uri)
static int	get_refer_info (struct sip_pvt *transferer, struct sip_request *outgoing_req)
	Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.
static int	get_also_info (struct sip_pvt *p, struct sip_request *oreq)
	Call transfer support (old way, deprecated by the IETF)
static int	parse_ok_contact (struct sip_pvt *pvt, struct sip_request *req)
	Save contact header for 200 OK on INVITE.
static int	set_address_from_contact (struct sip_pvt *pvt)
	Change the other partys IP address based on given contact.
static void	check_via (struct sip_pvt *p, const struct sip_request *req)
	check Via: header for hostname, port and rport request/answer
static int	get_rpid (struct sip_pvt *p, struct sip_request *oreq)
	Get name, number and presentation from remote party id header, returns true if a valid header was found and it was different from the current caller id.
static int	get_rdnis (struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason)
	Get referring dnis.
static enum sip_get_dest_result	get_destination (struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id)
	Find out who the call is for.
static int	transmit_state_notify (struct sip_pvt *p, struct state_notify_data *data, int full, int timeout)
	Used in the SUBSCRIBE notification subsystem (RFC3265)
static void	update_connectedline (struct sip_pvt *p, const void *data, size_t datalen)
	Notify peer that the connected line has changed.
static void	update_redirecting (struct sip_pvt *p, const void *data, size_t datalen)
	Send a provisional response indicating that a call was redirected.
static int	get_domain (const char *str, char *domain, int len)
	Extract domain from SIP To/From header.
static void	get_realm (struct sip_pvt *p, const struct sip_request *req)
	Choose realm based on From header and then To header or use globaly configured realm. Realm from From/To header should be listed among served domains in config file: domain=...
static char *	get_content (struct sip_request *req)
	Get message body content.
static void *	_sip_tcp_helper_thread (struct ast_tcptls_session_instance *tcptls_session)
	SIP TCP thread management function This function reads from the socket, parses the packet into a request.
static void *	sip_tcp_worker_fn (void *data)
	SIP TCP connection handler.
static void	initialize_initreq (struct sip_pvt *p, struct sip_request *req)
	Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.
static int	init_req (struct sip_request *req, int sipmethod, const char *recip)
	Initialize SIP request.
static void	deinit_req (struct sip_request *req)
	Deinitialize SIP response/request.
static int	reqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, uint32_t seqno, int newbranch)
	Initialize a SIP request message (not the initial one in a dialog)
static void	initreqprep (struct sip_request *req, struct sip_pvt *p, int sipmethod, const char *const explicit_uri)
	Initiate new SIP request to peer/user.
static int	init_resp (struct sip_request *resp, const char *msg)
	Initialize SIP response, based on SIP request.
static int	resp_needs_contact (const char *msg, enum sipmethod method)
	Test if this response needs a contact header.
static int	respprep (struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req)
	Prepare SIP response packet.
static struct ast_sockaddr *	sip_real_dst (const struct sip_pvt *p)
	The real destination address for a write.
static void	build_via (struct sip_pvt *p)
	Build a Via header for a request.
static int	create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)
	Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.
static int	create_addr (struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog)
	create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success
static char *	generate_random_string (char *buf, size_t size)
	Generate 32 byte random string for callid's etc.
static void	build_callid_pvt (struct sip_pvt *pvt)
	Build SIP Call-ID value for a non-REGISTER transaction.
static void	change_callid_pvt (struct sip_pvt *pvt, const char *callid)
static void	build_callid_registry (struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain)
	Build SIP Call-ID value for a REGISTER transaction.
static void	build_localtag_registry (struct sip_registry *reg)
	Build SIP From tag value for REGISTER.
static void	make_our_tag (struct sip_pvt *pvt)
	Make our SIP dialog tag.
static int	add_header (struct sip_request *req, const char *var, const char *value)
	Add header to SIP message.
static int	add_max_forwards (struct sip_pvt *dialog, struct sip_request *req)
	Add 'Max-Forwards' header to SIP message.
static int	add_content (struct sip_request *req, const char *line)
	Add content (not header) to SIP message.
static int	finalize_content (struct sip_request *req)
	Add 'Content-Length' header and content to SIP message.
static void	destroy_msg_headers (struct sip_pvt *pvt)
static int	add_text (struct sip_request *req, struct sip_pvt *p)
	Add text body to SIP message.
static int	add_digit (struct sip_request *req, char digit, unsigned int duration, int mode)
	Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.
static int	add_rpid (struct sip_request *req, struct sip_pvt *p)
	Add Remote-Party-ID header to SIP message.
static int	add_vidupdate (struct sip_request *req)
	add XML encoded media control with update
static void	add_route (struct sip_request *req, struct sip_route *route)
	Add route header into request per learned route.
static int	copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy one header field from one request to another.
static int	copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy all headers from one request to another.
static int	copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)
	Copy SIP VIA Headers from the request to the response.
static void	set_destination (struct sip_pvt *p, char *uri)
	Set destination from SIP URI.
static void	add_date (struct sip_request *req)
	Add date header to SIP message.
static void	add_expires (struct sip_request *req, int expires)
	Add Expires header to SIP message.
static void	build_contact (struct sip_pvt *p)
	Build contact header - the contact header we send out.
static int	handle_incoming (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock)
	Handle incoming SIP requests (methods)
static int	handle_request_update (struct sip_pvt *p, struct sip_request *req)
	bare-bones support for SIP UPDATE
static int	handle_request_invite (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const char *e, int *nounlock)
	Handle incoming INVITE request.
static int	handle_request_refer (struct sip_pvt *p, struct sip_request *req, uint32_t seqno, int *nounlock)
static int	handle_request_bye (struct sip_pvt *p, struct sip_request *req)
	Handle incoming BYE request.
static int	handle_request_register (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Handle incoming REGISTER request.
static int	handle_request_cancel (struct sip_pvt *p, struct sip_request *req)
	Handle incoming CANCEL request.
static int	handle_request_message (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Handle incoming MESSAGE request.
static int	handle_request_subscribe (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e)
	Handle incoming SUBSCRIBE request.
static void	handle_request_info (struct sip_pvt *p, struct sip_request *req)
	Receive SIP INFO Message.
static int	handle_request_options (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
	Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.
static int	handle_invite_replaces (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *nounlock)
	Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.
static int	handle_request_notify (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e)
	Handle incoming notifications.
static int	local_attended_transfer (struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, uint32_t seqno, int *nounlock)
	Find all call legs and bridge transferee with target called from handle_request_refer.
static void	handle_response_publish (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static void	handle_response_invite (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
	Handle SIP response to INVITE dialogue.
static void	handle_response_notify (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static void	handle_response_refer (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static void	handle_response_subscribe (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static int	handle_response_register (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
	Handle responses on REGISTER to services.
static void	handle_response (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
	Handle SIP response in dialogue.
static int	setup_srtp (struct sip_srtp **srtp)
static int	process_crypto (struct sip_pvt *p, struct ast_rtp_instance *rtp, struct sip_srtp **srtp, const char *a)
static int	transmit_response_with_t38_sdp (struct sip_pvt *p, char *msg, struct sip_request *req, int retrans)
	Used for 200 OK and 183 early media.
static struct ast_udptl *	sip_get_udptl_peer (struct ast_channel *chan)
static int	sip_set_udptl_peer (struct ast_channel *chan, struct ast_udptl *udptl)
static void	change_t38_state (struct sip_pvt *p, int state)
	Change the T38 state on a SIP dialog.
static void	proc_422_rsp (struct sip_pvt *p, struct sip_request *rsp)
	Handle 422 response to INVITE with session-timer requested.
static int	proc_session_timer (const void *vp)
	Session-Timers: Process session refresh timeout event.
static void	stop_session_timer (struct sip_pvt *p)
	Session-Timers: Stop session timer.
static void	start_session_timer (struct sip_pvt *p)
	Session-Timers: Start session timer.
static void	restart_session_timer (struct sip_pvt *p)
	Session-Timers: Restart session timer.
static const char *	strefresherparam2str (enum st_refresher r)
static int	parse_session_expires (const char *p_hdrval, int *const p_interval, enum st_refresher_param *const p_ref)
	Session-Timers: Function for parsing Session-Expires header.
static int	parse_minse (const char *p_hdrval, int *const p_interval)
	Session-Timers: Function for parsing Min-SE header.
static int	st_get_se (struct sip_pvt *p, int max)
	Get Max or Min SE (session timer expiry)
static enum st_refresher	st_get_refresher (struct sip_pvt *p)
	Get the entity (UAC or UAS) that's acting as the session-timer refresher.
static enum st_mode	st_get_mode (struct sip_pvt *p, int no_cached)
	Get the session-timer mode.
static struct sip_st_dlg *	sip_st_alloc (struct sip_pvt *const p)
	Allocate Session-Timers struct w/in dialog.
static int	sip_set_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active)
static int	sip_subscribe_mwi (const char *value, int lineno)
	Parse mwi=> line in sip.conf and add to list.
static void	sip_subscribe_mwi_destroy (struct sip_subscription_mwi *mwi)
	Destroy MWI subscription object.
static void	sip_send_all_mwi_subscriptions (void)
	Send all MWI subscriptions.
static int	sip_subscribe_mwi_do (const void *data)
	Send a subscription or resubscription for MWI.
static int	__sip_subscribe_mwi_do (struct sip_subscription_mwi *mwi)
	Actually setup an MWI subscription or resubscribe.
static int	sip_cc_agent_init (struct ast_cc_agent *agent, struct ast_channel *chan)
static int	sip_cc_agent_start_offer_timer (struct ast_cc_agent *agent)
static int	sip_cc_agent_stop_offer_timer (struct ast_cc_agent *agent)
static void	sip_cc_agent_respond (struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason)
static int	sip_cc_agent_status_request (struct ast_cc_agent *agent)
static int	sip_cc_agent_start_monitoring (struct ast_cc_agent *agent)
static int	sip_cc_agent_recall (struct ast_cc_agent *agent)
static void	sip_cc_agent_destructor (struct ast_cc_agent *agent)
static int	find_by_notify_uri_helper (void *obj, void *arg, int flags)
static struct ast_cc_agent *	find_sip_cc_agent_by_notify_uri (const char *const uri)
static int	find_by_subscribe_uri_helper (void *obj, void *arg, int flags)
static struct ast_cc_agent *	find_sip_cc_agent_by_subscribe_uri (const char *const uri)
static int	find_by_callid_helper (void *obj, void *arg, int flags)
static struct ast_cc_agent *	find_sip_cc_agent_by_original_callid (struct sip_pvt *pvt)
static int	sip_offer_timer_expire (const void *data)
static int	sip_monitor_instance_hash_fn (const void *obj, const int flags)
static int	sip_monitor_instance_cmp_fn (void *obj, void *arg, int flags)
static void	sip_monitor_instance_destructor (void *data)
static struct sip_monitor_instance *	sip_monitor_instance_init (int core_id, const char *const subscribe_uri, const char *const peername, const char *const device_name)
static int	find_sip_monitor_instance_by_subscription_pvt (void *obj, void *arg, int flags)
static int	find_sip_monitor_instance_by_suspension_entry (void *obj, void *arg, int flags)
static int	sip_cc_monitor_request_cc (struct ast_cc_monitor *monitor, int *available_timer_id)
static int	sip_cc_monitor_suspend (struct ast_cc_monitor *monitor)
static int	sip_cc_monitor_unsuspend (struct ast_cc_monitor *monitor)
static int	sip_cc_monitor_cancel_available_timer (struct ast_cc_monitor *monitor, int *sched_id)
static void	sip_cc_monitor_destructor (void *private_data)
static int	construct_pidf_body (enum sip_cc_publish_state state, char *pidf_body, size_t size, const char *presentity)
static int	sip_get_cc_information (struct sip_request *req, char *subscribe_uri, size_t size, enum ast_cc_service_type *service)
static void	sip_handle_cc (struct sip_pvt *pvt, struct sip_request *req, enum ast_cc_service_type service)
struct sip_pvt *	dialog_ref_debug (struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
struct sip_pvt *	dialog_unref_debug (struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
static const char *	map_x_s (const struct _map_x_s *table, int x, const char *errorstring)
	map from an integer value to a string. If no match is found, return errorstring
static int	map_s_x (const struct _map_x_s *table, const char *s, int errorvalue)
	map from a string to an integer value, case insensitive. If no match is found, return errorvalue.
static enum AST_REDIRECTING_REASON	sip_reason_str_to_code (const char *text)
static const char *	sip_reason_code_to_str (enum AST_REDIRECTING_REASON code)
static void	tcptls_packet_destructor (void *obj)
static void	sip_tcptls_client_args_destructor (void *obj)
static void	sip_threadinfo_destructor (void *obj)
static struct sip_threadinfo *	sip_threadinfo_create (struct ast_tcptls_session_instance *tcptls_session, int transport)
	creates a sip_threadinfo object and links it into the threadt table.
static int	sip_tcptls_write (struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
	used to indicate to a tcptls thread that data is ready to be written
static void	sip_websocket_callback (struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
	SIP WebSocket connection handler.
static int	sip_check_authtimeout (time_t start)
	Check if the authtimeout has expired.
static int	sip_tls_read (struct sip_request *req, struct sip_request *reqcpy, struct ast_tcptls_session_instance *tcptls_session, int authenticated, time_t start, struct sip_threadinfo *me)
	Read a SIP request or response from a TLS connection.
static int	read_raw_content_length (const char *message)
	Get the content length from an unparsed SIP message.
static enum message_integrity	check_message_integrity (struct ast_str **request, struct ast_str **overflow)
	Check that a message received over TCP is a full message.
static int	sip_tcp_read (struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session, int authenticated, time_t start)
	Read SIP request or response from a TCP connection.
void *	sip_unref_peer (struct sip_peer *peer, char *tag)
struct sip_peer *	sip_ref_peer (struct sip_peer *peer, char *tag)
static void	peer_sched_cleanup (struct sip_peer *peer)
static int	match_and_cleanup_peer_sched (void *peerobj, void *arg, int flags)
static void	unlink_peers_from_tables (peer_unlink_flag_t flag)
static void	unlink_marked_peers_from_tables (void)
static void	unlink_all_peers_from_tables (void)
static void	unlink_peer_from_tables (struct sip_peer *peer)
static void	ref_proxy (struct sip_pvt *pvt, struct sip_proxy *proxy)
	maintain proper refcounts for a sip_pvt's outboundproxy
void	dialog_unlink_all (struct sip_pvt *dialog)
	Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.
static struct sip_registry *	registry_addref (struct sip_registry *reg, char *tag)
	Add object reference to SIP registry.
static void	pvt_set_needdestroy (struct sip_pvt *pvt, const char *reason)
static void	sip_alreadygone (struct sip_pvt *dialog)
	Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.
static int	proxy_update (struct sip_proxy *proxy)
static struct sip_proxy *	proxy_from_config (const char *proxy, int sipconf_lineno, struct sip_proxy *dest)
	Parse proxy string and return an ao2_alloc'd proxy. If dest is non-NULL, no allocation is performed and dest is used instead. On error NULL is returned.
unsigned int	port_str2int (const char *pt, unsigned int standard)
	converts ascii port to int representation. If no pt buffer is provided or the pt has errors when being converted to an int value, the port provided as the standard is used.
static struct sip_proxy *	obproxy_get (struct sip_pvt *dialog, struct sip_peer *peer)
	Get default outbound proxy or global proxy.
static int	get_transport_str2enum (const char *transport)
	Return int representing a bit field of transport types found in const char *transport.
static const char *	get_transport_list (unsigned int transports)
	Return configuration of transports for a device.
const char *	sip_get_transport (enum sip_transport t)
	Return transport as string.
static const char *	get_srv_protocol (enum sip_transport t)
	Return protocol string for srv dns query.
static const char *	get_srv_service (enum sip_transport t)
	Return service string for srv dns query.
static const char *	get_transport_pvt (struct sip_pvt *p)
	Return transport of dialog.
static void	append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)
	Append to SIP dialog history with arg list.
void	sip_scheddestroy_final (struct sip_pvt *p, int ms)
	Schedule final destruction of SIP dialog. This can not be canceled. This function is used to keep a dialog around for a period of time in order to properly respond to any retransmits.
void	sip_scheddestroy (struct sip_pvt *p, int ms)
	Schedule destruction of SIP dialog.
int	sip_cancel_destroy (struct sip_pvt *p)
	Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.
int	__sip_ack (struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod)
	Acknowledges receipt of a packet and stops retransmission called with p locked.
void	__sip_pretend_ack (struct sip_pvt *p)
	Pretend to ack all packets called with p locked.
int	__sip_semi_ack (struct sip_pvt *p, uint32_t seqno, int resp, int sipmethod)
	Acks receipt of packet, keep it around (used for provisional responses)
static void	add_blank (struct sip_request *req)
	add a blank line if no body
static int	send_provisional_keepalive_full (struct sip_pvt *pvt, int with_sdp)
static int	send_provisional_keepalive (const void *data)
static int	send_provisional_keepalive_with_sdp (const void *data)
static void	update_provisional_keepalive (struct sip_pvt *pvt, int with_sdp)
static void	add_required_respheader (struct sip_request *req)
static void	enable_dsp_detect (struct sip_pvt *p)
static void	disable_dsp_detect (struct sip_pvt *p)
const char *	find_closing_quote (const char *start, const char *lim)
	Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.
static void	destroy_mailbox (struct sip_mailbox *mailbox)
static void	clear_peer_mailboxes (struct sip_peer *peer)
static struct ast_variable *	get_insecure_variable_from_sippeers (const char *column, const char *value)
static struct ast_variable *	get_insecure_variable_from_sipregs (const char *column, const char *value, struct ast_variable **var)
static int	realtime_peer_by_name (const char *const *name, struct ast_sockaddr *addr, const char *ipaddr, struct ast_variable **var, struct ast_variable **varregs)
static struct ast_variable *	realtime_peer_get_sippeer_helper (const char **name, struct ast_variable **varregs)
static int	realtime_peer_by_addr (const char **name, struct ast_sockaddr *addr, const char *ipaddr, const char *callbackexten, struct ast_variable **var, struct ast_variable **varregs)
static int	register_realtime_peers_with_callbackextens (void)
static int	find_by_name (void *obj, void *arg, void *data, int flags)
static struct sip_peer *	sip_find_peer_full (const char *peer, struct ast_sockaddr *addr, char *callbackexten, int realtime, int which_objects, int devstate_only, int transport)
struct sip_peer *	sip_find_peer (const char *peer, struct ast_sockaddr *addr, int realtime, int which_objects, int devstate_only, int transport)
	Locate device by name or ip address.
static struct sip_peer *	sip_find_peer_by_ip_and_exten (struct ast_sockaddr *addr, char *callbackexten, int transport)
static void	set_t38_capabilities (struct sip_pvt *p)
	Set the global T38 capabilities on a SIP dialog structure.
static void	copy_socket_data (struct sip_socket *to_sock, const struct sip_socket *from_sock)
static int	dialog_initialize_dtls_srtp (const struct sip_pvt *dialog, struct ast_rtp_instance *rtp, struct sip_srtp **srtp)
	Initialize DTLS-SRTP support on an RTP instance.
static int	dialog_initialize_rtp (struct sip_pvt *dialog)
	Initialize RTP portion of a dialog.
static int	default_sip_port (enum sip_transport type)
	The default sip port for the given transport.
static void	offered_media_list_destroy (struct sip_pvt *p)
	Destroy SDP media offer list.
void	__sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)
	Execute destruction of SIP dialog structure, release memory.
static void	sip_destroy_fn (void *p)
struct sip_pvt *	sip_destroy (struct sip_pvt *p)
	Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.
int	hangup_sip2cause (int cause)
	Convert SIP hangup causes to Asterisk hangup causes.
const char *	hangup_cause2sip (int cause)
	Convert Asterisk hangup causes to SIP codes.
static int	reinvite_timeout (const void *data)
static int	interpret_t38_parameters (struct sip_pvt *p, const struct ast_control_t38_parameters *parameters)
	Helper function which updates T.38 capability information and triggers a reinvite.
static int	initialize_udptl (struct sip_pvt *p)
static int	sipinfo_send (struct ast_channel *chan, struct ast_variable *headers, const char *content_type, const char *content, const char *useragent_filter)
static struct ast_channel *	sip_new (struct sip_pvt *i, int state, const char *title, const char *linkedid, struct ast_callid *callid)
	Initiate a call in the SIP channel.
static char *	get_content_line (struct sip_request *req, char *name, char delimiter)
	Get a specific line from the message content.
static const char *	find_full_alias (const char *name, const char *_default)
	Find full SIP alias.
const char *	sip_get_header (const struct sip_request *req, const char *name)
	Get header from SIP request.
static void	__init_sip_content_buf (void)
static struct ast_frame *	sip_rtp_read (struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
	Read RTP from network.
static char *	generate_uri (struct sip_pvt *pvt, char *buf, size_t size)
static void	sip_pvt_callid_set (struct sip_pvt *pvt, struct ast_callid *callid)
struct sip_pvt *	sip_alloc (ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, struct sip_request *req, struct ast_callid *logger_callid)
	Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.
static int	process_via (struct sip_pvt *p, const struct sip_request *req)
	Process the Via header according to RFC 3261 section 18.2.2.
static enum match_req_res	match_req_to_dialog (struct sip_pvt *sip_pvt_ptr, struct match_req_args *arg)
static void	forked_invite_init (struct sip_request *req, const char *new_theirtag, struct sip_pvt *original, struct ast_sockaddr *addr)
	This function creates a dialog to handle a forked request. This dialog exists only to properly terminiate the the forked request immediately.
static void	mark_method_allowed (unsigned int *allowed_methods, enum sipmethod method)
static void	mark_method_unallowed (unsigned int *allowed_methods, enum sipmethod method)
static void	mark_parsed_methods (unsigned int *methods, char *methods_str)
static void	change_hold_state (struct sip_pvt *dialog, struct sip_request *req, int holdstate, int sendonly)
	Change hold state for a call.
static int	get_ip_and_port_from_sdp (struct sip_request *req, const enum media_type media, struct ast_sockaddr *addr)
static int	sockaddr_is_null_or_any (const struct ast_sockaddr *addr)
static int	has_media_stream (struct sip_pvt *p, enum media_type m)
	Check the media stream list to see if the given type already exists.
static int	add_supported (struct sip_pvt *pvt, struct sip_request *req)
	Add "Supported" header to sip message. Since some options may be disabled in the config, the sip_pvt must be inspected to determine what is supported for this dialog.
static int	transmit_response_with_sip_etag (struct sip_pvt *p, const char *msg, const struct sip_request *req, struct sip_esc_entry *esc_entry, int need_new_etag)
static int	transmit_response_with_minse (struct sip_pvt *p, const char *msg, const struct sip_request *req, int minse_int)
	Transmit 422 response with Min-SE header (Session-Timers)
static int	transmit_response_with_retry_after (struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *seconds)
	Append Retry-After header field when transmitting response.
static int	transmit_response_with_minexpires (struct sip_pvt *p, const char *msg, const struct sip_request *req, int minexpires)
	Append Min-Expires header, content length before transmitting response.
static void	add_msg_header (struct sip_pvt *pvt, const char *hdr_name, const char *hdr_value)
static void	add_vcodec_to_sdp (const struct sip_pvt *p, struct ast_format *format, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add video codec offer to SDP offer/answer body in INVITE or 200 OK.
static void	add_tcodec_to_sdp (const struct sip_pvt *p, struct ast_format *format, struct ast_str **m_buf, struct ast_str **a_buf, int debug, int *min_packet_size)
	Add text codec offer to SDP offer/answer body in INVITE or 200 OK.
static unsigned int	t38_get_rate (enum ast_control_t38_rate rate)
	Get Max T.38 Transmission rate from T38 capabilities.
static void	get_our_media_address (struct sip_pvt *p, int needvideo, int needtext, struct ast_sockaddr *addr, struct ast_sockaddr *vaddr, struct ast_sockaddr *taddr, struct ast_sockaddr *dest, struct ast_sockaddr *vdest, struct ast_sockaddr *tdest)
	Set all IP media addresses for this call.
static void	get_crypto_attrib (struct sip_pvt *p, struct sip_srtp *srtp, const char **a_crypto)
static char *	get_sdp_rtp_profile (const struct sip_pvt *p, unsigned int secure, struct ast_rtp_instance *instance)
static void	on_dns_update_registry (struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
static void	on_dns_update_peer (struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
static void	on_dns_update_mwi (struct ast_sockaddr *old, struct ast_sockaddr *new, void *data)
static struct ast_channel *	find_ringing_channel (struct ao2_container *device_state_info, struct sip_pvt *p)
static int	allow_notify_user_presence (struct sip_pvt *p)
static void	state_notify_build_xml (struct state_notify_data *data, int full, const char *exten, const char *context, struct ast_str **tmp, struct sip_pvt *p, int subscribed, const char *mfrom, const char *mto)
	Builds XML portion of NOTIFY messages for presence or dialog updates.
static int	manager_sipnotify (struct mansession *s, const struct message *m)
static const char *	sip_sanitized_host (const char *host)
void	sip_auth_headers (enum sip_auth_type code, char **header, char **respheader)
	return the request and response header for a 401 or 407 code
static int	parse_uri_legacy_check (char *uri, const char *scheme, char **user, char **pass, char **hostport, char **transport)
	parse uri in a way that allows semicolon stripping if legacy mode is enabled
static int	__set_address_from_contact (const char *fullcontact, struct ast_sockaddr *addr, int tcp)
static void	build_nonce (struct sip_pvt *p, int forceupdate)
	builds the sip_pvt's nonce field which is used for the authentication challenge. When forceupdate is not set, the nonce is only updated if the current one is stale. In this case, a stalenonce is one which has already received a response, if a nonce has not received a response it is not always necessary or beneficial to create a new one.
void	sip_digest_parser (char *c, struct digestkeys *keys)
	Takes the digest response and parses it.
static void	network_change_event_subscribe (void)
static void	network_change_event_unsubscribe (void)
static void	acl_change_event_subscribe (void)
static void	acl_change_event_unsubscribe (void)
static int	network_change_event_sched_cb (const void *data)
static void	cb_extensionstate_destroy (int id, void *data)
static char *	terminate_uri (char *uri)
static void	extract_host_from_hostport (char **hostport)
	Terminate a host:port at the ':'.
static void	update_peer_lastmsgssent (struct sip_peer *peer, int value, int locked)
static void	sip_set_redirstr (struct sip_pvt *p, char *reason)
	Translate referring cause.
static int	get_pai (struct sip_pvt *p, struct sip_request *req)
	Parse the parts of the P-Asserted-Identity header on an incoming packet. Returns 1 if a valid header is found and it is different from the current caller id.
static enum check_auth_result	check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct ast_sockaddr *addr, struct sip_peer **authpeer, enum xmittype reliable, char *calleridname, char *uri2)
	Validate device authentication.
static int	set_message_vars_from_req (struct ast_msg *msg, struct sip_request *req)
static const char *	stmode2str (enum st_mode m)
static enum st_mode	str2stmode (const char *s)
static enum st_refresher	str2strefresherparam (const char *s)
static const char *	strefresher2str (enum st_refresher r)
static const char *	autocreatepeer2str (enum autocreatepeer_mode r)
static char *	sip_show_tcp (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show active TCP connections.
static char *	sip_show_users (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI Command 'SIP Show Users'.
static int	manager_show_registry (struct mansession *s, const struct message *m)
	Show SIP registrations in the manager API.
static int	manager_sip_show_peers (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
int	peercomparefunc (const void *a, const void *b)
static int	peer_dump_func (void *userobj, void *arg, int flags)
static int	dialog_dump_func (void *userobj, void *arg, int flags)
static const char *	trust_id_outbound2str (int mode)
static int	dialog_checkrtp_cb (void *dialogobj, void *arg, int flags)
	Check RTP Timeout on dialogs.
static int	dialog_needdestroy (void *dialogobj, void *arg, int flags)
	Match dialogs that need to be destroyed.
static int	manager_sip_show_peer (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static void	send_manager_peer_status (struct mansession *s, struct sip_peer *peer, const char *idText)
static int	manager_sip_peer_status (struct mansession *s, const struct message *m)
	Show SIP peers in the manager API.
static int	manager_sip_qualify_peer (struct mansession *s, const struct message *m)
	Qualify SIP peers in the manager API.
static const char *	faxec2str (int faxec)
static char *	complete_sip_user (const char *word, int state)
	Do completion on user name.
static char *	complete_sip_show_user (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show user' CLI.
static char *	sip_show_user (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	Show one user in detail.
static int	show_chanstats_cb (void *__cur, void *__arg, int flags)
	Callback for show_chanstats.
static int	show_channels_cb (void *__cur, void *__arg, int flags)
	callback for show channel|subscription
static char *	sip_show_channels (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
	CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.
static char *	complete_sipch (const char *line, const char *word, int pos, int state)
	Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.
static int	do_register_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code)
	Authenticate for outbound registration.
static int	do_proxy_auth (struct sip_pvt *p, struct sip_request *req, enum sip_auth_type code, int sipmethod, int init)
	Add authentication on outbound SIP packet.
static int	func_header_read (struct ast_channel *chan, const char *function, char *data, char *buf, size_t len)
	Read SIP header (dialplan function)
static int	func_check_sipdomain (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	Dial plan function to check if domain is local.
static int	function_sippeer (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPPEER()} Dialplan function - reads peer data
static int	function_sipchaninfo_read (struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
	${SIPCHANINFO()} Dialplan function - reads sip channel data
static void	change_redirecting_information (struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, struct ast_set_party_redirecting *update_redirecting, int set_call_forward)
	update redirecting information for a channel based on headers
static void	handle_response_update (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
	Handle authentication challenge for SIP UPDATE.
static void	sip_queue_hangup_cause (struct sip_pvt *p, int cause)
static void	handle_response_peerpoke (struct sip_pvt *p, int resp, struct sip_request *req)
	Handle qualification responses (OPTIONS)
static void	handle_response_info (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static int	do_message_auth (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static void	handle_response_message (struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno)
static int	handle_cc_notify (struct sip_pvt *pvt, struct sip_request *req)
static int	sip_t38_abort (const void *data)
	Called to deny a T38 reinvite if the core does not respond to our request.
static int	handle_request_invite_st (struct sip_pvt *p, struct sip_request *req, const char *required, int reinvite)
static int	sip_msg_send (const struct ast_msg *msg, const char *to, const char *from)
static int	block_msg_header (const char *header_name)
static enum sip_publish_type	determine_sip_publish_type (struct sip_request *req, const char *const event, const char *const etag, const char *const expires, int *expires_int)
static int	pidf_validate_tuple (struct ast_xml_node *tuple_node)
static int	pidf_validate_presence (struct ast_xml_doc *doc)
static int	sip_pidf_validate (struct sip_request *req, struct ast_xml_doc **pidf_doc)
	Makes sure that body is properly formatted PIDF.
static int	handle_sip_publish_initial (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const int expires)
static int	handle_sip_publish_refresh (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag, const int expires)
static int	handle_sip_publish_modify (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag, const int expires)
static int	handle_sip_publish_remove (struct sip_pvt *p, struct sip_request *req, struct event_state_compositor *esc, const char *const etag)
static int	handle_request_publish (struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const uint32_t seqno, const char *uri)
static void	add_peer_mwi_subs (struct sip_peer *peer)
static int	handle_cc_subscribe (struct sip_pvt *p, struct sip_request *req)
static int	threadinfo_locate_cb (void *obj, void *arg, int flags)
static struct ast_tcptls_session_instance *	sip_tcp_locate (struct ast_sockaddr *s)
	Find thread for TCP/TLS session (based on IP/Port.
static int	get_cached_mwi (struct sip_peer *peer, int *new, int *old)
	Get cached MWI info.
static int	sip_send_keepalive (const void *data)
	Send keep alive packet to peer.
static int	handle_t38_options (struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v, int *maxdatagram)
	Handle T.38 configuration options common to users and peers.
static void	destroy_realm_authentication (void *obj)
static struct ast_variable *	add_var (const char *buf, struct ast_variable *list)
	implement the setvar config line
static void	add_peer_mailboxes (struct sip_peer *peer, const char *value)
static int	peer_markall_func (void *device, void *arg, int flags)
static int	peer_markall_autopeers_func (void *device, void *arg, int flags)
static void	sip_set_default_format_capabilities (struct ast_format_cap *cap)
static void	display_nat_warning (const char *cat, int reason, struct ast_flags *flags)
static void	cleanup_all_regs (void)
static int	apply_directmedia_acl (struct sip_pvt *p, struct ast_acl_list *directmediaacl, const char *op)
static int	sip_allow_anyrtp_remote (struct ast_channel *chan1, struct ast_channel *chan2, char *rtptype)
static int	sip_allow_rtp_remote (struct ast_channel *chan1, struct ast_channel *chan2)
static int	sip_allow_vrtp_remote (struct ast_channel *chan1, struct ast_channel *chan2)
static enum ast_rtp_glue_result	sip_get_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static enum ast_rtp_glue_result	sip_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static enum ast_rtp_glue_result	sip_get_trtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance)
static void	sip_get_codec (struct ast_channel *chan, struct ast_format_cap *result)
static int	sip_removeheader (struct ast_channel *chan, const char *data)
	Remove SIP headers added previously with SipAddHeader application.
static int	sip_is_xml_parsable (void)
static int	reload (void)
	Part of Asterisk module interface.
static int	peer_hash_cb (const void *obj, const int flags)
static int	peer_cmp_cb (void *obj, void *arg, int flags)
static int	peer_iphash_cb (const void *obj, const int flags)
static int	peer_ipcmp_cb (void *obj, void *arg, int flags)
static int	threadt_hash_cb (const void *obj, const int flags)
static int	threadt_cmp_cb (void *obj, void *arg, int flags)
static int	dialog_hash_cb (const void *obj, const int flags)
static int	dialog_cmp_cb (void *obj, void *arg, int flags)
static void	sip_register_tests (void)
	SIP test registration.
static void	sip_unregister_tests (void)
	SIP test registration.
	AST_DATA_STRUCTURE (sip_peer, DATA_EXPORT_SIP_PEER)
static int	peers_data_provider_get (const struct ast_data_search *search, struct ast_data *data_root)
static int	load_module (void)
	PBX load module - initialization.
static int	unload_module (void)
	PBX unload module API.
static void	__reg_module (void)
static void	__unreg_module (void)

---

Detailed Description

Implementation of Session Initiation Protocol.

Author:

Mark Spencer <markster@digium.com>

See Also:

• CREDITS

Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support Configuration file sip.conf

********** IMPORTANT *

Note:

TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration settings, dialplan commands and dialplans apps/functions See SIP TCP and TLS support

******** General TODO:s

Todo:

Better support of forking

VIA branch tag transaction checking

Transaction support

******** Wishlist: Improvements

• Support of SIP domains for devices, so that we match on username in the From: header
• Connect registrations with a specific device on the incoming call. It's not done automatically in Asterisk

Overview of the handling of SIP sessions

The SIP channel handles several types of SIP sessions, or dialogs, not all of them being "telephone calls".

• Incoming calls that will be sent to the PBX core
• Outgoing calls, generated by the PBX
• SIP subscriptions and notifications of states and voicemail messages
• SIP registrations, both inbound and outbound
• SIP peer management (peerpoke, OPTIONS)
• SIP text messages

In the SIP channel, there's a list of active SIP dialogs, which includes all of these when they are active. "sip show channels" in the CLI will show most of these, excluding subscriptions which are shown by "sip show subscriptions"

incoming packets

Incoming packets are received in the monitoring thread, then handled by sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread. sipsock_read() function parses the packet and matches an existing dialog or starts a new SIP dialog.

sipsock_read sends the packet to handle_incoming(), that parses a bit more. If it is a response to an outbound request, the packet is sent to handle_response(). If it is a request, handle_incoming() sends it to one of a list of functions depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc sipsock_read locks the ast_channel if it exists (an active call) and unlocks it after we have processed the SIP message.

A new INVITE is sent to handle_request_invite(), that will end up starting a new channel in the PBX, the new channel after that executing in a separate channel thread. This is an incoming "call". When the call is answered, either by a bridged channel or the PBX itself the sip_answer() function is called.

The actual media - Video or Audio - is mostly handled by the RTP subsystem in rtp.c

Outbound calls

Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up

The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

---

Define Documentation

#define append_history	(		p,
			event,
			fmt,
			args...
	)		append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history.

Returns:

Always returns 0

Definition at line 2325 of file chan_sip.c.

Referenced by __sip_autodestruct(), __sip_reliable_xmit(), auto_congest(), build_reply_digest(), change_hold_state(), check_auth(), do_message_auth(), do_register_auth(), extensionstate_update(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_response(), handle_response_invite(), local_attended_transfer(), obproxy_get(), pvt_set_needdestroy(), retrans_pkt(), send_request(), send_response(), sip_cancel_destroy(), sip_fixup(), sip_hangup(), sip_new(), sip_park_thread(), sip_reregister(), sip_scheddestroy(), sip_set_rtp_peer(), transmit_register(), transmit_reinvite_with_sdp(), transmit_response_with_auth(), and update_connectedline().

#define BOGUS_PEER_MD5SECRET   "intentionally_invalid_md5_string"

We can recognise the bogus peer by this invalid MD5 hash.

Definition at line 1194 of file chan_sip.c.

Referenced by check_auth(), load_module(), and sip_reload().

#define check_request_transport	(		peer,
			tmpl
	)

generic function for determining if a correct transport is being used to contact a peer

this is done as a macro so that the "tmpl" var can be passed either a sip_request or a sip_peer

Definition at line 2412 of file chan_sip.c.

Referenced by create_addr_from_peer(), and register_verify().

#define CONTAINER_UNLINK	(		container,
			obj,
			tag
	)

Unlink the given object from the container and return TRUE if it was in the container.

Definition at line 8523 of file chan_sip.c.

Referenced by change_callid_pvt().

#define DATA_EXPORT_SIP_PEER

(

MEMBER

)

Definition at line 34411 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-47.47s %-9.9s %-6.6s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-39.39s %-6.6s %-12.12s %8d %-20.20s %-25.25s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-15.15s %-11.11s %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-30.30s %-12.12s %-10.10s %-10.10s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT   "%-15.15s %-15.15s %-15.15s %-15.15s %-3.3s %-3.3s %-15.15s %-10.10s %-10.10s\n"

Definition at line 20945 of file chan_sip.c.

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

Definition at line 20944 of file chan_sip.c.

#define FORMAT2   "%-47.47s %9.9s %6.6s\n"

Definition at line 20944 of file chan_sip.c.

#define FORMAT2   "%-39.39s %-6.6s %-12.12s %8.8s %-20.20s %-25.25s\n"

Definition at line 20944 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-11.11s %-8.8s %-10.10s %-10.10s ( %%) %-6.6s %-10.10s %-10.10s ( %%) %-6.6s\n"

Definition at line 20944 of file chan_sip.c.

#define FORMAT2   "%-15.15s %-15.15s %-15.15s %-15.15s %-7.7s %-15.15s %-10.10s %-10.10s\n"

Definition at line 20944 of file chan_sip.c.

#define FORMAT3   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6s\n"

Definition at line 20943 of file chan_sip.c.

Referenced by sip_show_channels().

#define FORMAT4   "%-15.15s %-15.15s %-15.15s %-15.15s %-13.13s %-15.15s %-10.10s %-6.6d\n"

Definition at line 20942 of file chan_sip.c.

Referenced by show_channels_cb().

#define PEERS_FORMAT2   "%-25.25s %-39.39s %-3.3s %-10.10s %-10.10s %-3.3s %-8s %-11s %-32.32s %s\n"

Definition at line 19114 of file chan_sip.c.

Referenced by _sip_show_peers(), and _sip_show_peers_one().

#define SIP_PEDANTIC_DECODE

(

str

)

Value:

if (sip_cfg.pedanticsipchecking && !ast_strlen_zero(str)) { \
      ast_uri_decode(str, ast_uri_sip_user); \
   }  \

Definition at line 758 of file chan_sip.c.

Referenced by check_user_full(), get_also_info(), get_destination(), get_refer_info(), register_verify(), and sip_msg_send().

#define sip_pvt_lock

(

x

)

ao2_lock(x)

Definition at line 1180 of file chan_sip.c.

Referenced by __sip_ack(), __sip_autodestruct(), auto_congest(), complete_sipch(), extensionstate_update(), find_call(), forked_invite_init(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_bye(), handle_request_invite(), handle_request_refer(), handle_request_subscribe(), local_attended_transfer(), parse_register_contact(), proc_session_timer(), register_verify(), retrans_pkt(), show_channels_cb(), show_chanstats_cb(), sip_allow_anyrtp_remote(), sip_answer(), sip_call(), sip_cc_agent_destructor(), sip_cc_agent_init(), sip_cc_agent_recall(), sip_cc_agent_respond(), sip_cc_monitor_request_cc(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_monitor_instance_destructor(), sip_msg_send(), sip_new(), sip_park_thread(), sip_pvt_lock_full(), sip_queryoption(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_send_mwi_to_peer(), sip_senddigit_begin(), sip_senddigit_end(), sip_sendtext(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_setoption(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), sipinfo_send(), transmit_publish(), and update_call_counter().

#define sip_pvt_trylock

(

x

)

ao2_trylock(x)

Definition at line 1181 of file chan_sip.c.

Referenced by dialog_checkrtp_cb(), dialog_needdestroy(), forked_invite_init(), and sip_hangup().

#define sip_pvt_unlock

(

x

)

ao2_unlock(x)

Definition at line 1182 of file chan_sip.c.

Referenced by __sip_ack(), __sip_autodestruct(), auto_congest(), complete_sipch(), dialog_checkrtp_cb(), dialog_needdestroy(), dialog_unlink_all(), extensionstate_update(), find_call(), forked_invite_init(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_refer(), handle_request_subscribe(), local_attended_transfer(), parse_register_contact(), proc_session_timer(), register_verify(), retrans_pkt(), send_provisional_keepalive_full(), show_channels_cb(), show_chanstats_cb(), sip_allow_anyrtp_remote(), sip_answer(), sip_call(), sip_cc_agent_destructor(), sip_cc_agent_init(), sip_cc_agent_recall(), sip_cc_agent_respond(), sip_cc_monitor_request_cc(), sip_dtmfmode(), sip_fixup(), sip_get_rtp_peer(), sip_get_trtp_peer(), sip_get_udptl_peer(), sip_get_vrtp_peer(), sip_hangup(), sip_indicate(), sip_monitor_instance_destructor(), sip_msg_send(), sip_new(), sip_park_thread(), sip_pvt_lock_full(), sip_queryoption(), sip_queue_hangup_cause(), sip_read(), sip_reg_timeout(), sip_reinvite_retry(), sip_request_call(), sip_send_mwi_to_peer(), sip_senddigit_begin(), sip_senddigit_end(), sip_sendtext(), sip_set_rtp_peer(), sip_set_udptl_peer(), sip_setoption(), sip_show_channel(), sip_show_history(), sip_t38_abort(), sip_transfer(), sip_write(), sipinfo_send(), transmit_publish(), and update_call_counter().

#define SIP_TRANSPORT_STR_BUFSIZE   128

Size of the SIP transport buffer.

Definition at line 1216 of file chan_sip.c.

Referenced by get_transport_list().

#define UNLINK	(		element,
			head,
			prev
	)

some list management macros.

Definition at line 1291 of file chan_sip.c.

Referenced by __sip_ack(), handle_request_cancel(), and retrans_pkt().

---

Enumeration Type Documentation

enum match_req_res

Enumerator:

SIP_REQ_MATCH
SIP_REQ_NOT_MATCH
SIP_REQ_LOOP_DETECTED
SIP_REQ_FORKED

Definition at line 8850 of file chan_sip.c.

                   {
   SIP_REQ_MATCH,
   SIP_REQ_NOT_MATCH,
   SIP_REQ_LOOP_DETECTED, /* multiple incoming requests with same call-id but different branch parameters have been detected */
   SIP_REQ_FORKED, /* An outgoing request has been forked as result of receiving two differing 200ok responses. */
};

enum message_integrity

Indication of a TCP message's integrity.

Enumerator:

MESSAGE_INVALID	The message has an error in it with regards to its Content-Length header
MESSAGE_FRAGMENT	The message is incomplete
MESSAGE_FRAGMENT_COMPLETE	The data contains a complete message plus a fragment of another.
MESSAGE_COMPLETE	The message is complete

Definition at line 2813 of file chan_sip.c.

                       {
   /*!
    * The message has an error in it with
    * regards to its Content-Length header
    */
   MESSAGE_INVALID,
   /*!
    * The message is incomplete
    */
   MESSAGE_FRAGMENT,
   /*!
    * The data contains a complete message
    * plus a fragment of another.
    */
   MESSAGE_FRAGMENT_COMPLETE,
   /*!
    * The message is complete
    */
   MESSAGE_COMPLETE,
};

enum peer_unlink_flag_t

Enumerator:

SIP_PEERS_MARKED
SIP_PEERS_ALL

Definition at line 3307 of file chan_sip.c.

             {
   SIP_PEERS_MARKED,
   SIP_PEERS_ALL,
} peer_unlink_flag_t;

---

Function Documentation

static const char * __get_header	(	const struct sip_request *	req,
		const char *	name,
		int *	start
	)		[static]

Definition at line 8256 of file chan_sip.c.

References ast_skip_blanks(), find_alias(), len(), match(), and sip_cfg.

Referenced by build_route(), copy_all_header(), copy_via_headers(), func_header_read(), handle_incoming(), handle_request_subscribe(), handle_response_register(), parse_register_contact(), and sip_get_header().

{
   /*
    * Technically you can place arbitrary whitespace both before and after the ':' in
    * a header, although RFC3261 clearly says you shouldn't before, and place just
    * one afterwards.  If you shouldn't do it, what absolute idiot decided it was
    * a good idea to say you can do it, and if you can do it, why in the hell would.
    * you say you shouldn't.
    * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
    * and we always allow spaces after that for compatibility.
    */
   const char *sname = find_alias(name, NULL);
   int x, len = strlen(name), slen = (sname ? 1 : 0);
   for (x = *start; x < req->headers; x++) {
      const char *header = REQ_OFFSET_TO_STR(req, header[x]);
      int smatch = 0, match = !strncasecmp(header, name, len);
      if (slen) {
         smatch = !strncasecmp(header, sname, slen);
      }
      if (match || smatch) {
         /* skip name */
         const char *r = header + (match ? len : slen );
         if (sip_cfg.pedanticsipchecking) {
            r = ast_skip_blanks(r);
         }

         if (*r == ':') {
            *start = x+1;
            return ast_skip_blanks(r+1);
         }
      }
   }

   /* Don't return NULL, so sip_get_header is always a valid pointer */
   return "";
}

static void __init_sip_content_buf

(

void

)

[static]

Definition at line 8303 of file chan_sip.c.

{

static void __init_sip_transport_str_buf

(

void

)

[static]

A per-thread buffer for transport to string conversion.

Definition at line 1213 of file chan_sip.c.

{  \

static void __init_ts_temp_pvt

(

void

)

[static]

A per-thread temporary pvt structure.

Definition at line 1210 of file chan_sip.c.

{  \

static void __reg_module

(

void

)

[static]

Definition at line 34976 of file chan_sip.c.

static int __set_address_from_contact	(	const char *	fullcontact,
		struct ast_sockaddr *	addr,
		int	tcp
	)		[static]

Definition at line 15930 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_sockaddr_port, ast_sockaddr_resolve_first_transport(), ast_sockaddr_set_port, ast_strlen_zero(), get_transport_str2enum(), LOG_WARNING, and parse_uri_legacy_check().

Referenced by build_peer(), set_address_from_contact(), and sip_request_call().

{
   char *hostport, *transport;
   char contact_buf[256];
   char *contact;

   /* Work on a copy */
   ast_copy_string(contact_buf, fullcontact, sizeof(contact_buf));
   contact = contact_buf;

   /* 
    * We have only the part in <brackets> here so we just need to parse a SIP URI.
    *
    * Note: The outbound proxy could be using UDP between the proxy and Asterisk.
    * We still need to be able to send to the remote agent through the proxy.
    */

   if (parse_uri_legacy_check(contact, "sip:,sips:", &contact, NULL, &hostport,
            &transport)) {
      ast_log(LOG_WARNING, "Invalid contact uri %s (missing sip: or sips:), attempting to use anyway\n", fullcontact);
   }

   /* XXX This could block for a long time XXX */
   /* We should only do this if it's a name, not an IP */
   /* \todo - if there's no PORT number in contact - we are required to check NAPTR/SRV records
      to find transport, port address and hostname. If there's a port number, we have to
      assume that the hostport part is a host name and only look for an A/AAAA record in DNS.
   */

   /* If we took in an invalid URI, hostport may not have been initialized */
   /* ast_sockaddr_resolve requires an initialized hostport string. */
   if (ast_strlen_zero(hostport)) {
      ast_log(LOG_WARNING, "Invalid URI: parse_uri failed to acquire hostport\n");
      return -1;
   }

   if (ast_sockaddr_resolve_first_transport(addr, hostport, 0, get_transport_str2enum(transport))) {
      ast_log(LOG_WARNING, "Invalid host name in Contact: (can't "
         "resolve in DNS) : '%s'\n", hostport);
      return -1;
   }

   /* set port */
   if (!ast_sockaddr_port(addr)) {
      ast_sockaddr_set_port(addr,
                  (get_transport_str2enum(transport) ==
                   SIP_TRANSPORT_TLS ||
                   !strncasecmp(fullcontact, "sips", 4)) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }

   return 0;
}

int __sip_ack	(	struct sip_pvt *	p,
		uint32_t	seqno,
		int	resp,
		int	sipmethod
	)

Acknowledges receipt of a packet and stops retransmission called with p locked.

Definition at line 4507 of file chan_sip.c.

References ast_debug, ast_free, ast_sched_del(), FALSE, ref_proxy(), sip_pvt_lock, sip_pvt_unlock, TRUE, and UNLINK.

Referenced by __sip_pretend_ack(), handle_incoming(), handle_request_invite(), handle_request_publish(), and handle_response().

{
   struct sip_pkt *cur, *prev = NULL;
   const char *msg = "Not Found";   /* used only for debugging */
   int res = FALSE;

   /* If we have an outbound proxy for this dialog, then delete it now since
     the rest of the requests in this dialog needs to follow the routing.
     If obforcing is set, we will keep the outbound proxy during the whole
     dialog, regardless of what the SIP rfc says
   */
   if (p->outboundproxy && !p->outboundproxy->force){
      ref_proxy(p, NULL);
   }

   for (cur = p->packets; cur; prev = cur, cur = cur->next) {
      if (cur->seqno != seqno || cur->is_resp != resp) {
         continue;
      }
      if (cur->is_resp || cur->method == sipmethod) {
         res = TRUE;
         msg = "Found";
         if (!resp && (seqno == p->pendinginvite)) {
            ast_debug(1, "Acked pending invite %u\n", p->pendinginvite);
            p->pendinginvite = 0;
         }
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
         }
         /* This odd section is designed to thwart a
          * race condition in the packet scheduler. There are
          * two conditions under which deleting the packet from the
          * scheduler can fail.
          *
          * 1. The packet has been removed from the scheduler because retransmission
          * is being attempted. The problem is that if the packet is currently attempting
          * retransmission and we are at this point in the code, then that MUST mean
          * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
          * lock temporarily to allow retransmission.
          *
          * 2. The packet has reached its maximum number of retransmissions and has
          * been permanently removed from the packet scheduler. If this is the case, then
          * the packet's retransid will be set to -1. The atomicity of the setting and checking
          * of the retransid to -1 is ensured since in both cases p's lock is held.
          */
         while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         }
         UNLINK(cur, p->packets, prev);
         dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
         if (cur->data) {
            ast_free(cur->data);
         }
         ast_free(cur);
         break;
      }
   }
   ast_debug(1, "Stopping retransmission on '%s' of %s %u: Match %s\n",
      p->callid, resp ? "Response" : "Request", seqno, msg);
   return res;
}

static int __sip_autodestruct

(

const void *

data

)

[static]

Kill a SIP dialog (called only by the scheduler) The scheduler has a reference to this dialog when p->autokillid != -1, and we are called using that reference. So if the event is not rescheduled, we need to call dialog_unref().

Definition at line 4367 of file chan_sip.c.

References __sip_pretend_ack(), append_history, AST_CAUSE_PROTOCOL_ERROR, ast_channel_name(), ast_channel_unlock, ast_channel_unref, ast_debug, AST_EXTENSION_DEACTIVATED, ast_log(), ast_queue_hangup_with_cause(), dialog_unlink_all(), LOG_WARNING, method_match(), NONE, pvt_set_needdestroy(), sip_methods, sip_pvt_lock, sip_pvt_lock_full(), sip_pvt_unlock, sip_scheddestroy(), state_notify_data::state, stop_media_flows(), cfsip_methods::text, transmit_request_with_auth(), transmit_state_notify(), and TRUE.

Referenced by sip_scheddestroy(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *)data;
   struct ast_channel *owner;

   /* If this is a subscription, tell the phone that we got a timeout */
   if (p->subscribed && p->subscribed != MWI_NOTIFICATION && p->subscribed != CALL_COMPLETION) {
      struct state_notify_data data = { 0, };
      data.state = AST_EXTENSION_DEACTIVATED;

      transmit_state_notify(p, &data, 1, TRUE); /* Send last notification */
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "timeout");
      ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
      return 10000;  /* Reschedule this destruction so that we know that it's gone */
   }

   /* If there are packets still waiting for delivery, delay the destruction */
   if (p->packets) {
      if (!p->needdestroy) {
         char method_str[31];
         ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
         append_history(p, "ReliableXmit", "timeout");
         if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
            if (p->ongoing_reinvite || method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
               pvt_set_needdestroy(p, "autodestruct");
            }
         }
         return 10000;
      } else {
         /* They've had their chance to respond. Time to bail */
         __sip_pretend_ack(p);
      }
   }

   /* Reset schedule ID */
   p->autokillid = -1;

   /*
    * Lock both the pvt and the channel safely so that we can queue up a frame.
    */
   owner = sip_pvt_lock_full(p);
   if (owner) {
      ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner %s in place (Method: %s). Rescheduling destruction for 10000 ms\n", p->callid, ast_channel_name(owner), sip_methods[p->method].text);
      ast_queue_hangup_with_cause(owner, AST_CAUSE_PROTOCOL_ERROR);
      ast_channel_unlock(owner);
      ast_channel_unref(owner);
      sip_pvt_unlock(p);
      return 10000;
   } else if (p->refer && !p->alreadygone) {
      ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
      stop_media_flows(p);
      transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
      append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      append_history(p, "AutoDestroy", "%s", p->callid);
      ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
      sip_pvt_unlock(p);
      dialog_unlink_all(p); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
      sip_pvt_lock(p);
      /* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
      /* sip_destroy(p); */      /* Go ahead and destroy dialog. All attempts to recover is done */
      /* sip_destroy also absorbs the reference */
   }

   sip_pvt_unlock(p);

   dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");

   return 0;
}

void __sip_destroy	(	struct sip_pvt *	p,
		int	lockowner,
		int	lockdialoglist
	)

Execute destruction of SIP dialog structure, release memory.

Definition at line 6473 of file chan_sip.c.

References ao2_ref, ao2_t_ref, ast_callid_unref, ast_cc_config_params_destroy(), ast_channel_lock, ast_channel_name(), ast_channel_softhangup_internal_flag_add(), ast_channel_tech_pvt_set(), ast_channel_unlock, ast_debug, ast_format_cap_destroy(), ast_free, ast_free_acl_list(), AST_LIST_REMOVE_HEAD, ast_rtp_dtls_cfg_free(), ast_rtp_instance_destroy(), AST_SOFTHANGUP_DEV, ast_string_field_free_memory, ast_test_flag, ast_udptl_destroy(), ast_unref_namedgroups(), ast_variables_destroy(), ast_verbose(), ast_websocket_unref(), deinit_req(), destroy_msg_headers(), free_old_route(), offered_media_list_destroy(), registry_unref(), sip_debug_test_pvt(), sip_dump_history(), sip_methods, sip_srtp_destroy(), sip_unref_peer(), stop_session_timer(), cfsip_methods::text, and update_call_counter().

Referenced by sip_destroy().

{
   struct sip_request *req;

   /* Destroy Session-Timers if allocated */
   if (p->stimer) {
      p->stimer->quit_flag = 1;
      stop_session_timer(p);
      ast_free(p->stimer);
      p->stimer = NULL;
   }

   if (sip_debug_test_pvt(p))
      ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      update_call_counter(p, DEC_CALL_LIMIT);
      ast_debug(2, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
   }

   /* Unlink us from the owner if we have one */
   if (p->owner) {
      if (lockowner)
         ast_channel_lock(p->owner);
      ast_debug(1, "Detaching from %s\n", ast_channel_name(p->owner));
      ast_channel_tech_pvt_set(p->owner, NULL);
      /* Make sure that the channel knows its backend is going away */
      ast_channel_softhangup_internal_flag_add(p->owner, AST_SOFTHANGUP_DEV);
      if (lockowner)
         ast_channel_unlock(p->owner);
      /* Give the channel a chance to react before deallocation */
      usleep(1);
   }

   /* Remove link from peer to subscription of MWI */
   if (p->relatedpeer && p->relatedpeer->mwipvt == p)
      p->relatedpeer->mwipvt = dialog_unref(p->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   if (p->relatedpeer && p->relatedpeer->call == p)
      p->relatedpeer->call = dialog_unref(p->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   
   if (p->relatedpeer)
      p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting a dialog relatedpeer field in sip_destroy");
   
   if (p->registry) {
      if (p->registry->call == p)
         p->registry->call = dialog_unref(p->registry->call, "nulling out the registry's call dialog field in unlink_all");
      p->registry = registry_unref(p->registry, "delete p->registry");
   }
   
   if (p->mwi) {
      p->mwi->call = NULL;
      p->mwi = NULL;
   }

   if (dumphistory)
      sip_dump_history(p);

   if (p->options) {
      if (p->options->outboundproxy) {
         ao2_ref(p->options->outboundproxy, -1);
      }
      ast_free(p->options);
      p->options = NULL;
   }

   if (p->notify) {
      ast_variables_destroy(p->notify->headers);
      ast_free(p->notify->content);
      ast_free(p->notify);
      p->notify = NULL;
   }
   if (p->rtp) {
      ast_rtp_instance_destroy(p->rtp);
      p->rtp = NULL;
   }
   if (p->vrtp) {
      ast_rtp_instance_destroy(p->vrtp);
      p->vrtp = NULL;
   }
   if (p->trtp) {
      ast_rtp_instance_destroy(p->trtp);
      p->trtp = NULL;
   }
   if (p->udptl) {
      ast_udptl_destroy(p->udptl);
      p->udptl = NULL;
   }
   if (p->refer) {
      if (p->refer->refer_call) {
         p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
      }
      ast_string_field_free_memory(p->refer);
      ast_free(p->refer);
      p->refer = NULL;
   }
   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }
   deinit_req(&p->initreq);

   /* Clear history */
   if (p->history) {
      struct sip_history *hist;
      while ( (hist = AST_LIST_REMOVE_HEAD(p->history, list)) ) {
         ast_free(hist);
         p->history_entries--;
      }
      ast_free(p->history);
      p->history = NULL;
   }

   while ((req = AST_LIST_REMOVE_HEAD(&p->request_queue, next))) {
      ast_free(req);
   }

   offered_media_list_destroy(p);

   if (p->chanvars) {
      ast_variables_destroy(p->chanvars);
      p->chanvars = NULL;
   }

   destroy_msg_headers(p);

   if (p->srtp) {
      sip_srtp_destroy(p->srtp);
      p->srtp = NULL;
   }

   if (p->vsrtp) {
      sip_srtp_destroy(p->vsrtp);
      p->vsrtp = NULL;
   }

   if (p->tsrtp) {
      sip_srtp_destroy(p->tsrtp);
      p->tsrtp = NULL;
   }

   if (p->directmediaacl) {
      p->directmediaacl = ast_free_acl_list(p->directmediaacl);
   }

   ast_string_field_free_memory(p);

   ast_cc_config_params_destroy(p->cc_params);
   p->cc_params = NULL;

   if (p->epa_entry) {
      ao2_ref(p->epa_entry, -1);
      p->epa_entry = NULL;
   }

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   } else if (p->socket.ws_session) {
      ast_websocket_unref(p->socket.ws_session);
      p->socket.ws_session = NULL;
   }

   if (p->peerauth) {
      ao2_t_ref(p->peerauth, -1, "Removing active peer authentication");
      p->peerauth = NULL;
   }

   p->named_callgroups = ast_unref_namedgroups(p->named_callgroups);
   p->named_pickupgroups = ast_unref_namedgroups(p->named_pickupgroups);

   p->caps = ast_format_cap_destroy(p->caps);
   p->jointcaps = ast_format_cap_destroy(p->jointcaps);
   p->peercaps = ast_format_cap_destroy(p->peercaps);
   p->redircaps = ast_format_cap_destroy(p->redircaps);
   p->prefcaps = ast_format_cap_destroy(p->prefcaps);

   ast_rtp_dtls_cfg_free(&p->dtls_cfg);

   if (p->last_device_state_info) {
      ao2_ref(p->last_device_state_info, -1);
      p->last_device_state_info = NULL;
   }

   /* Lastly, kill the callid associated with the pvt */
   if (p->logger_callid) {
      ast_callid_unref(p->logger_callid);
   }
}

static int __sip_do_register

(

struct sip_registry *

r

)

[static]

Register with SIP proxy.

Returns:

see __sip_xmit

Definition at line 15119 of file chan_sip.c.

References transmit_register().

Referenced by sip_reregister().

{
   int res;

   res = transmit_register(r, SIP_REGISTER, NULL, NULL);
   return res;
}

void __sip_pretend_ack

(

struct sip_pvt *

p

)

Pretend to ack all packets called with p locked.

Definition at line 4574 of file chan_sip.c.

References __sip_ack(), ast_log(), ast_str_buffer(), find_sip_method(), LOG_WARNING, sip_methods, and cfsip_methods::text.

Referenced by __sip_autodestruct(), handle_request_bye(), handle_request_cancel(), and sip_reg_timeout().

{
   struct sip_pkt *cur = NULL;

   while (p->packets) {
      int method;
      if (cur == p->packets) {
         ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
         return;
      }
      cur = p->packets;
      method = (cur->method) ? cur->method : find_sip_method(ast_str_buffer(cur->data));
      __sip_ack(p, cur->seqno, cur->is_resp, method);
   }
}

static enum sip_result __sip_reliable_xmit	(	struct sip_pvt *	p,
		uint32_t	seqno,
		int	resp,
		struct ast_str *	data,
		int	fatal,
		int	sipmethod
	)		[static]

Todo:

According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited

Definition at line 4276 of file chan_sip.c.

References __sip_xmit(), append_history, ast_calloc, ast_debug, ast_free, ast_log(), AST_PTHREADT_NULL, AST_SCHED_DEL, AST_SCHED_REPLACE_VARIABLE, ast_str_buffer(), ast_str_create(), ast_str_set(), ast_str_strlen(), ast_tvnow(), DEFAULT_RETRANS, LOG_ERROR, and retrans_pkt().

Referenced by send_request(), and send_response().

{
   struct sip_pkt *pkt = NULL;
   int siptimer_a = DEFAULT_RETRANS;
   int xmitres = 0;
   int respid;

   if (sipmethod == SIP_INVITE) {
      /* Note this is a pending invite */
      p->pendinginvite = seqno;
   }

   /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
   /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
   /*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
   if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
      xmitres = __sip_xmit(p, data);   /* Send packet */
      if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
         append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
         return AST_FAILURE;
      } else {
         return AST_SUCCESS;
      }
   }

   if (!(pkt = ast_calloc(1, sizeof(*pkt)))) {
      return AST_FAILURE;
   }
   /* copy data, add a terminator and save length */
   if (!(pkt->data = ast_str_create(ast_str_strlen(data)))) {
      ast_free(pkt);
      return AST_FAILURE;
   }
   ast_str_set(&pkt->data, 0, "%s%s", ast_str_buffer(data), "\0");
   /* copy other parameters from the caller */
   pkt->method = sipmethod;
   pkt->seqno = seqno;
   pkt->is_resp = resp;
   pkt->is_fatal = fatal;
   pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
   pkt->next = p->packets;
   p->packets = pkt; /* Add it to the queue */
   if (resp) {
      /* Parse out the response code */
      if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
         pkt->response_code = respid;
      }
   }
   pkt->timer_t1 = p->timer_t1;  /* Set SIP timer T1 */
   pkt->retransid = -1;
   if (pkt->timer_t1) {
      siptimer_a = pkt->timer_t1;
   }

   pkt->time_sent = ast_tvnow(); /* time packet was sent */
   pkt->retrans_stop_time = 64 * (pkt->timer_t1 ? pkt->timer_t1 : DEFAULT_TIMER_T1); /* time in ms after pkt->time_sent to stop retransmission */

   /* Schedule retransmission */
   AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
   if (sipdebug) {
      ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);
   }

   xmitres = __sip_xmit(pkt->owner, pkt->data); /* Send packet */

   if (xmitres == XMIT_ERROR) {  /* Serious network trouble, no need to try again */
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
      ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
      AST_SCHED_DEL(sched, pkt->retransid);
      p->packets = pkt->next;
      pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
      ast_free(pkt->data);
      ast_free(pkt);
      return AST_FAILURE;
   } else {
      /* This is odd, but since the retrans timer starts at 500ms and the do_monitor thread
       * only wakes up every 1000ms by default, we have to poke the thread here to make
       * sure it successfully detects this must be retransmitted in less time than
       * it usually sleeps for. Otherwise it might not retransmit this packet for 1000ms. */
      if (monitor_thread != AST_PTHREADT_NULL) {
         pthread_kill(monitor_thread, SIGURG);
      }
      return AST_SUCCESS;
   }
}

int __sip_semi_ack	(	struct sip_pvt *	p,
		uint32_t	seqno,
		int	resp,
		int	sipmethod
	)

Acks receipt of packet, keep it around (used for provisional responses)

Definition at line 4591 of file chan_sip.c.

References ast_debug, AST_SCHED_DEL, ast_str_buffer(), FALSE, method_match(), sip_methods, cfsip_methods::text, and TRUE.

Referenced by handle_response(), and sip_hangup().

{
   struct sip_pkt *cur;
   int res = FALSE;

   for (cur = p->packets; cur; cur = cur->next) {
      if (cur->seqno == seqno && cur->is_resp == resp &&
         (cur->is_resp || method_match(sipmethod, ast_str_buffer(cur->data)))) {
         /* this is our baby */
         if (cur->retransid > -1) {
            if (sipdebug)
               ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
         }
         AST_SCHED_DEL(sched, cur->retransid);
         res = TRUE;
         break;
      }
   }
   ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %u: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
   return res;
}

static int __sip_subscribe_mwi_do

(

struct sip_subscription_mwi *

mwi

)

[static]

Actually setup an MWI subscription or resubscribe.

Definition at line 14342 of file chan_sip.c.

References ast_dnsmgr_lookup_cb(), ast_set_flag, ast_sip_ouraddrfor(), ast_sockaddr_port, ast_sockaddr_set_port, ast_string_field_set, ast_strlen_zero(), ASTOBJ_REF, ASTOBJ_UNREF, build_contact(), build_via(), change_callid_pvt(), create_addr(), dialog_unlink_all(), get_address_family_filter(), get_srv_protocol(), get_srv_service(), mwi_expiry, obproxy_get(), on_dns_update_mwi(), ref_proxy(), set_socket_transport(), sip_alloc(), sip_cfg, sip_subscribe_mwi_destroy(), and transmit_invite().

Referenced by sip_subscribe_mwi_do().

{
   /* If we have no DNS manager let's do a lookup */
   if (!mwi->dnsmgr) {
      char transport[MAXHOSTNAMELEN];
      struct sip_subscription_mwi *saved;
      snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(mwi->transport), get_srv_protocol(mwi->transport));

      mwi->us.ss.ss_family = get_address_family_filter(mwi->transport); /* Filter address family */
      saved = ASTOBJ_REF(mwi);
      ast_dnsmgr_lookup_cb(mwi->hostname, &mwi->us, &mwi->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_mwi, saved);
      if (!mwi->dnsmgr) {
         ASTOBJ_UNREF(saved, sip_subscribe_mwi_destroy); /* dnsmgr disabled, remove reference */
      }
   }

   /* If we already have a subscription up simply send a resubscription */
   if (mwi->call) {
      transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 0, NULL);
      return 0;
   }
   
   /* Create a dialog that we will use for the subscription */
   if (!(mwi->call = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, NULL))) {
      return -1;
   }

   ref_proxy(mwi->call, obproxy_get(mwi->call, NULL));

   if (!ast_sockaddr_port(&mwi->us) && mwi->portno) {
      ast_sockaddr_set_port(&mwi->us, mwi->portno);
   }
   
   /* Setup the destination of our subscription */
   if (create_addr(mwi->call, mwi->hostname, &mwi->us, 0)) {
      dialog_unlink_all(mwi->call);
      mwi->call = dialog_unref(mwi->call, "unref dialog after unlink_all");
      return 0;
   }

   mwi->call->expiry = mwi_expiry;
   
   if (!mwi->dnsmgr && mwi->portno) {
      ast_sockaddr_set_port(&mwi->call->sa, mwi->portno);
      ast_sockaddr_set_port(&mwi->call->recv, mwi->portno);
   } else {
      mwi->portno = ast_sockaddr_port(&mwi->call->sa);
   }
   
   /* Set various other information */
   if (!ast_strlen_zero(mwi->authuser)) {
      ast_string_field_set(mwi->call, peername, mwi->authuser);
      ast_string_field_set(mwi->call, authname, mwi->authuser);
      ast_string_field_set(mwi->call, fromuser, mwi->authuser);
   } else {
      ast_string_field_set(mwi->call, peername, mwi->username);
      ast_string_field_set(mwi->call, authname, mwi->username);
      ast_string_field_set(mwi->call, fromuser, mwi->username);
   }
   ast_string_field_set(mwi->call, username, mwi->username);
   if (!ast_strlen_zero(mwi->secret)) {
      ast_string_field_set(mwi->call, peersecret, mwi->secret);
   }
   set_socket_transport(&mwi->call->socket, mwi->transport);
   mwi->call->socket.port = htons(mwi->portno);
   ast_sip_ouraddrfor(&mwi->call->sa, &mwi->call->ourip, mwi->call);
   build_contact(mwi->call);
   build_via(mwi->call);

   /* Change the dialog callid. */
   change_callid_pvt(mwi->call, NULL);

   ast_set_flag(&mwi->call->flags[0], SIP_OUTGOING);
   
   /* Associate the call with us */
   mwi->call->mwi = ASTOBJ_REF(mwi);

   mwi->call->subscribed = MWI_NOTIFICATION;

   /* Actually send the packet */
   transmit_invite(mwi->call, SIP_SUBSCRIBE, 0, 2, NULL);

   return 0;
}

static int __sip_xmit	(	struct sip_pvt *	p,
		struct ast_str *	data
	)		[static]

Definition at line 3871 of file chan_sip.c.

References ast_debug, ast_log(), ast_sendto(), ast_sockaddr_stringify(), ast_str_buffer(), ast_str_strlen(), AST_WEBSOCKET_OPCODE_TEXT, ast_websocket_write(), errno, get_transport_pvt(), LOG_WARNING, sip_prepare_socket(), sip_real_dst(), and sip_tcptls_write().

Referenced by __sip_reliable_xmit(), retrans_pkt(), send_request(), and send_response().

{
   int res = 0;
   const struct ast_sockaddr *dst = sip_real_dst(p);

   ast_debug(2, "Trying to put '%.11s' onto %s socket destined for %s\n", ast_str_buffer(data), get_transport_pvt(p), ast_sockaddr_stringify(dst));

   if (sip_prepare_socket(p) < 0) {
      return XMIT_ERROR;
   }

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      res = ast_sendto(p->socket.fd, ast_str_buffer(data), ast_str_strlen(data), 0, dst);
   } else if (p->socket.tcptls_session) {
      res = sip_tcptls_write(p->socket.tcptls_session, ast_str_buffer(data), ast_str_strlen(data));
   } else if (p->socket.ws_session) {
      if (!(res = ast_websocket_write(p->socket.ws_session, AST_WEBSOCKET_OPCODE_TEXT, ast_str_buffer(data), ast_str_strlen(data)))) {
         /* The WebSocket API just returns 0 on success and -1 on failure, while this code expects the payload length to be returned */
         res = ast_str_strlen(data);
      }
   } else {
      ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
      return XMIT_ERROR;
   }

   if (res == -1) {
      switch (errno) {
      case EBADF:    /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
      case EHOSTUNREACH:   /* Host can't be reached */
      case ENETDOWN:    /* Interface down */
      case ENETUNREACH: /* Network failure */
      case ECONNREFUSED:      /* ICMP port unreachable */
         res = XMIT_ERROR; /* Don't bother with trying to transmit again */
      }
   }
   if (res != ast_str_strlen(data)) {
      ast_log(LOG_WARNING, "sip_xmit of %p (len %zu) to %s returned %d: %s\n", data, ast_str_strlen(data), ast_sockaddr_stringify(dst), res, strerror(errno));
   }

   return res;
}

static int __transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)		[static]

Base transmit response function.

Definition at line 12025 of file chan_sip.c.

References add_cc_call_info_to_response(), add_diversion(), add_header(), add_rpid(), ast_cause2str(), ast_channel_hangupcause(), ast_clear_flag, ast_log(), ast_test_flag, hangup_sip2cause(), LOG_WARNING, respprep(), send_response(), and sip_get_header().

Referenced by transmit_fake_auth_response(), transmit_response(), transmit_response_reliable(), and transmit_response_using_temp().

{
   struct sip_request resp;
   uint32_t seqno = 0;

   if (reliable && (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", sip_get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);

   if (ast_test_flag(&p->flags[0], SIP_SENDRPID)
         && ast_test_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND)
         && (!strncmp(msg, "180", 3) || !strncmp(msg, "183", 3))) {
      ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
      add_rpid(&resp, p);
   }
   if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
      add_cc_call_info_to_response(p, &resp);
   }

   /* If we are sending a 302 Redirect we can add a diversion header if the redirect information is set */
   if (!strncmp(msg, "302", 3)) {
      add_diversion(&resp, p);
   }

   /* If we are cancelling an incoming invite for some reason, add information
      about the reason why we are doing this in clear text */
   if (p->method == SIP_INVITE && msg[0] != '1') {
      char buf[20];

      if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON)) {
         int hangupcause = 0;

         if (p->owner && ast_channel_hangupcause(p->owner)) {
            hangupcause = ast_channel_hangupcause(p->owner);
         } else if (p->hangupcause) {
            hangupcause = p->hangupcause;
         } else {
            int respcode;
            if (sscanf(msg, "%30d ", &respcode))
               hangupcause = hangup_sip2cause(respcode);
         }

         if (hangupcause) {
            sprintf(buf, "Q.850;cause=%i", hangupcause & 0x7f);
            add_header(&resp, "Reason", buf);
         }
      }

      if (p->owner && ast_channel_hangupcause(p->owner)) {
         add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(ast_channel_hangupcause(p->owner)));
         snprintf(buf, sizeof(buf), "%d", ast_channel_hangupcause(p->owner));
         add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
      }
   }
   return send_response(p, &resp, reliable, seqno);
}

static void __unreg_module

(

void

)

[static]

Definition at line 34976 of file chan_sip.c.

static char * _sip_qualify_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)		[static]

Send qualify message to peer from cli or manager. Mostly for debugging.

Definition at line 19904 of file chan_sip.c.

References ast_cli(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, sip_find_peer(), sip_poke_peer(), sip_unref_peer(), and TRUE.

Referenced by manager_sip_qualify_peer(), and sip_qualify_peer().

{
   struct sip_peer *peer;
   int load_realtime;

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   if ((peer = sip_find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0))) {
      sip_poke_peer(peer, 1);
      sip_unref_peer(peer, "qualify: done with peer");
   } else if (type == 0) {
      ast_cli(fd, "Peer '%s' not found\n", argv[3]);
   } else {
      astman_send_error(s, m, "Peer not found");
   }
   return CLI_SUCCESS;
}

static char * _sip_show_peer	(	int	type,
		int	fd,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)		[static]

Show one peer in detail (main function)

Definition at line 19989 of file chan_sip.c.

References allowoverlap2str(), ao2_lock, ao2_t_ref, ao2_unlock, ARRAY_LEN, ast_acl_list_is_empty(), ast_callerid_merge(), ast_cdr_flags2str(), ast_check_realtime(), ast_cli(), AST_CLI_YESNO, ast_codec_pref_index(), AST_CODEC_PREF_SIZE, ast_describe_caller_presentation(), ast_getformatname(), ast_getformatname_multiple(), AST_LIST_TRAVERSE, ast_print_group(), ast_print_namedgroups(), ast_sched_when(), ast_sockaddr_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_str_alloca, ast_str_buffer(), ast_str_reset(), ast_strlen_zero(), ast_test_flag, astman_append(), astman_get_header(), astman_send_error(), CLI_SHOWUSAGE, CLI_SUCCESS, comedia_string(), dtmfmode2str(), FALSE, faxec2str(), force_rport_string(), get_transport_list(), insecure2str(), ast_variable::name, ast_variable::next, peer_mailboxes_to_str(), peer_status(), print_codec_to_cli(), print_group(), print_named_groups(), S_OR, sip_cfg, sip_find_peer(), sip_get_transport(), sip_unref_peer(), status, stmode2str(), strefresherparam2str(), text, transfermode2str(), TRUE, trust_id_outbound2str(), and ast_variable::value.

Referenced by manager_sip_show_peer(), and sip_show_peer().

{
   char status[30] = "";
   char cbuf[256];
   struct sip_peer *peer;
   char codec_buf[512];
   struct ast_codec_pref *pref;
   struct ast_variable *v;
   int x = 0, load_realtime;
   struct ast_format codec;
   int realtimepeers;

   realtimepeers = ast_check_realtime("sippeers");

   if (argc < 4)
      return CLI_SHOWUSAGE;

   load_realtime = (argc == 5 && !strcmp(argv[4], "load")) ? TRUE : FALSE;
   peer = sip_find_peer(argv[3], NULL, load_realtime, FINDPEERS, FALSE, 0);

   if (s) {    /* Manager */
      if (peer) {
         const char *id = astman_get_header(m, "ActionID");

         astman_append(s, "Response: Success\r\n");
         if (!ast_strlen_zero(id))
            astman_append(s, "ActionID: %s\r\n", id);
      } else {
         snprintf (cbuf, sizeof(cbuf), "Peer %s not found.", argv[3]);
         astman_send_error(s, m, cbuf);
         return CLI_SUCCESS;
      }
   }
   if (peer && type==0 ) { /* Normal listing */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      struct sip_auth_container *credentials;

      ao2_lock(peer);
      credentials = peer->auth;
      if (credentials) {
         ao2_t_ref(credentials, +1, "Ref peer auth for show");
      }
      ao2_unlock(peer);

      ast_cli(fd, "\n\n");
      ast_cli(fd, "  * Name       : %s\n", peer->name);
      ast_cli(fd, "  Description  : %s\n", peer->description);
      if (realtimepeers) { /* Realtime is enabled */
         ast_cli(fd, "  Realtime peer: %s\n", peer->is_realtime ? "Yes, cached" : "No");
      }
      ast_cli(fd, "  Secret       : %s\n", ast_strlen_zero(peer->secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  MD5Secret    : %s\n", ast_strlen_zero(peer->md5secret)?"<Not set>":"<Set>");
      ast_cli(fd, "  Remote Secret: %s\n", ast_strlen_zero(peer->remotesecret)?"<Not set>":"<Set>");
      if (credentials) {
         struct sip_auth *auth;

         AST_LIST_TRAVERSE(&credentials->list, auth, node) {
            ast_cli(fd, "  Realm-auth   : Realm %-15.15s User %-10.20s %s\n",
               auth->realm,
               auth->username,
               !ast_strlen_zero(auth->secret)
                  ? "<Secret set>"
                  : (!ast_strlen_zero(auth->md5secret)
                     ? "<MD5secret set>" : "<Not set>"));
         }
         ao2_t_ref(credentials, -1, "Unref peer auth for show");
      }
      ast_cli(fd, "  Context      : %s\n", peer->context);
      ast_cli(fd, "  Record On feature : %s\n", peer->record_on_feature);
      ast_cli(fd, "  Record Off feature : %s\n", peer->record_off_feature);
      ast_cli(fd, "  Subscr.Cont. : %s\n", S_OR(peer->subscribecontext, "<Not set>") );
      ast_cli(fd, "  Language     : %s\n", peer->language);
      ast_cli(fd, "  Tonezone     : %s\n", peer->zone[0] != '\0' ? peer->zone : "<Not set>");
      if (!ast_strlen_zero(peer->accountcode))
         ast_cli(fd, "  Accountcode  : %s\n", peer->accountcode);
      ast_cli(fd, "  AMA flags    : %s\n", ast_cdr_flags2str(peer->amaflags));
      ast_cli(fd, "  Transfer mode: %s\n", transfermode2str(peer->allowtransfer));
      ast_cli(fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         ast_cli(fd, "  FromUser     : %s\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         ast_cli(fd, "  FromDomain   : %s Port %d\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
      ast_cli(fd, "  Callgroup    : ");
      print_group(fd, peer->callgroup, 0);
      ast_cli(fd, "  Pickupgroup  : ");
      print_group(fd, peer->pickupgroup, 0);
      ast_cli(fd, "  Named Callgr : ");
      print_named_groups(fd, peer->named_callgroups, 0);
      ast_cli(fd, "  Nam. Pickupgr: ");
      print_named_groups(fd, peer->named_pickupgroups, 0);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_cli(fd, "  MOH Suggest  : %s\n", peer->mohsuggest);
      ast_cli(fd, "  Mailbox      : %s\n", ast_str_buffer(mailbox_str));
      ast_cli(fd, "  VM Extension : %s\n", peer->vmexten);
      ast_cli(fd, "  LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
      ast_cli(fd, "  Call limit   : %d\n", peer->call_limit);
      ast_cli(fd, "  Max forwards : %d\n", peer->maxforwards);
      if (peer->busy_level)
         ast_cli(fd, "  Busy level   : %d\n", peer->busy_level);
      ast_cli(fd, "  Dynamic      : %s\n", AST_CLI_YESNO(peer->host_dynamic));
      ast_cli(fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, "<unspecified>"));
      ast_cli(fd, "  MaxCallBR    : %d kbps\n", peer->maxcallbitrate);
      ast_cli(fd, "  Expire       : %ld\n", ast_sched_when(sched, peer->expire));
      ast_cli(fd, "  Insecure     : %s\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      ast_cli(fd, "  Force rport  : %s\n", force_rport_string(peer->flags));
      ast_cli(fd, "  Symmetric RTP: %s\n", comedia_string(peer->flags));
      ast_cli(fd, "  ACL          : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->acl) == 0));
      ast_cli(fd, "  DirectMedACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->directmediaacl) == 0));
      ast_cli(fd, "  T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 EC mode : %s\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      ast_cli(fd, "  T.38 MaxDtgrm: %d\n", peer->t38_maxdatagram);
      ast_cli(fd, "  DirectMedia  : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
      ast_cli(fd, "  PromiscRedir : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
      ast_cli(fd, "  User=Phone   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
      ast_cli(fd, "  Video Support: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) || ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS)));
      ast_cli(fd, "  Text Support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)));
      ast_cli(fd, "  Ign SDP ver  : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_IGNORESDPVERSION)));
      ast_cli(fd, "  Trust RPID   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_TRUSTRPID)));
      ast_cli(fd, "  Send RPID    : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_SENDRPID)));
      ast_cli(fd, "  TrustIDOutbnd: %s\n", trust_id_outbound2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)));
      ast_cli(fd, "  Subscriptions: %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
      ast_cli(fd, "  Overlap dial : %s\n", allowoverlap2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWOVERLAP)));
      if (peer->outboundproxy)
         ast_cli(fd, "  Outb. proxy  : %s %s\n", ast_strlen_zero(peer->outboundproxy->name) ? "<not set>" : peer->outboundproxy->name,
                     peer->outboundproxy->force ? "(forced)" : "");

      /* - is enumerated */
      ast_cli(fd, "  DTMFmode     : %s\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      ast_cli(fd, "  Timer T1     : %d\n", peer->timer_t1);
      ast_cli(fd, "  Timer B      : %d\n", peer->timer_b);
      ast_cli(fd, "  ToHost       : %s\n", peer->tohost);
      ast_cli(fd, "  Addr->IP     : %s\n", ast_sockaddr_stringify(&peer->addr));
      ast_cli(fd, "  Defaddr->IP  : %s\n", ast_sockaddr_stringify(&peer->defaddr));
      ast_cli(fd, "  Prim.Transp. : %s\n", sip_get_transport(peer->socket.type));
      ast_cli(fd, "  Allowed.Trsp : %s\n", get_transport_list(peer->transports));
      if (!ast_strlen_zero(sip_cfg.regcontext))
         ast_cli(fd, "  Reg. exten   : %s\n", peer->regexten);
      ast_cli(fd, "  Def. Username: %s\n", peer->username);
      ast_cli(fd, "  SIP Options  : ");
      if (peer->sipoptions) {
         int lastoption = -1;
         for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
            if (sip_options[x].id != lastoption) {
               if (peer->sipoptions & sip_options[x].id)
                  ast_cli(fd, "%s ", sip_options[x].text);
               lastoption = x;
            }
         }
      } else
         ast_cli(fd, "(none)");

      ast_cli(fd, "\n");
      ast_cli(fd, "  Codecs       : ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->caps);
      ast_cli(fd, "%s\n", codec_buf);
      ast_cli(fd, "  Codec Order  : (");
      print_codec_to_cli(fd, &peer->prefs);
      ast_cli(fd, ")\n");

      ast_cli(fd, "  Auto-Framing : %s\n", AST_CLI_YESNO(peer->autoframing));
      ast_cli(fd, "  Status       : ");
      peer_status(peer, status, sizeof(status));
      ast_cli(fd, "%s\n", status);
      ast_cli(fd, "  Useragent    : %s\n", peer->useragent);
      ast_cli(fd, "  Reg. Contact : %s\n", peer->fullcontact);
      ast_cli(fd, "  Qualify Freq : %d ms\n", peer->qualifyfreq);
      ast_cli(fd, "  Keepalive    : %d ms\n", peer->keepalive * 1000);
      if (peer->chanvars) {
         ast_cli(fd, "  Variables    :\n");
         for (v = peer->chanvars ; v ; v = v->next)
            ast_cli(fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(fd, "  Sess-Timers  : %s\n", stmode2str(peer->stimer.st_mode_oper));
      ast_cli(fd, "  Sess-Refresh : %s\n", strefresherparam2str(peer->stimer.st_ref));
      ast_cli(fd, "  Sess-Expires : %d secs\n", peer->stimer.st_max_se);
      ast_cli(fd, "  Min-Sess     : %d secs\n", peer->stimer.st_min_se);
      ast_cli(fd, "  RTP Engine   : %s\n", peer->engine);
      ast_cli(fd, "  Parkinglot   : %s\n", peer->parkinglot);
      ast_cli(fd, "  Use Reason   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)));
      ast_cli(fd, "  Encryption   : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP)));
      ast_cli(fd, "\n");
      peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer ptr");
   } else  if (peer && type == 1) { /* manager listing */
      char buffer[256];
      struct ast_str *tmp_str = ast_str_alloca(512);
      astman_append(s, "Channeltype: SIP\r\n");
      astman_append(s, "ObjectName: %s\r\n", peer->name);
      astman_append(s, "ChanObjectType: peer\r\n");
      astman_append(s, "SecretExist: %s\r\n", ast_strlen_zero(peer->secret)?"N":"Y");
      astman_append(s, "RemoteSecretExist: %s\r\n", ast_strlen_zero(peer->remotesecret)?"N":"Y");
      astman_append(s, "MD5SecretExist: %s\r\n", ast_strlen_zero(peer->md5secret)?"N":"Y");
      astman_append(s, "Context: %s\r\n", peer->context);
      astman_append(s, "Language: %s\r\n", peer->language);
      astman_append(s, "ToneZone: %s\r\n", peer->zone[0] != '\0' ? peer->zone : "<Not set>");
      if (!ast_strlen_zero(peer->accountcode))
         astman_append(s, "Accountcode: %s\r\n", peer->accountcode);
      astman_append(s, "AMAflags: %s\r\n", ast_cdr_flags2str(peer->amaflags));
      astman_append(s, "CID-CallingPres: %s\r\n", ast_describe_caller_presentation(peer->callingpres));
      if (!ast_strlen_zero(peer->fromuser))
         astman_append(s, "SIP-FromUser: %s\r\n", peer->fromuser);
      if (!ast_strlen_zero(peer->fromdomain))
         astman_append(s, "SIP-FromDomain: %s\r\nSip-FromDomain-Port: %d\r\n", peer->fromdomain, (peer->fromdomainport) ? peer->fromdomainport : STANDARD_SIP_PORT);
      astman_append(s, "Callgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->callgroup));
      astman_append(s, "Pickupgroup: ");
      astman_append(s, "%s\r\n", ast_print_group(buffer, sizeof(buffer), peer->pickupgroup));
      astman_append(s, "Named Callgroup: ");
      astman_append(s, "%s\r\n", ast_print_namedgroups(&tmp_str, peer->named_callgroups));
      ast_str_reset(tmp_str);
      astman_append(s, "Named Pickupgroup: ");
      astman_append(s, "%s\r\n", ast_print_namedgroups(&tmp_str, peer->named_pickupgroups));
      ast_str_reset(tmp_str);
      astman_append(s, "MOHSuggest: %s\r\n", peer->mohsuggest);
      peer_mailboxes_to_str(&tmp_str, peer);
      astman_append(s, "VoiceMailbox: %s\r\n", ast_str_buffer(tmp_str));
      astman_append(s, "TransferMode: %s\r\n", transfermode2str(peer->allowtransfer));
      astman_append(s, "LastMsgsSent: %d\r\n", peer->lastmsgssent);
      astman_append(s, "Maxforwards: %d\r\n", peer->maxforwards);
      astman_append(s, "Call-limit: %d\r\n", peer->call_limit);
      astman_append(s, "Busy-level: %d\r\n", peer->busy_level);
      astman_append(s, "MaxCallBR: %d kbps\r\n", peer->maxcallbitrate);
      astman_append(s, "Dynamic: %s\r\n", peer->host_dynamic?"Y":"N");
      astman_append(s, "Callerid: %s\r\n", ast_callerid_merge(cbuf, sizeof(cbuf), peer->cid_name, peer->cid_num, ""));
      astman_append(s, "RegExpire: %ld seconds\r\n", ast_sched_when(sched, peer->expire));
      astman_append(s, "SIP-AuthInsecure: %s\r\n", insecure2str(ast_test_flag(&peer->flags[0], SIP_INSECURE)));
      astman_append(s, "SIP-Forcerport: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ?
            (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "A" : "a") :
            (ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "Y" : "N"));
      astman_append(s, "SIP-Comedia: %s\r\n", ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ?
            (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "A" : "a") :
            (ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "Y" : "N"));
      astman_append(s, "ACL: %s\r\n", (ast_acl_list_is_empty(peer->acl) ? "N" : "Y"));
      astman_append(s, "SIP-CanReinvite: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-DirectMedia: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)?"Y":"N"));
      astman_append(s, "SIP-PromiscRedir: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)?"Y":"N"));
      astman_append(s, "SIP-UserPhone: %s\r\n", (ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)?"Y":"N"));
      astman_append(s, "SIP-VideoSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38Support: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)?"Y":"N"));
      astman_append(s, "SIP-T.38EC: %s\r\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
      astman_append(s, "SIP-T.38MaxDtgrm: %d\r\n", peer->t38_maxdatagram);
      astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
      astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresherparam2str(peer->stimer.st_ref));
      astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
      astman_append(s, "SIP-Sess-Min: %d\r\n", peer->stimer.st_min_se);
      astman_append(s, "SIP-RTP-Engine: %s\r\n", peer->engine);
      astman_append(s, "SIP-Encryption: %s\r\n", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP) ? "Y" : "N");

      /* - is enumerated */
      astman_append(s, "SIP-DTMFmode: %s\r\n", dtmfmode2str(ast_test_flag(&peer->flags[0], SIP_DTMF)));
      astman_append(s, "ToHost: %s\r\n", peer->tohost);
      astman_append(s, "Address-IP: %s\r\nAddress-Port: %d\r\n", ast_sockaddr_stringify_addr(&peer->addr), ast_sockaddr_port(&peer->addr));
      astman_append(s, "Default-addr-IP: %s\r\nDefault-addr-port: %d\r\n", ast_sockaddr_stringify_addr(&peer->defaddr), ast_sockaddr_port(&peer->defaddr));
      astman_append(s, "Default-Username: %s\r\n", peer->username);
      if (!ast_strlen_zero(sip_cfg.regcontext))
         astman_append(s, "RegExtension: %s\r\n", peer->regexten);
      astman_append(s, "Codecs: ");
      ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, peer->caps);
      astman_append(s, "%s\r\n", codec_buf);
      astman_append(s, "CodecOrder: ");
      pref = &peer->prefs;
      for(x = 0; x < AST_CODEC_PREF_SIZE ; x++) {
         if (!(ast_codec_pref_index(pref, x, &codec))) {
            break;
         }
         astman_append(s, "%s", ast_getformatname(&codec));
         if ((x < (AST_CODEC_PREF_SIZE - 1)) && ast_codec_pref_index(pref, x+1, &codec))
            astman_append(s, ",");
      }

      astman_append(s, "\r\n");
      astman_append(s, "Status: ");
      peer_status(peer, status, sizeof(status));
      astman_append(s, "%s\r\n", status);
      astman_append(s, "SIP-Useragent: %s\r\n", peer->useragent);
      astman_append(s, "Reg-Contact: %s\r\n", peer->fullcontact);
      astman_append(s, "QualifyFreq: %d ms\r\n", peer->qualifyfreq);
      astman_append(s, "Parkinglot: %s\r\n", peer->parkinglot);
      if (peer->chanvars) {
         for (v = peer->chanvars ; v ; v = v->next) {
            astman_append(s, "ChanVariable: %s=%s\r\n", v->name, v->value);
         }
      }
      astman_append(s, "SIP-Use-Reason-Header: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_Q850_REASON)) ? "Y" : "N");
      astman_append(s, "Description: %s\r\n", peer->description);

      peer = sip_unref_peer(peer, "sip_show_peer: sip_unref_peer: done with peer");

   } else {
      ast_cli(fd, "Peer %s not found.\n", argv[3]);
      ast_cli(fd, "\n");
   }

   return CLI_SUCCESS;
}

static char * _sip_show_peers	(	int	fd,
		int *	total,
		struct mansession *	s,
		const struct message *	m,
		int	argc,
		const char *	argv[]
	)		[static]

Execute sip show peers command.

Definition at line 19129 of file chan_sip.c.

References _sip_show_peers_one(), ao2_callback, ao2_container_count(), ao2_iterator_destroy(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_calloc, ast_check_realtime(), ast_cli(), ast_free, ast_log(), AST_LOG_ERROR, ast_strlen_zero(), astman_get_header(), CLI_FAILURE, CLI_SHOWUSAGE, CLI_SUCCESS, FALSE, show_peers_context::havepattern, id, show_peers_context::idtext, OBJ_MULTIPLE, peercomparefunc(), PEERS_FORMAT2, show_peers_context::peers_mon_offline, show_peers_context::peers_mon_online, show_peers_context::peers_unmon_offline, show_peers_context::peers_unmon_online, show_peers_context::realtimepeers, show_peers_context::regexbuf, sip_unref_peer(), and TRUE.

Referenced by manager_sip_show_peers(), and sip_show_peers().

{
   struct show_peers_context cont = {
      .havepattern = FALSE,
      .idtext = "",

      .peers_mon_online = 0,
      .peers_mon_offline = 0,
      .peers_unmon_online = 0,
      .peers_unmon_offline = 0,
   };

   struct sip_peer *peer;
   struct ao2_iterator* it_peers;

   int total_peers = 0;
   const char *id;
   struct sip_peer **peerarray;
   int k;

   cont.realtimepeers = ast_check_realtime("sippeers");

   if (s) { /* Manager - get ActionID */
      id = astman_get_header(m, "ActionID");
      if (!ast_strlen_zero(id)) {
         snprintf(cont.idtext, sizeof(cont.idtext), "ActionID: %s\r\n", id);
      }
   }

   switch (argc) {
   case 5:
      if (!strcasecmp(argv[3], "like")) {
         if (regcomp(&cont.regexbuf, argv[4], REG_EXTENDED | REG_NOSUB)) {
            return CLI_SHOWUSAGE;
         }
         cont.havepattern = TRUE;
      } else {
         return CLI_SHOWUSAGE;
      }
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (!s) {
      /* Normal list */
      ast_cli(fd, PEERS_FORMAT2, "Name/username", "Host", "Dyn", "Forcerport", "Comedia", "ACL", "Port", "Status", "Description", (cont.realtimepeers ? "Realtime" : ""));
   }

   ao2_lock(peers);
   if (!(it_peers = ao2_callback(peers, OBJ_MULTIPLE, NULL, NULL))) {
      ast_log(AST_LOG_ERROR, "Unable to create iterator for peers container for sip show peers\n");
      ao2_unlock(peers);
      return CLI_FAILURE;
   }
   if (!(peerarray = ast_calloc(sizeof(struct sip_peer *), ao2_container_count(peers)))) {
      ast_log(AST_LOG_ERROR, "Unable to allocate peer array for sip show peers\n");
      ao2_iterator_destroy(it_peers);
      ao2_unlock(peers);
      return CLI_FAILURE;
   }
   ao2_unlock(peers);

   while ((peer = ao2_t_iterator_next(it_peers, "iterate thru peers table"))) {
      ao2_lock(peer);

      if (!(peer->type & SIP_TYPE_PEER)) {
         ao2_unlock(peer);
         sip_unref_peer(peer, "unref peer because it's actually a user");
         continue;
      }

      if (cont.havepattern && regexec(&cont.regexbuf, peer->name, 0, NULL, 0)) {
         ao2_unlock(peer);
         sip_unref_peer(peer, "toss iterator peer ptr before continue");
         continue;
      }

      peerarray[total_peers++] = peer;
      ao2_unlock(peer);
   }
   ao2_iterator_destroy(it_peers);

   qsort(peerarray, total_peers, sizeof(struct sip_peer *), peercomparefunc);

   for(k = 0; k < total_peers; k++) {
      peerarray[k] = _sip_show_peers_one(fd, s, &cont, peerarray[k]);
   }

   if (!s) {
      ast_cli(fd, "%d sip peers [Monitored: %d online, %d offline Unmonitored: %d online, %d offline]\n",
              total_peers, cont.peers_mon_online, cont.peers_mon_offline, cont.peers_unmon_online, cont.peers_unmon_offline);
   }

   if (cont.havepattern) {
      regfree(&cont.regexbuf);
   }

   if (total) {
      *total = total_peers;
   }

   ast_free(peerarray);

   return CLI_SUCCESS;
}

static struct sip_peer * _sip_show_peers_one	(	int	fd,
		struct mansession *	s,
		struct show_peers_context *	cont,
		struct sip_peer *	peer
	)		[static, read]

Emit informations for one peer during sip show peers command.

Definition at line 19238 of file chan_sip.c.

References ao2_lock, ao2_unlock, ast_acl_list_is_empty(), ast_cli(), ast_copy_string(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), ast_strlen_zero(), ast_test_flag, astman_append(), comedia_string(), force_rport_string(), show_peers_context::havepattern, show_peers_context::idtext, name, peer_status(), PEERS_FORMAT2, show_peers_context::peers_mon_offline, show_peers_context::peers_mon_online, show_peers_context::peers_unmon_offline, show_peers_context::peers_unmon_online, show_peers_context::realtimepeers, show_peers_context::regexbuf, sip_unref_peer(), and status.

Referenced by _sip_show_peers().

{
   /* _sip_show_peers_one() is separated from _sip_show_peers() to properly free the ast_strdupa
    * (this is executed in a loop in _sip_show_peers() )
    */

   char name[256];
   char status[20] = "";
   char pstatus;

   /*
    * tmp_port and tmp_host store copies of ast_sockaddr_stringify strings since the
    * string pointers for that function aren't valid between subsequent calls to
    * ast_sockaddr_stringify functions
    */
   char *tmp_port;
   char *tmp_host;

   tmp_port = ast_sockaddr_isnull(&peer->addr) ?
      "0" : ast_strdupa(ast_sockaddr_stringify_port(&peer->addr));

   tmp_host = ast_sockaddr_isnull(&peer->addr) ?
      "(Unspecified)" : ast_strdupa(ast_sockaddr_stringify_addr(&peer->addr));

   ao2_lock(peer);
   if (cont->havepattern && regexec(&cont->regexbuf, peer->name, 0, NULL, 0)) {
      ao2_unlock(peer);
      return sip_unref_peer(peer, "toss iterator peer ptr no match");
   }

   if (!ast_strlen_zero(peer->username) && !s) {
      snprintf(name, sizeof(name), "%s/%s", peer->name, peer->username);
   } else {
      ast_copy_string(name, peer->name, sizeof(name));
   }

   pstatus = peer_status(peer, status, sizeof(status));
   if (pstatus == 1) {
      cont->peers_mon_online++;
   } else if (pstatus == 0) {
      cont->peers_mon_offline++;
   } else {
      if (ast_sockaddr_isnull(&peer->addr) ||
          !ast_sockaddr_port(&peer->addr)) {
         cont->peers_unmon_offline++;
      } else {
         cont->peers_unmon_online++;
      }
   }

   if (!s) { /* Normal CLI list */
      ast_cli(fd, PEERS_FORMAT2, name,
      tmp_host,
      peer->host_dynamic ? " D " : "   ", /* Dynamic or not? */
      force_rport_string(peer->flags),
      comedia_string(peer->flags),
      (!ast_acl_list_is_empty(peer->acl)) ? " A " : "   ",       /* permit/deny */
      tmp_port, status,
      peer->description ? peer->description : "",
      cont->realtimepeers ? (peer->is_realtime ? "Cached RT" : "") : "");
   } else { /* Manager format */
      /* The names here need to be the same as other channels */
      astman_append(s,
      "Event: PeerEntry\r\n%s"
      "Channeltype: SIP\r\n"
      "ObjectName: %s\r\n"
      "ChanObjectType: peer\r\n" /* "peer" or "user" */
      "IPaddress: %s\r\n"
      "IPport: %s\r\n"
      "Dynamic: %s\r\n"
      "AutoForcerport: %s\r\n"
      "Forcerport: %s\r\n"
      "AutoComedia: %s\r\n"
      "Comedia: %s\r\n"
      "VideoSupport: %s\r\n"
      "TextSupport: %s\r\n"
      "ACL: %s\r\n"
      "Status: %s\r\n"
      "RealtimeDevice: %s\r\n"
      "Description: %s\r\n\r\n",
      cont->idtext,
      peer->name,
      ast_sockaddr_isnull(&peer->addr) ? "-none-" : tmp_host,
      ast_sockaddr_isnull(&peer->addr) ? "0" : tmp_port,
      peer->host_dynamic ? "yes" : "no",  /* Dynamic or not? */
      ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) ? "yes" : "no",
      ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) ? "yes" : "no",
      ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA) ? "yes" : "no",
      ast_test_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP) ? "yes" : "no",
      ast_test_flag(&peer->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "yes" : "no",  /* VIDEOSUPPORT=yes? */
      ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT) ? "yes" : "no",   /* TEXTSUPPORT=yes? */
      ast_acl_list_is_empty(peer->acl) ? "no" : "yes",       /* permit/deny/acl */
      status,
      cont->realtimepeers ? (peer->is_realtime ? "yes" : "no") : "no",
      peer->description);
   }
   ao2_unlock(peer);

   return sip_unref_peer(peer, "toss iterator peer ptr");
}

static void * _sip_tcp_helper_thread

(

struct ast_tcptls_session_instance *

tcptls_session

)

[static]

SIP TCP thread management function This function reads from the socket, parses the packet into a request.

Definition at line 3034 of file chan_sip.c.

References ao2_lock, ao2_ref, ao2_t_find, ao2_t_ref, ao2_t_unlink, ao2_unlock, ast_atomic_fetchadd_int(), ast_debug, AST_LIST_REMOVE_HEAD, ast_log(), ast_poll, ast_str_buffer(), ast_str_create(), ast_str_reset(), ast_str_strlen(), ast_tcptls_client_start(), ast_tcptls_close_session_file(), ast_tcptls_server_write(), cleanup(), ast_tcptls_session_instance::client, deinit_req(), errno, ast_tcptls_session_instance::fd, handle_request_do(), LOG_ERROR, LOG_WARNING, OBJ_POINTER, ast_tcptls_session_instance::overflow_buf, ast_tcptls_session_instance::parent, ast_tcptls_session_instance::remote_address, set_socket_transport(), sip_check_authtimeout(), sip_tcp_read(), sip_threadinfo_create(), sip_tls_read(), and ast_tcptls_session_instance::ssl.

Referenced by sip_tcp_worker_fn().

{
   int res, timeout = -1, authenticated = 0, flags;
   time_t start;
   struct sip_request req = { 0, } , reqcpy = { 0, };
   struct sip_threadinfo *me = NULL;
   char buf[1024] = "";
   struct pollfd fds[2] = { { 0 }, { 0 }, };
   struct ast_tcptls_session_args *ca = NULL;

   /* If this is a server session, then the connection has already been
    * setup. Check if the authlimit has been reached and if not create the
    * threadinfo object so we can access this thread for writing.
    *
    * if this is a client connection more work must be done.
    * 1. We own the parent session args for a client connection.  This pointer needs
    *    to be held on to so we can decrement it's ref count on thread destruction.
    * 2. The threadinfo object was created before this thread was launched, however
    *    it must be found within the threadt table.
    * 3. Last, the tcptls_session must be started.
    */
   if (!tcptls_session->client) {
      if (ast_atomic_fetchadd_int(&unauth_sessions, +1) >= authlimit) {
         /* unauth_sessions is decremented in the cleanup code */
         goto cleanup;
      }

      if ((flags = fcntl(tcptls_session->fd, F_GETFL)) == -1) {
         ast_log(LOG_ERROR, "error setting socket to non blocking mode, fcntl() failed: %s\n", strerror(errno));
         goto cleanup;
      }

      flags |= O_NONBLOCK;
      if (fcntl(tcptls_session->fd, F_SETFL, flags) == -1) {
         ast_log(LOG_ERROR, "error setting socket to non blocking mode, fcntl() failed: %s\n", strerror(errno));
         goto cleanup;
      }

      if (!(me = sip_threadinfo_create(tcptls_session, tcptls_session->ssl ? SIP_TRANSPORT_TLS : SIP_TRANSPORT_TCP))) {
         goto cleanup;
      }
      ao2_t_ref(me, +1, "Adding threadinfo ref for tcp_helper_thread");
   } else {
      struct sip_threadinfo tmp = {
         .tcptls_session = tcptls_session,
      };

      if ((!(ca = tcptls_session->parent)) ||
         (!(me = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread"))) ||
         (!(tcptls_session = ast_tcptls_client_start(tcptls_session)))) {
         goto cleanup;
      }
   }

   flags = 1;
   if (setsockopt(tcptls_session->fd, SOL_SOCKET, SO_KEEPALIVE, &flags, sizeof(flags))) {
      ast_log(LOG_ERROR, "error enabling TCP keep-alives on sip socket: %s\n", strerror(errno));
      goto cleanup;
   }

   me->threadid = pthread_self();
   ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "TLS" : "TCP");

   /* set up pollfd to watch for reads on both the socket and the alert_pipe */
   fds[0].fd = tcptls_session->fd;
   fds[1].fd = me->alert_pipe[0];
   fds[0].events = fds[1].events = POLLIN | POLLPRI;

   if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
      goto cleanup;
   }
   if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET))) {
      goto cleanup;
   }

   if(time(&start) == -1) {
      ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
      goto cleanup;
   }

   for (;;) {
      struct ast_str *str_save;

      if (!tcptls_session->client && req.authenticated && !authenticated) {
         authenticated = 1;
         ast_atomic_fetchadd_int(&unauth_sessions, -1);
      }

      /* calculate the timeout for unauthenticated server sessions */
      if (!tcptls_session->client && !authenticated ) {
         if ((timeout = sip_check_authtimeout(start)) < 0) {
            goto cleanup;
         }

         if (timeout == 0) {
            ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "TLS": "TCP");
            goto cleanup;
         }
      } else {
         timeout = -1;
      }

      if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
         res = ast_poll(fds, 2, timeout); /* polls for both socket and alert_pipe */
         if (res < 0) {
            ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "TLS": "TCP", res);
            goto cleanup;
         } else if (res == 0) {
            /* timeout */
            ast_debug(2, "SIP %s server timed out\n", tcptls_session->ssl ? "TLS": "TCP");
            goto cleanup;
         }
      }

      /* 
       * handle the socket event, check for both reads from the socket fd or TCP overflow buffer,
       * and writes from alert_pipe fd.
       */
      if (fds[0].revents || (ast_str_strlen(tcptls_session->overflow_buf) > 0)) { /* there is data on the socket to be read */
         fds[0].revents = 0;

         /* clear request structure */
         str_save = req.data;
         memset(&req, 0, sizeof(req));
         req.data = str_save;
         ast_str_reset(req.data);

         str_save = reqcpy.data;
         memset(&reqcpy, 0, sizeof(reqcpy));
         reqcpy.data = str_save;
         ast_str_reset(reqcpy.data);

         memset(buf, 0, sizeof(buf));

         if (tcptls_session->ssl) {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TLS);
            req.socket.port = htons(ourport_tls);
         } else {
            set_socket_transport(&req.socket, SIP_TRANSPORT_TCP);
            req.socket.port = htons(ourport_tcp);
         }
         req.socket.fd = tcptls_session->fd;
         if (tcptls_session->ssl) {
            res = sip_tls_read(&req, &reqcpy, tcptls_session, authenticated, start, me);
         } else {
            res = sip_tcp_read(&req, tcptls_session, authenticated, start);
         }

         if (res < 0) {
            goto cleanup;
         }

         req.socket.tcptls_session = tcptls_session;
         req.socket.ws_session = NULL;
         handle_request_do(&req, &tcptls_session->remote_address);
      }

      if (fds[1].revents) { /* alert_pipe indicates there is data in the send queue to be sent */
         enum sip_tcptls_alert alert;
         struct tcptls_packet *packet;

         fds[1].revents = 0;

         if (read(me->alert_pipe[0], &alert, sizeof(alert)) == -1) {
            ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
            continue;
         }

         switch (alert) {
         case TCPTLS_ALERT_STOP:
            goto cleanup;
         case TCPTLS_ALERT_DATA:
            ao2_lock(me);
            if (!(packet = AST_LIST_REMOVE_HEAD(&me->packet_q, entry))) {
               ast_log(LOG_WARNING, "TCPTLS thread alert_pipe indicated packet should be sent, but frame_q is empty\n");
            }
            ao2_unlock(me);

            if (packet) {
               if (ast_tcptls_server_write(tcptls_session, ast_str_buffer(packet->data), packet->len) == -1) {
                  ast_log(LOG_WARNING, "Failure to write to tcp/tls socket\n");
               }
               ao2_t_ref(packet, -1, "tcptls packet sent, this is no longer needed");
            }
            break;
         default:
            ast_log(LOG_ERROR, "Unknown tcptls thread alert '%d'\n", alert);
         }
      }
   }

   ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "TLS" : "TCP");

cleanup:
   if (tcptls_session && !tcptls_session->client && !authenticated) {
      ast_atomic_fetchadd_int(&unauth_sessions, -1);
   }

   if (me) {
      ao2_t_unlink(threadt, me, "Removing tcptls helper thread, thread is closing");
      ao2_t_ref(me, -1, "Removing tcp_helper_threads threadinfo ref");
   }
   deinit_req(&reqcpy);
   deinit_req(&req);

   /* if client, we own the parent session arguments and must decrement ref */
   if (ca) {
      ao2_t_ref(ca, -1, "closing tcptls thread, getting rid of client tcptls_session arguments");
   }

   if (tcptls_session) {
      ao2_lock(tcptls_session);
      ast_tcptls_close_session_file(tcptls_session);
      tcptls_session->parent = NULL;
      ao2_unlock(tcptls_session);

      ao2_ref(tcptls_session, -1);
      tcptls_session = NULL;
   }
   return NULL;
}

static void acl_change_event_cb	(	const struct ast_event *	event,
		void *	userdata
	)		[static]

Definition at line 29160 of file chan_sip.c.

References ast_log(), ast_mutex_lock, ast_mutex_unlock, ast_verbose(), CHANNEL_ACL_RELOAD, LOG_NOTICE, restart_monitor(), sip_reload_lock, and TRUE.

Referenced by acl_change_event_subscribe().

{
   ast_log(LOG_NOTICE, "Reloading chan_sip in response to ACL change event.\n");

   ast_mutex_lock(&sip_reload_lock);

   if (sip_reloading) {
      ast_verbose("Previous SIP reload not yet done\n");
   } else {
      sip_reloading = TRUE;
      sip_reloadreason = CHANNEL_ACL_RELOAD;
   }

   ast_mutex_unlock(&sip_reload_lock);

   restart_monitor();
}

static void acl_change_event_subscribe

(

void

)

[static]

Definition at line 16610 of file chan_sip.c.

References acl_change_event_cb(), AST_EVENT_ACL_CHANGE, AST_EVENT_IE_END, and ast_event_subscribe().

Referenced by build_peer(), and reload_config().

{
   if (!acl_change_event_subscription) {
      acl_change_event_subscription = ast_event_subscribe(AST_EVENT_ACL_CHANGE,
      acl_change_event_cb, "Manager must react to Named ACL changes", NULL, AST_EVENT_IE_END);
   }

}

static void acl_change_event_unsubscribe

(

void

)

[static]

Definition at line 16619 of file chan_sip.c.

References ast_event_unsubscribe().

Referenced by unload_module().

{
   if (acl_change_event_subscription) {
      acl_change_event_subscription = ast_event_unsubscribe(acl_change_event_subscription);
   }
}

static void add_blank

(

struct sip_request *

req

)

[static]

add a blank line if no body

Definition at line 4622 of file chan_sip.c.

References ast_str_append().

Referenced by send_request(), and send_response().

{
   if (!req->lines) {
      /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
      ast_str_append(&req->data, 0, "\r\n");
   }
}

static void add_cc_call_info_to_response	(	struct sip_pvt *	p,
		struct sip_request *	resp
	)		[static]

Definition at line 13597 of file chan_sip.c.

References add_header(), ao2_ref, ast_copy_string(), ast_log(), ast_str_alloca, ast_str_buffer(), ast_str_set(), ast_strlen_zero(), find_sip_cc_agent_by_original_callid(), generate_uri(), LOG_WARNING, and ast_cc_agent::private_data.

Referenced by __transmit_response(), and transmit_response_with_sdp().

{
   char uri[SIPBUFSIZE];
   struct ast_str *header = ast_str_alloca(SIPBUFSIZE);
   struct ast_cc_agent *agent = find_sip_cc_agent_by_original_callid(p);
   struct sip_cc_agent_pvt *agent_pvt;

   if (!agent) {
      /* Um, what? How could the SIP_OFFER_CC flag be set but there not be an
       * agent? Oh well, we'll just warn and return without adding the header.
       */
      ast_log(LOG_WARNING, "Can't find SIP CC agent for call '%s' even though OFFER_CC flag was set?\n", p->callid);
      return;
   }

   agent_pvt = agent->private_data;

   if (!ast_strlen_zero(agent_pvt->subscribe_uri)) {
      ast_copy_string(uri, agent_pvt->subscribe_uri, sizeof(uri));
   } else {
      generate_uri(p, uri, sizeof(uri));
      ast_copy_string(agent_pvt->subscribe_uri, uri, sizeof(agent_pvt->subscribe_uri));
   }
   /* XXX Hardcode "NR" as the m reason for now. This should perhaps be changed
    * to be more accurate. This parameter has no bearing on the actual operation
    * of the feature; it's just there for informational purposes.
    */
   ast_str_set(&header, 0, "<%s>;purpose=call-completion;m=%s", uri, "NR");
   add_header(resp, "Call-Info", ast_str_buffer(header));
   ao2_ref(agent, -1);
}

static void add_codec_to_sdp	(	const struct sip_pvt *	p,
		struct ast_format *	codec,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)		[static]

Add codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 12752 of file chan_sip.c.

References ast_codec_pref_getsize(), AST_FORMAT_G719, AST_FORMAT_G723_1, AST_FORMAT_G729A, AST_FORMAT_ILBC, ast_format_sdp_generate(), AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), AST_RTP_OPT_G726_NONSTANDARD, ast_str_append(), ast_test_flag, ast_verbose(), ast_format_list::cur_ms, ast_format::id, and ast_rtp_codecs::pref.

Referenced by add_sdp().

{
   int rtp_code;
   struct ast_format_list fmt;
   const char *mime;
   unsigned int rate;

   if (debug)
      ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format));

   if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1) ||
       !(mime = ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0)) ||
       !(rate = ast_rtp_lookup_sample_rate2(1, format, 0))) {
      return;
   }

   if (p->rtp) {
      struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
      fmt = ast_codec_pref_getsize(pref, format);
   } else /* I don't see how you couldn't have p->rtp, but good to check for and error out if not there like earlier code */
      return;
   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);

   ast_format_sdp_generate(format, rtp_code, a_buf);

   switch ((int) format->id) {
   case AST_FORMAT_G729A:
      /* Indicate that we don't support VAD (G.729 annex B) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
      break;
   case AST_FORMAT_G723_1:
      /* Indicate that we don't support VAD (G.723.1 annex A) */
      ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
      break;
   case AST_FORMAT_ILBC:
      /* Add information about us using only 20/30 ms packetization */
      ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
      break;
   case AST_FORMAT_SIREN7:
      /* Indicate that we only expect 32Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
      break;
   case AST_FORMAT_SIREN14:
      /* Indicate that we only expect 48Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
      break;
   case AST_FORMAT_G719:
      /* Indicate that we only expect 64Kbps */
      ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
      break;
   }

   if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
      *min_packet_size = fmt.cur_ms;

   /* Our first codec packetization processed cannot be zero */
   if ((*min_packet_size)==0 && fmt.cur_ms)
      *min_packet_size = fmt.cur_ms;
}

static int add_content	(	struct sip_request *	req,
		const char *	line
	)		[static]

Add content (not header) to SIP message.

Definition at line 11451 of file chan_sip.c.

References ast_log(), ast_str_append(), and LOG_WARNING.

Referenced by add_digit(), add_sdp(), add_text(), add_vidupdate(), sipinfo_send(), transmit_cc_notify(), transmit_invite(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), and transmit_state_notify().

{
   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more content when the content has been finalized\n");
      return -1;
   }

   ast_str_append(&req->content, 0, "%s", line);
   return 0;
}

static void add_date

(

struct sip_request *

req

)

[static]

Add date header to SIP message.

Definition at line 12208 of file chan_sip.c.

References add_header().

Referenced by transmit_invite(), transmit_response_with_date(), transmit_response_with_minse(), and transmit_response_with_unsupported().

{
   char tmp[256];
   struct tm tm;
   time_t t = time(NULL);

   gmtime_r(&t, &tm);
   strftime(tmp, sizeof(tmp), "%a, %d %b %Y %T GMT", &tm);
   add_header(req, "Date", tmp);
}

static int add_digit	(	struct sip_request *	req,
		char	digit,
		unsigned int	duration,
		int	mode
	)		[static]

Add DTMF INFO tone to sip message Mode = 0 for application/dtmf-relay (Cisco) 1 for application/dtmf.

Definition at line 12474 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_info_with_digit().

{
   char tmp[256];
   int event;
   if (mode) {
      /* Application/dtmf short version used by some implementations */
      if ('0' <= digit && digit <= '9') {
         event = digit - '0';
      } else if (digit == '*') {
         event = 10;
      } else if (digit == '#') {
         event = 11;
      } else if ('A' <= digit && digit <= 'D') {
         event = 12 + digit - 'A';
      } else if ('a' <= digit && digit <= 'd') {
         event = 12 + digit - 'a';
      } else {
         /* Unknown digit */
         event = 0;
      }
      snprintf(tmp, sizeof(tmp), "%d\r\n", event);
      add_header(req, "Content-Type", "application/dtmf");
      add_content(req, tmp);
   } else {
      /* Application/dtmf-relay as documented by Cisco */
      snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=%u\r\n", digit, duration);
      add_header(req, "Content-Type", "application/dtmf-relay");
      add_content(req, tmp);
   }
   return 0;
}

static void add_diversion	(	struct sip_request *	req,
		struct sip_pvt *	pvt
	)		[static]

Add "Diversion" header to outgoing message.

We need to add a Diversion header if the owner channel of this dialog has redirecting information associated with it.

Parameters:

req	The request/response to which we will add the header
pvt	The sip_pvt which represents the call-leg

Definition at line 14017 of file chan_sip.c.

References add_header(), ast_channel_redirecting(), ast_channel_redirecting_effective_from(), ast_sockaddr_stringify_host_remote(), ast_strlen_zero(), ast_party_id::name, ast_party_id::number, ast_party_redirecting::reason, sip_cfg, sip_reason_code_to_str(), ast_party_name::str, ast_party_number::str, ast_party_name::valid, and ast_party_number::valid.

Referenced by __transmit_response(), transmit_invite(), and update_redirecting().

{
   struct ast_party_id diverting_from;
   const char *reason;
   char header_text[256];

   /* We skip this entirely if the configuration doesn't allow diversion headers */
   if (!sip_cfg.send_diversion) {
      return;
   }

   if (!pvt->owner) {
      return;
   }

   diverting_from = ast_channel_redirecting_effective_from(pvt->owner);
   if (!diverting_from.number.valid
      || ast_strlen_zero(diverting_from.number.str)) {
      return;
   }

   reason = sip_reason_code_to_str(ast_channel_redirecting(pvt->owner)->reason);

   /* We at least have a number to place in the Diversion header, which is enough */
   if (!diverting_from.name.valid
      || ast_strlen_zero(diverting_from.name.str)) {
      snprintf(header_text, sizeof(header_text), "<sip:%s@%s>;reason=%s", diverting_from.number.str,
            ast_sockaddr_stringify_host_remote(&pvt->ourip), reason);
   } else {
      snprintf(header_text, sizeof(header_text), "\"%s\" <sip:%s@%s>;reason=%s",
            diverting_from.name.str, diverting_from.number.str,
            ast_sockaddr_stringify_host_remote(&pvt->ourip), reason);
   }

   add_header(req, "Diversion", header_text);
}

static void add_dtls_to_sdp	(	struct ast_rtp_instance *	instance,
		struct ast_str **	a_buf
	)		[static]

Add DTLS attributes to SDP.

Definition at line 12709 of file chan_sip.c.

References ast_rtp_engine_dtls::active, AST_RTP_DTLS_CONNECTION_EXISTING, AST_RTP_DTLS_CONNECTION_NEW, AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), ast_str_append(), ast_rtp_engine_dtls::get_connection, ast_rtp_engine_dtls::get_fingerprint, and ast_rtp_engine_dtls::get_setup.

Referenced by add_sdp().

{
   struct ast_rtp_engine_dtls *dtls;
   const char *fingerprint;

   if (!instance || !(dtls = ast_rtp_instance_get_dtls(instance)) || !dtls->active(instance)) {
      return;
   }

   switch (dtls->get_connection(instance)) {
   case AST_RTP_DTLS_CONNECTION_NEW:
      ast_str_append(a_buf, 0, "a=connection:new\r\n");
      break;
   case AST_RTP_DTLS_CONNECTION_EXISTING:
      ast_str_append(a_buf, 0, "a=connection:existing\r\n");
      break;
   default:
      break;
   }

   switch (dtls->get_setup(instance)) {
   case AST_RTP_DTLS_SETUP_ACTIVE:
      ast_str_append(a_buf, 0, "a=setup:active\r\n");
      break;
   case AST_RTP_DTLS_SETUP_PASSIVE:
      ast_str_append(a_buf, 0, "a=setup:passive\r\n");
      break;
   case AST_RTP_DTLS_SETUP_ACTPASS:
      ast_str_append(a_buf, 0, "a=setup:actpass\r\n");
      break;
   case AST_RTP_DTLS_SETUP_HOLDCONN:
      ast_str_append(a_buf, 0, "a=setup:holdconn\r\n");
      break;
   default:
      break;
   }

   if ((fingerprint = dtls->get_fingerprint(instance, AST_RTP_DTLS_HASH_SHA1))) {
      ast_str_append(a_buf, 0, "a=fingerprint:SHA-1 %s\r\n", fingerprint);
   }
}

static void add_expires	(	struct sip_request *	req,
		int	expires
	)		[static]

Add Expires header to SIP message.

Definition at line 12220 of file chan_sip.c.

References add_header().

Referenced by respprep(), transmit_invite(), and transmit_register().

{
   char tmp[32];

   snprintf(tmp, sizeof(tmp), "%d", expires);
   add_header(req, "Expires", tmp);
}

static int add_header	(	struct sip_request *	req,
		const char *	var,
		const char *	value
	)		[static]

Add header to SIP message.

Definition at line 11393 of file chan_sip.c.

References ast_log(), ast_str_append(), ast_str_strlen(), find_alias(), LOG_WARNING, and sip_cfg.

Referenced by __transmit_response(), add_cc_call_info_to_response(), add_date(), add_digit(), add_diversion(), add_expires(), add_max_forwards(), add_required_respheader(), add_route(), add_rpid(), add_sdp(), add_supported(), add_text(), add_vidupdate(), copy_all_header(), copy_header(), copy_via_headers(), finalize_content(), initreqprep(), reqprep(), respprep(), sipinfo_send(), transmit_cc_notify(), transmit_info_with_aoc(), transmit_invite(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), transmit_response_with_allow(), transmit_response_with_auth(), transmit_response_with_minexpires(), transmit_response_with_minse(), transmit_response_with_retry_after(), transmit_response_with_sip_etag(), transmit_response_with_unsupported(), transmit_state_notify(), and update_connectedline().

{
   if (req->headers == SIP_MAX_HEADERS) {
      ast_log(LOG_WARNING, "Out of SIP header space\n");
      return -1;
   }

   if (req->lines) {
      ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
      return -1;
   }

   if (sip_cfg.compactheaders) {
      var = find_alias(var, var);
   }

   ast_str_append(&req->data, 0, "%s: %s\r\n", var, value);
   req->header[req->headers] = ast_str_strlen(req->data);

   req->headers++;

   return 0;   
}

static void add_ice_to_sdp	(	struct ast_rtp_instance *	instance,
		struct ast_str **	a_buf
	)		[static]

Add ICE attributes to SDP.

Definition at line 12641 of file chan_sip.c.

References ast_rtp_engine_ice_candidate::address, ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_ref, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, AST_RTP_ICE_COMPONENT_RTCP, ast_rtp_instance_get_ice(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_stringify_host(), ast_sockaddr_stringify_port(), ast_str_append(), ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::get_local_candidates, ast_rtp_engine_ice::get_password, ast_rtp_engine_ice::get_ufrag, ast_rtp_engine_ice_candidate::id, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by add_sdp().

{
   struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(instance);
   const char *username, *password;
   struct ao2_container *candidates;
   struct ao2_iterator i;
   struct ast_rtp_engine_ice_candidate *candidate;

   /* If no ICE support is present we can't very well add the attributes */
   if (!ice || !(candidates = ice->get_local_candidates(instance))) {
      return;
   }

   if ((username = ice->get_ufrag(instance))) {
      ast_str_append(a_buf, 0, "a=ice-ufrag:%s\r\n", username);
   }
   if ((password = ice->get_password(instance))) {
      ast_str_append(a_buf, 0, "a=ice-pwd:%s\r\n", password);
   }

   i = ao2_iterator_init(candidates, 0);

   while ((candidate = ao2_iterator_next(&i))) {
      ast_str_append(a_buf, 0, "a=candidate:%s %d %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
      ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_host(&candidate->address));

      if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX
         && candidate->id == AST_RTP_ICE_COMPONENT_RTCP) {
         ast_str_append(a_buf, 0, "%d typ ", ast_sockaddr_port(&candidate->address) + 1);
      } else {
         ast_str_append(a_buf, 0, "%s typ ", ast_sockaddr_stringify_port(&candidate->address));
      }

      if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
         ast_str_append(a_buf, 0, "host");
      } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
         ast_str_append(a_buf, 0, "srflx");
      } else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_RELAYED) {
         ast_str_append(a_buf, 0, "relay");
      }

      if (!ast_sockaddr_isnull(&candidate->relay_address)) {
         ast_str_append(a_buf, 0, " raddr %s ", ast_sockaddr_stringify_host(&candidate->relay_address));
         ast_str_append(a_buf, 0, "rport %s", ast_sockaddr_stringify_port(&candidate->relay_address));
      }

      ast_str_append(a_buf, 0, "\r\n");
      ao2_ref(candidate, -1);
   }

   ao2_iterator_destroy(&i);

   ao2_ref(candidates, -1);
}

static int add_max_forwards	(	struct sip_pvt *	dialog,
		struct sip_request *	req
	)		[static]

Add 'Max-Forwards' header to SIP message.

Precondition:

dialog is assumed to be locked while calling this function

Definition at line 11421 of file chan_sip.c.

References add_header().

Referenced by initreqprep(), reqprep(), and transmit_register().

{
   char clen[10];

   snprintf(clen, sizeof(clen), "%d", dialog->maxforwards);

   return add_header(req, "Max-Forwards", clen);
}

static void add_msg_header	(	struct sip_pvt *	pvt,
		const char *	hdr_name,
		const char *	hdr_value
	)		[static]

Definition at line 12420 of file chan_sip.c.

References ast_calloc, and AST_LIST_INSERT_TAIL.

Referenced by sip_msg_send().

{
   size_t hdr_len_name;
   size_t hdr_len_value;
   struct sip_msg_hdr *node;
   char *pos;

   hdr_len_name = strlen(hdr_name) + 1;
   hdr_len_value = strlen(hdr_value) + 1;

   node = ast_calloc(1, sizeof(*node) + hdr_len_name + hdr_len_value);
   if (!node) {
      return;
   }
   pos = node->stuff;
   node->name = pos;
   strcpy(pos, hdr_name);
   pos += hdr_len_name;
   node->value = pos;
   strcpy(pos, hdr_value);

   AST_LIST_INSERT_TAIL(&pvt->msg_headers, node, next);
}

static void add_noncodec_to_sdp	(	const struct sip_pvt *	p,
		int	format,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug
	)		[static]

Add RFC 2833 DTMF offer to SDP.

Definition at line 12902 of file chan_sip.c.

References ast_rtp_codecs_payload_code(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), and ast_verbose().

Referenced by add_sdp().

{
   int rtp_code;

   if (debug)
      ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0));
   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, NULL, format)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(0, NULL, format, 0),
             ast_rtp_lookup_sample_rate2(0, NULL, format));
   if (format == AST_RTP_DTMF)   /* Indicate we support DTMF and FLASH... */
      ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}

static void add_peer_mailboxes	(	struct sip_peer *	peer,
		const char *	value
	)		[static]

Todo:

document this function

Definition at line 30578 of file chan_sip.c.

References ast_calloc, AST_LIST_INSERT_TAIL, AST_LIST_TRAVERSE, ast_strip(), ast_strlen_zero(), context, mailbox, mbox(), and S_OR.

Referenced by build_peer().

{
   char *next, *mbox, *context;

   next = ast_strdupa(value);

   while ((mbox = context = strsep(&next, ","))) {
      struct sip_mailbox *mailbox;
      int duplicate = 0;
      /* remove leading/trailing whitespace from mailbox string */
      mbox = ast_strip(mbox);
      strsep(&context, "@");

      if (ast_strlen_zero(mbox)) {
         continue;
      }

      /* Check whether the mailbox is already in the list */
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         if (!strcmp(mailbox->mailbox, mbox) && !strcmp(S_OR(mailbox->context, ""), S_OR(context, ""))) {
            duplicate = 1;
            break;
         }
      }
      if (duplicate) {
         continue;
      }

      if (!(mailbox = ast_calloc(1, sizeof(*mailbox) + strlen(mbox) + strlen(S_OR(context, ""))))) {
         continue;
      }

      if (!ast_strlen_zero(context)) {
         mailbox->context = mailbox->mailbox + strlen(mbox) + 1;
         strcpy(mailbox->context, context); /* SAFE */
      }
      strcpy(mailbox->mailbox, mbox); /* SAFE */

      AST_LIST_INSERT_TAIL(&peer->mailboxes, mailbox, entry);
   }
}

static void add_peer_mwi_subs

(

struct sip_peer *

peer

)

[static]

Definition at line 27612 of file chan_sip.c.

References AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, ast_event_subscribe(), ast_event_unsubscribe(), AST_LIST_TRAVERSE, mailbox, mwi_event_cb(), and S_OR.

Referenced by build_peer(), and handle_request_subscribe().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      if (mailbox->event_sub) {
         ast_event_unsubscribe(mailbox->event_sub);
      }

      mailbox->event_sub = ast_event_subscribe(AST_EVENT_MWI, mwi_event_cb, "SIP mbox event", peer,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
   }
}

static void add_realm_authentication	(	struct sip_auth_container **	credentials,
		const char *	configuration,
		int	lineno
	)		[static]

Definition at line 30366 of file chan_sip.c.

References ao2_t_alloc, ast_calloc, ast_copy_string(), ast_debug, AST_LIST_INSERT_TAIL, ast_log(), ast_strlen_zero(), ast_verb, destroy_realm_authentication(), LOG_WARNING, and secret.

Referenced by build_peer(), and reload_config().

{
   char *authcopy;
   char *username=NULL, *realm=NULL, *secret=NULL, *md5secret=NULL;
   struct sip_auth *auth;

   if (ast_strlen_zero(configuration)) {
      /* Nothing to add */
      return;
   }

   ast_debug(1, "Auth config ::  %s\n", configuration);

   authcopy = ast_strdupa(configuration);
   username = authcopy;

   /* split user[:secret] and relm */
   realm = strrchr(username, '@');
   if (realm)
      *realm++ = '\0';
   if (ast_strlen_zero(username) || ast_strlen_zero(realm)) {
      ast_log(LOG_WARNING, "Format for authentication entry is user[:secret]@realm at line %d\n", lineno);
      return;
   }

   /* parse username at ':' for secret, or '#" for md5secret */
   if ((secret = strchr(username, ':'))) {
      *secret++ = '\0';
   } else if ((md5secret = strchr(username, '#'))) {
      *md5secret++ = '\0';
   }

   /* Create the continer if needed. */
   if (!*credentials) {
      *credentials = ao2_t_alloc(sizeof(**credentials), destroy_realm_authentication,
         "Create realm auth container.");
      if (!*credentials) {
         /* Failed to create the credentials container. */
         return;
      }
   }

   /* Create the authentication credential entry. */
   auth = ast_calloc(1, sizeof(*auth));
   if (!auth) {
      return;
   }
   ast_copy_string(auth->realm, realm, sizeof(auth->realm));
   ast_copy_string(auth->username, username, sizeof(auth->username));
   if (secret)
      ast_copy_string(auth->secret, secret, sizeof(auth->secret));
   if (md5secret)
      ast_copy_string(auth->md5secret, md5secret, sizeof(auth->md5secret));

   /* Add credential to container list. */
   AST_LIST_INSERT_TAIL(&(*credentials)->list, auth, node);

   ast_verb(3, "Added authentication for realm %s\n", realm);
}

static void add_required_respheader

(

struct sip_request *

req

)

[static]

Definition at line 4702 of file chan_sip.c.

References add_header(), ARRAY_LEN, ast_free, ast_str_append(), ast_str_buffer(), ast_str_create(), ast_str_strlen(), str, and text.

Referenced by transmit_response_with_sdp().

{
   struct ast_str *str;
   int i;

   if (!req->reqsipoptions) {
      return;
   }

   str = ast_str_create(32);

   for (i = 0; i < ARRAY_LEN(sip_options); ++i) {
      if (!(req->reqsipoptions & sip_options[i].id)) {
         continue;
      }
      if (ast_str_strlen(str) > 0) {
         ast_str_append(&str, 0, ", ");
      }
      ast_str_append(&str, 0, "%s", sip_options[i].text);
   }

   if (ast_str_strlen(str) > 0) {
      add_header(req, "Require", ast_str_buffer(str));
   }

   ast_free(str);
}

static void add_route	(	struct sip_request *	req,
		struct sip_route *	route
	)		[static]

Add route header into request per learned route.

Definition at line 11562 of file chan_sip.c.

References add_header(), and ast_copy_string().

Referenced by initreqprep(), and reqprep().

{
   char r[SIPBUFSIZE*2], *p;
   int n, rem = sizeof(r);

   if (!route)
      return;

   p = r;
   for (;route ; route = route->next) {
      n = strlen(route->hop);
      if (rem < n+3) /* we need room for ",<route>" */
         break;
      if (p != r) {  /* add a separator after fist route */
         *p++ = ',';
         --rem;
      }
      *p++ = '<';
      ast_copy_string(p, route->hop, rem); /* cannot fail */
      p += n;
      *p++ = '>';
      rem -= (n+2);
   }
   *p = '\0';
   add_header(req, "Route", r);
}

static int add_rpid	(	struct sip_request *	req,
		struct sip_pvt *	p
	)		[static]

Add Remote-Party-ID header to SIP message.

Precondition:

if p->owner exists, it must be locked

Definition at line 12510 of file chan_sip.c.

References add_header(), ast_channel_connected_effective_id(), ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_ALLOWED_NETWORK_NUMBER, AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN, AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRES_NUMBER_NOT_AVAILABLE, AST_PRES_PROHIB_NETWORK_NUMBER, AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, AST_PRES_RESTRICTION, ast_sockaddr_stringify_host_remote(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_set(), ast_strlen_zero(), ast_test_flag, ast_uri_encode(), ast_uri_sip_user, ast_party_id::name, ast_party_id::number, S_COR, ast_party_name::str, ast_party_number::str, ast_party_name::valid, and ast_party_number::valid.

Referenced by __transmit_response(), transmit_invite(), transmit_reinvite_with_sdp(), transmit_response_with_sdp(), and update_connectedline().

{
   struct ast_str *tmp = ast_str_alloca(256);
   char tmp2[256];
   char *lid_num;
   char *lid_name;
   int lid_pres;
   const char *fromdomain;
   const char *privacy = NULL;
   const char *screen = NULL;
   struct ast_party_id connected_id;
   const char *anonymous_string = "\"Anonymous\" <sip:anonymous@anonymous.invalid>";

   if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
      return 0;
   }

   if (!p->owner) {
      return 0;
   }
   connected_id = ast_channel_connected_effective_id(p->owner);
   lid_num = S_COR(connected_id.number.valid, connected_id.number.str, NULL);
   if (!lid_num) {
      return 0;
   }
   lid_name = S_COR(connected_id.name.valid, connected_id.name.str, NULL);
   if (!lid_name) {
      lid_name = lid_num;
   }
   lid_pres = ast_party_id_presentation(&connected_id);

   if (((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) &&
         (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_NO)) {
      /* If pres is not allowed and we don't trust the peer, we don't apply an RPID header */
      return 0;
   }

   fromdomain = p->fromdomain;
   if (!fromdomain ||
         ((ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_YES) &&
         !strcmp("anonymous.invalid", fromdomain))) {
      /* If the fromdomain is NULL or if it was set to anonymous.invalid due to privacy settings and we trust the peer,
       * use the host IP address */
      fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);
   }

   lid_num = ast_uri_encode(lid_num, tmp2, sizeof(tmp2), ast_uri_sip_user);

   if (ast_test_flag(&p->flags[0], SIP_SENDRPID_PAI)) {
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) != SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY) {
         /* trust_id_outbound = yes - Always give full information even if it's private, but append a privacy header
          * When private data is included */
         ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
         if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
            add_header(req, "Privacy", "id");
         }
      } else {
         /* trust_id_outbound = legacy - behave in a non RFC-3325 compliant manner and send anonymized data when
          * when handling private data. */
         if ((lid_pres & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
            ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>", lid_name, lid_num, fromdomain);
         } else {
            ast_str_set(&tmp, -1, "%s", anonymous_string);
         }
      }
      add_header(req, "P-Asserted-Identity", ast_str_buffer(tmp));
   } else {
      ast_str_set(&tmp, -1, "\"%s\" <sip:%s@%s>;party=%s", lid_name, lid_num, fromdomain, p->outgoing_call ? "calling" : "called");

      switch (lid_pres) {
      case AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED:
      case AST_PRES_ALLOWED_USER_NUMBER_FAILED_SCREEN:
         privacy = "off";
         screen = "no";
         break;
      case AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN:
      case AST_PRES_ALLOWED_NETWORK_NUMBER:
         privacy = "off";
         screen = "yes";
         break;
      case AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED:
      case AST_PRES_PROHIB_USER_NUMBER_FAILED_SCREEN:
         privacy = "full";
         screen = "no";
         break;
      case AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN:
      case AST_PRES_PROHIB_NETWORK_NUMBER:
         privacy = "full";
         screen = "yes";
         break;
      case AST_PRES_NUMBER_NOT_AVAILABLE:
         break;
      default:
         if ((lid_pres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
            privacy = "full";
         }
         else
            privacy = "off";
         screen = "no";
         break;
      }

      if (!ast_strlen_zero(privacy) && !ast_strlen_zero(screen)) {
         ast_str_append(&tmp, -1, ";privacy=%s;screen=%s", privacy, screen);
      }

      add_header(req, "Remote-Party-ID", ast_str_buffer(tmp));
   }
   return 0;
}

static enum sip_result add_sdp	(	struct sip_request *	resp,
		struct sip_pvt *	p,
		int	oldsdp,
		int	add_audio,
		int	add_t38
	)		[static]

Add Session Description Protocol message.

If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism is used in Session-Timers where RE-INVITEs are used for refreshing SIP sessions without modifying the media session in any way.

Definition at line 13071 of file chan_sip.c.

References add_codec_to_sdp(), add_content(), add_dtls_to_sdp(), add_header(), add_ice_to_sdp(), add_noncodec_to_sdp(), add_tcodec_to_sdp(), add_vcodec_to_sdp(), ao2_lock, ao2_t_link, ao2_t_unlink, ao2_unlock, ast_codec_pref_index(), AST_CODEC_PREF_SIZE, ast_debug, ast_format_cap_add(), ast_format_cap_alloc_nolock(), ast_format_cap_copy(), ast_format_cap_destroy(), ast_format_cap_get_compatible_format(), ast_format_cap_has_type(), ast_format_cap_is_empty(), ast_format_cap_iscompatible(), ast_format_cap_iter_end(), ast_format_cap_iter_next(), ast_format_cap_iter_start(), ast_format_cap_joint_copy(), AST_FORMAT_GET_TYPE, AST_FORMAT_TYPE_AUDIO, AST_FORMAT_TYPE_TEXT, AST_FORMAT_TYPE_VIDEO, ast_free, ast_getformatname_multiple(), ast_internal_timing_enabled(), AST_LIST_EMPTY, AST_LIST_TRAVERSE, ast_log(), ast_random(), AST_RTP_MAX, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_is_ipv4_mapped(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_addr_remote(), ast_sockaddr_stringify_port(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_create(), ast_strlen_zero(), AST_T38_RATE_MANAGEMENT_LOCAL_TCF, AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF, ast_test_flag, ast_udptl_get_error_correction_scheme(), ast_udptl_get_local_max_datagram(), ast_udptl_get_us(), ast_verbose(), debug, FALSE, get_crypto_attrib(), get_our_media_address(), get_sdp_rtp_profile(), ast_format::id, LOG_WARNING, sip_debug_test_pvt(), t38_get_rate(), TRUE, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, UDPTL_ERROR_CORRECTION_REDUNDANCY, and version.

Referenced by transmit_invite(), transmit_reinvite_with_sdp(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), and update_connectedline().

{
   struct ast_format_cap *alreadysent = ast_format_cap_alloc_nolock();
   struct ast_format_cap *tmpcap = ast_format_cap_alloc_nolock();
   int res = AST_SUCCESS;
   int doing_directmedia = FALSE;
   struct ast_sockaddr addr = { {0,} };
   struct ast_sockaddr vaddr = { {0,} };
   struct ast_sockaddr taddr = { {0,} };
   struct ast_sockaddr udptladdr = { {0,} };
   struct ast_sockaddr dest = { {0,} };
   struct ast_sockaddr vdest = { {0,} };
   struct ast_sockaddr tdest = { {0,} };
   struct ast_sockaddr udptldest = { {0,} };

   /* SDP fields */
   struct offered_media *offer;
   char *version =   "v=0\r\n";     /* Protocol version */
   char subject[256];            /* Subject of the session */
   char owner[256];           /* Session owner/creator */
   char connection[256];            /* Connection data */
   char *session_time = "t=0 0\r\n";         /* Time the session is active */
   char bandwidth[256] = "";        /* Max bitrate */
   char *hold = "";
   struct ast_str *m_audio = ast_str_alloca(256);  /* Media declaration line for audio */
   struct ast_str *m_video = ast_str_alloca(256);  /* Media declaration line for video */
   struct ast_str *m_text = ast_str_alloca(256);   /* Media declaration line for text */
   struct ast_str *m_modem = ast_str_alloca(256);  /* Media declaration line for modem */
   struct ast_str *a_audio = ast_str_create(256); /* Attributes for audio */
   struct ast_str *a_video = ast_str_create(256); /* Attributes for video */
   struct ast_str *a_text = ast_str_create(256);  /* Attributes for text */
   struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
   const char *a_crypto = NULL;
   const char *v_a_crypto = NULL;
   const char *t_a_crypto = NULL;

   int x;
   struct ast_format tmp_fmt;
   int needaudio = FALSE;
   int needvideo = FALSE;
   int needtext = FALSE;
   int debug = sip_debug_test_pvt(p);
   int min_audio_packet_size = 0;
   int min_video_packet_size = 0;
   int min_text_packet_size = 0;

   char codecbuf[SIPBUFSIZE];
   char buf[SIPBUFSIZE];

   /* Set the SDP session name */
   snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);

   if (!alreadysent || !tmpcap) {
      res = AST_FAILURE;
      goto add_sdp_cleanup;
   }
   if (!p->rtp) {
      ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
      res = AST_FAILURE;
      goto add_sdp_cleanup;

   }
   /* XXX We should not change properties in the SIP dialog until
      we have acceptance of the offer if this is a re-invite */

   /* Set RTP Session ID and version */
   if (!p->sessionid) {
      p->sessionid = (int)ast_random();
      p->sessionversion = p->sessionid;
   } else {
      if (oldsdp == FALSE)
         p->sessionversion++;
   }

   if (add_audio) {
      doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && !(ast_format_cap_is_empty(p->redircaps))) ? TRUE : FALSE;
      /* Check if we need video in this call */
      if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_VIDEO)) && !p->novideo) {
         ast_format_cap_joint_copy(p->jointcaps, p->redircaps, tmpcap);
         if (doing_directmedia && !ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_VIDEO)) {
            ast_debug(2, "This call needs video offers, but caller probably did not offer it!\n");
         } else if (p->vrtp) {
            needvideo = TRUE;
            ast_debug(2, "This call needs video offers!\n");
         } else {
            ast_debug(2, "This call needs video offers, but there's no video support enabled!\n");
         }
      }
      /* Check if we need text in this call */
      if ((ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_TEXT)) && !p->notext) {
         if (sipdebug_text)
            ast_verbose("We think we can do text\n");
         if (p->trtp) {
            if (sipdebug_text) {
               ast_verbose("And we have a text rtp object\n");
            }
            needtext = TRUE;
            ast_debug(2, "This call needs text offers! \n");
         } else {
            ast_debug(2, "This call needs text offers, but there's no text support enabled ! \n");
         }
      }
   }

   get_our_media_address(p, needvideo, needtext, &addr, &vaddr, &taddr, &dest, &vdest, &tdest);

   snprintf(owner, sizeof(owner), "o=%s %d %d IN %s %s\r\n",
       ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner,
       p->sessionid, p->sessionversion,
       (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
         "IP6" : "IP4",
       ast_sockaddr_stringify_addr_remote(&dest));

   snprintf(connection, sizeof(connection), "c=IN %s %s\r\n",
       (ast_sockaddr_is_ipv6(&dest) && !ast_sockaddr_is_ipv4_mapped(&dest)) ?
         "IP6" : "IP4",
       ast_sockaddr_stringify_addr_remote(&dest));

   if (add_audio) {
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) {
         hold = "a=recvonly\r\n";
         doing_directmedia = FALSE;
      } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE) {
         hold = "a=inactive\r\n";
         doing_directmedia = FALSE;
      } else {
         hold = "a=sendrecv\r\n";
      }

      ast_format_cap_copy(tmpcap, p->jointcaps);

      /* XXX note, Video and Text are negated - 'true' means 'no' */
      ast_debug(1, "** Our capability: %s Video flag: %s Text flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), tmpcap),
           p->novideo ? "True" : "False", p->notext ? "True" : "False");
      ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcaps));

      if (doing_directmedia) {
         ast_format_cap_joint_copy(p->jointcaps, p->redircaps, tmpcap);
         ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), tmpcap));
      }

      /* Check if we need audio */
      if (ast_format_cap_has_type(tmpcap, AST_FORMAT_TYPE_AUDIO))
         needaudio = TRUE;

      if (debug) {
         ast_verbose("Audio is at %s\n", ast_sockaddr_stringify_port(&addr));
      }

      /* Ok, we need video. Let's add what we need for video and set codecs.
         Video is handled differently than audio since we can not transcode. */
      if (needvideo) {
         get_crypto_attrib(p, p->vsrtp, &v_a_crypto);
         ast_str_append(&m_video, 0, "m=video %d %s", ast_sockaddr_port(&vdest),
                   get_sdp_rtp_profile(p, v_a_crypto ? 1 : 0, p->vrtp));

         /* Build max bitrate string */
         if (p->maxcallbitrate)
            snprintf(bandwidth, sizeof(bandwidth), "b=CT:%d\r\n", p->maxcallbitrate);
         if (debug) {
            ast_verbose("Video is at %s\n", ast_sockaddr_stringify(&vdest));
         }

         if (!doing_directmedia) {
            if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
               add_ice_to_sdp(p->vrtp, &a_video);
            }

            add_dtls_to_sdp(p->vrtp, &a_video);
         }
      }

      /* Ok, we need text. Let's add what we need for text and set codecs.
         Text is handled differently than audio since we can not transcode. */
      if (needtext) {
         if (sipdebug_text)
            ast_verbose("Lets set up the text sdp\n");
         get_crypto_attrib(p, p->tsrtp, &t_a_crypto);
         ast_str_append(&m_text, 0, "m=text %d %s", ast_sockaddr_port(&tdest),
                   get_sdp_rtp_profile(p, t_a_crypto ? 1 : 0, p->trtp));
         if (debug) {  /* XXX should I use tdest below ? */
            ast_verbose("Text is at %s\n", ast_sockaddr_stringify(&taddr));
         }

         if (!doing_directmedia) {
            if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
               add_ice_to_sdp(p->trtp, &a_text);
            }

            add_dtls_to_sdp(p->trtp, &a_text);
         }
      }

      /* Start building generic SDP headers */

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      get_crypto_attrib(p, p->srtp, &a_crypto);
      ast_str_append(&m_audio, 0, "m=audio %d %s", ast_sockaddr_port(&dest),
                get_sdp_rtp_profile(p, a_crypto ? 1 : 0, p->rtp));

      /* Now, start adding audio codecs. These are added in this order:
         - First what was requested by the calling channel
         - Then preferences in order from sip.conf device config for this peer/user
         - Then other codecs in capabilities, including video
      */

      /* Prefer the audio codec we were requested to use, first, no matter what
         Note that p->prefcodec can include video codecs, so ignore them
      */
      ast_format_cap_iter_start(p->prefcaps);
      while (!(ast_format_cap_iter_next(p->prefcaps, &tmp_fmt))) {
         if (AST_FORMAT_GET_TYPE(tmp_fmt.id) != AST_FORMAT_TYPE_AUDIO ||
            !ast_format_cap_iscompatible(tmpcap, &tmp_fmt)) {
            continue;
         }
         add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
         ast_format_cap_add(alreadysent, &tmp_fmt);
      }
      ast_format_cap_iter_end(p->prefcaps);

      /* Start by sending our preferred audio/video codecs */
      for (x = 0; x < AST_CODEC_PREF_SIZE; x++) {
         struct ast_format pref;

         if (!(ast_codec_pref_index(&p->prefs, x, &pref)))
            break;

         if (!ast_format_cap_get_compatible_format(tmpcap, &pref, &tmp_fmt))
            continue;

         if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt))
            continue;

         if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) {
            add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
         } else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) {
            add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
         } else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) {
            add_tcodec_to_sdp(p, &tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size);
         }

         ast_format_cap_add(alreadysent, &tmp_fmt);
      }

      /* Now send any other common audio and video codecs, and non-codec formats: */
      ast_format_cap_iter_start(tmpcap);
      while (!(ast_format_cap_iter_next(tmpcap, &tmp_fmt))) {
         if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt))
            continue;

         if (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_AUDIO) {
            add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size);
         } else if (needvideo && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_VIDEO)) {
            add_vcodec_to_sdp(p, &tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
         } else if (needtext && (AST_FORMAT_GET_TYPE(tmp_fmt.id) == AST_FORMAT_TYPE_TEXT)) {
            add_tcodec_to_sdp(p, &tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size);
         }
      }
      ast_format_cap_iter_end(tmpcap);

      /* Now add DTMF RFC2833 telephony-event as a codec */
      for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
         if (!(p->jointnoncodeccapability & x))
            continue;

         add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
      }

      ast_debug(3, "-- Done with adding codecs to SDP\n");

      if (!p->owner || !ast_internal_timing_enabled(p->owner))
         ast_str_append(&a_audio, 0, "a=silenceSupp:off - - - -\r\n");

      if (min_audio_packet_size)
         ast_str_append(&a_audio, 0, "a=ptime:%d\r\n", min_audio_packet_size);

      /* XXX don't think you can have ptime for video */
      if (min_video_packet_size)
         ast_str_append(&a_video, 0, "a=ptime:%d\r\n", min_video_packet_size);

      /* XXX don't think you can have ptime for text */
      if (min_text_packet_size)
         ast_str_append(&a_text, 0, "a=ptime:%d\r\n", min_text_packet_size);

      if (!doing_directmedia) {
         if (ast_test_flag(&p->flags[2], SIP_PAGE3_ICE_SUPPORT)) {
            add_ice_to_sdp(p->rtp, &a_audio);
         }

         add_dtls_to_sdp(p->rtp, &a_audio);
      }
   }

   if (add_t38) {
      /* Our T.38 end is */
      ast_udptl_get_us(p->udptl, &udptladdr);

      /* We don't use directmedia for T.38, so keep the destination the same as our IP address. */
      ast_sockaddr_copy(&udptldest, &p->ourip);
      ast_sockaddr_set_port(&udptldest, ast_sockaddr_port(&udptladdr));

      if (debug) {
         ast_debug(1, "T.38 UDPTL is at %s port %d\n", ast_sockaddr_stringify_addr(&p->ourip), ast_sockaddr_port(&udptladdr));
      }

      /* We break with the "recommendation" and send our IP, in order that our
         peer doesn't have to ast_gethostbyname() us */

      ast_str_append(&m_modem, 0, "m=image %d udptl t38\r\n", ast_sockaddr_port(&udptldest));

      if (ast_sockaddr_cmp(&udptldest, &dest)) {
         ast_str_append(&m_modem, 0, "c=IN %s %s\r\n",
               (ast_sockaddr_is_ipv6(&udptldest) && !ast_sockaddr_is_ipv4_mapped(&udptldest)) ?
               "IP6" : "IP4", ast_sockaddr_stringify_addr_remote(&udptldest));
      }

      ast_str_append(&a_modem, 0, "a=T38FaxVersion:%d\r\n", p->t38.our_parms.version);
      ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", t38_get_rate(p->t38.our_parms.rate));
      if (p->t38.our_parms.fill_bit_removal) {
         ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
      }
      if (p->t38.our_parms.transcoding_mmr) {
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingMMR\r\n");
      }
      if (p->t38.our_parms.transcoding_jbig) {
         ast_str_append(&a_modem, 0, "a=T38FaxTranscodingJBIG\r\n");
      }
      switch (p->t38.our_parms.rate_management) {
      case AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF:
         ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:transferredTCF\r\n");
         break;
      case AST_T38_RATE_MANAGEMENT_LOCAL_TCF:
         ast_str_append(&a_modem, 0, "a=T38FaxRateManagement:localTCF\r\n");
         break;
      }
      ast_str_append(&a_modem, 0, "a=T38FaxMaxDatagram:%u\r\n", ast_udptl_get_local_max_datagram(p->udptl));
      switch (ast_udptl_get_error_correction_scheme(p->udptl)) {
      case UDPTL_ERROR_CORRECTION_NONE:
         break;
      case UDPTL_ERROR_CORRECTION_FEC:
         ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPFEC\r\n");
         break;
      case UDPTL_ERROR_CORRECTION_REDUNDANCY:
         ast_str_append(&a_modem, 0, "a=T38FaxUdpEC:t38UDPRedundancy\r\n");
         break;
      }
   }

   if (needaudio)
      ast_str_append(&m_audio, 0, "\r\n");
   if (needvideo)
      ast_str_append(&m_video, 0, "\r\n");
   if (needtext)
      ast_str_append(&m_text, 0, "\r\n");

   add_header(resp, "Content-Type", "application/sdp");
   add_content(resp, version);
   add_content(resp, owner);
   add_content(resp, subject);
   add_content(resp, connection);
   /* only if video response is appropriate */
   if (needvideo) {
      add_content(resp, bandwidth);
   }
   add_content(resp, session_time);
   /* if this is a response to an invite, order our offers properly */
   if (!AST_LIST_EMPTY(&p->offered_media)) {
      AST_LIST_TRAVERSE(&p->offered_media, offer, next) {
         switch (offer->type) {
         case SDP_AUDIO:
            if (needaudio) {
               add_content(resp, ast_str_buffer(m_audio));
               add_content(resp, ast_str_buffer(a_audio));
               add_content(resp, hold);
               if (a_crypto) {
                  add_content(resp, a_crypto);
               }
            } else {
               add_content(resp, offer->decline_m_line);
            }
            break;
         case SDP_VIDEO:
            if (needvideo) { /* only if video response is appropriate */
               add_content(resp, ast_str_buffer(m_video));
               add_content(resp, ast_str_buffer(a_video));
               add_content(resp, hold);   /* Repeat hold for the video stream */
               if (v_a_crypto) {
                  add_content(resp, v_a_crypto);
               }
            } else {
               add_content(resp, offer->decline_m_line);
            }
            break;
         case SDP_TEXT:
            if (needtext) { /* only if text response is appropriate */
               add_content(resp, ast_str_buffer(m_text));
               add_content(resp, ast_str_buffer(a_text));
               add_content(resp, hold);   /* Repeat hold for the text stream */
               if (t_a_crypto) {
                  add_content(resp, t_a_crypto);
               }
            } else {
               add_content(resp, offer->decline_m_line);
            }
            break;
         case SDP_IMAGE:
            if (add_t38) {
               add_content(resp, ast_str_buffer(m_modem));
               add_content(resp, ast_str_buffer(a_modem));
            } else {
               add_content(resp, offer->decline_m_line);
            }
            break;
         case SDP_UNKNOWN:
            add_content(resp, offer->decline_m_line);
            break;
         }
      }
   } else {
      /* generate new SDP from scratch, no offers */
      if (needaudio) {
         add_content(resp, ast_str_buffer(m_audio));
         add_content(resp, ast_str_buffer(a_audio));
         add_content(resp, hold);
         if (a_crypto) {
            add_content(resp, a_crypto);
         }
      }
      if (needvideo) { /* only if video response is appropriate */
         add_content(resp, ast_str_buffer(m_video));
         add_content(resp, ast_str_buffer(a_video));
         add_content(resp, hold);   /* Repeat hold for the video stream */
         if (v_a_crypto) {
            add_content(resp, v_a_crypto);
         }
      }
      if (needtext) { /* only if text response is appropriate */
         add_content(resp, ast_str_buffer(m_text));
         add_content(resp, ast_str_buffer(a_text));
         add_content(resp, hold);   /* Repeat hold for the text stream */
         if (t_a_crypto) {
            add_content(resp, t_a_crypto);
         }
      }
      if (add_t38) {
         add_content(resp, ast_str_buffer(m_modem));
         add_content(resp, ast_str_buffer(a_modem));
      }
   }

   /* Update lastrtprx when we send our SDP */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   /*
    * We unlink this dialog and link again into the
    * dialogs_rtpcheck container so its not in there twice.
    */
   ao2_lock(dialogs_rtpcheck);
   ao2_t_unlink(dialogs_rtpcheck, p, "unlink pvt into dialogs_rtpcheck container");
   ao2_t_link(dialogs_rtpcheck, p, "link pvt into dialogs_rtpcheck container");
   ao2_unlock(dialogs_rtpcheck);

   ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap));

add_sdp_cleanup:
   ast_free(a_text);
   ast_free(a_video);
   ast_free(a_audio);
   alreadysent = ast_format_cap_destroy(alreadysent);
   tmpcap = ast_format_cap_destroy(tmpcap);

   return res;
}

static int add_sip_domain	(	const char *	domain,
		const enum domain_mode	mode,
		const char *	context
	)		[static]

Add SIP domain to list of domains we are responsible for.

Definition at line 30275 of file chan_sip.c.

References ast_calloc, ast_copy_string(), ast_debug, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_UNLOCK, ast_log(), ast_strlen_zero(), and LOG_WARNING.

Referenced by reload_config().

{
   struct domain *d;

   if (ast_strlen_zero(domain)) {
      ast_log(LOG_WARNING, "Zero length domain.\n");
      return 1;
   }

   if (!(d = ast_calloc(1, sizeof(*d))))
      return 0;

   ast_copy_string(d->domain, domain, sizeof(d->domain));

   if (!ast_strlen_zero(context))
      ast_copy_string(d->context, context, sizeof(d->context));

   d->mode = mode;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_INSERT_TAIL(&domain_list, d, list);
   AST_LIST_UNLOCK(&domain_list);

   if (sipdebug)
      ast_debug(1, "Added local SIP domain '%s'\n", domain);

   return 1;
}

static int add_supported	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)		[static]

Add "Supported" header to sip message. Since some options may be disabled in the config, the sip_pvt must be inspected to determine what is supported for this dialog.

Definition at line 11381 of file chan_sip.c.

References add_header(), and st_get_mode().

Referenced by respprep(), transmit_invite(), transmit_notify_with_sipfrag(), transmit_reinvite_with_sdp(), and update_connectedline().

{
   int res;
   if (st_get_mode(pvt, 0) != SESSION_TIMER_MODE_REFUSE) {
      res = add_header(req, "Supported", "replaces, timer");
   } else {
      res = add_header(req, "Supported", "replaces");
   }
   return res;
}

static void add_tcodec_to_sdp	(	const struct sip_pvt *	p,
		struct ast_format *	format,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)		[static]

Add text codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 12847 of file chan_sip.c.

References ast_format_set(), AST_FORMAT_T140, AST_FORMAT_T140RED, ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), ast_verbose(), and ast_format::id.

Referenced by add_sdp().

{
   int rtp_code;

   if (!p->trtp)
      return;

   if (debug)
      ast_verbose("Adding text codec %d (%s) to SDP\n", format->id, ast_getformatname(format));

   if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -1)
      return;

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
             ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
             ast_rtp_lookup_sample_rate2(1, format, 0));
   /* Add fmtp code here */

   if (format->id == AST_FORMAT_T140RED) {
      struct ast_format tmp_fmt;
      int t140code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, ast_format_set(&tmp_fmt, AST_FORMAT_T140, 0), 0);
      ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
          t140code,
          t140code,
          t140code);

   }
}

static int add_text	(	struct sip_request *	req,
		struct sip_pvt *	p
	)		[static]

Add text body to SIP message.

Definition at line 12445 of file chan_sip.c.

References add_content(), add_header(), AST_LIST_TRAVERSE, and ast_strlen_zero().

Referenced by transmit_message(), and transmit_request_with_auth().

{
   const char *content_type = NULL;
   struct sip_msg_hdr *node;

   /* Add any additional MESSAGE headers. */
   AST_LIST_TRAVERSE(&p->msg_headers, node, next) {
      if (!strcasecmp(node->name, "Content-Type")) {
         /* Save content type */
         content_type = node->value;
      } else {
         add_header(req, node->name, node->value);
      }
   }
   if (ast_strlen_zero(content_type)) {
      /* "Content-Type" not set - use default value */
      content_type = "text/plain;charset=UTF-8";
   }
   add_header(req, "Content-Type", content_type);

   /* XXX Convert \n's to \r\n's XXX */
   add_content(req, p->msg_body);
   return 0;
}

static struct ast_variable* add_var	(	const char *	buf,
		struct ast_variable *	list
	)		[static, read]

implement the setvar config line

Definition at line 30455 of file chan_sip.c.

References ast_variable_new(), and ast_variable::next.

Referenced by build_peer().

{
   struct ast_variable *tmpvar = NULL;
   char *varname = ast_strdupa(buf), *varval = NULL;
   
   if ((varval = strchr(varname, '='))) {
      *varval++ = '\0';
      if ((tmpvar = ast_variable_new(varname, varval, ""))) {
         tmpvar->next = list;
         list = tmpvar;
      }
   }
   return list;
}

static void add_vcodec_to_sdp	(	const struct sip_pvt *	p,
		struct ast_format *	format,
		struct ast_str **	m_buf,
		struct ast_str **	a_buf,
		int	debug,
		int *	min_packet_size
	)		[static]

Add video codec offer to SDP offer/answer body in INVITE or 200 OK.

Definition at line 12820 of file chan_sip.c.

References ast_format_sdp_generate(), ast_getformatname(), ast_rtp_codecs_payload_code(), ast_rtp_instance_get_codecs(), ast_rtp_lookup_mime_subtype2(), ast_rtp_lookup_sample_rate2(), ast_str_append(), ast_verbose(), and ast_format::id.

Referenced by add_sdp().

{
   int rtp_code;
   const char *subtype;
   unsigned int rate;

   if (!p->vrtp)
      return;

   if (debug)
      ast_verbose("Adding video codec %d (%s) to SDP\n", format->id, ast_getformatname(format));

   if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -1) ||
       !(subtype = ast_rtp_lookup_mime_subtype2(1, format, 0, 0)) ||
       !(rate = ast_rtp_lookup_sample_rate2(1, format, 0))) {
      return;
   }

   ast_str_append(m_buf, 0, " %d", rtp_code);
   ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, subtype, rate);

   ast_format_sdp_generate(format, rtp_code, a_buf);
}

static int add_vidupdate

(

struct sip_request *

req

)

[static]

add XML encoded media control with update

Note:

XML: The only way to turn 0 bits of information into a few hundred. (markster)

Definition at line 12623 of file chan_sip.c.

References add_content(), and add_header().

Referenced by transmit_info_with_vidupdate().

{
   const char *xml_is_a_huge_waste_of_space =
      "<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
      " <media_control>\r\n"
      "  <vc_primitive>\r\n"
      "   <to_encoder>\r\n"
      "    <picture_fast_update>\r\n"
      "    </picture_fast_update>\r\n"
      "   </to_encoder>\r\n"
      "  </vc_primitive>\r\n"
      " </media_control>\r\n";
   add_header(req, "Content-Type", "application/media_control+xml");
   add_content(req, xml_is_a_huge_waste_of_space);
   return 0;
}

static int allow_notify_user_presence

(

struct sip_pvt *

p

)

[static]

Definition at line 14454 of file chan_sip.c.

Referenced by cb_extensionstate(), handle_request_subscribe(), and state_notify_build_xml().

{
   return (strstr(p->useragent, "Digium")) ? 1 : 0;
}

static const char * allowoverlap2str

(

int

mode

)

[static]

Convert AllowOverlap setting to printable string.

Definition at line 19452 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), and sip_show_settings().

{
   return map_x_s(allowoverlapstr, mode, "<error>");
}

static void append_history_full	(	struct sip_pvt *	p,
		const char *	fmt,
			...
	)		[static]

Append to SIP dialog history with arg list.

Definition at line 4065 of file chan_sip.c.

References append_history_va().

{
   va_list ap;

   if (!p) {
      return;
   }

   if (!p->do_history && !recordhistory && !dumphistory) {
      return;
   }

   va_start(ap, fmt);
   append_history_va(p, fmt, ap);
   va_end(ap);

   return;
}

static void append_history_va	(	struct sip_pvt *	p,
		const char *	fmt,
		va_list	ap
	)		[static]

Append to SIP dialog history with arg list.

Definition at line 4037 of file chan_sip.c.

References ast_calloc, ast_free, AST_LIST_INSERT_TAIL, and AST_LIST_REMOVE_HEAD.

Referenced by append_history_full().

{
   char buf[80], *c = buf; /* max history length */
   struct sip_history *hist;
   int l;

   vsnprintf(buf, sizeof(buf), fmt, ap);
   strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
   l = strlen(buf) + 1;
   if (!(hist = ast_calloc(1, sizeof(*hist) + l))) {
      return;
   }
   if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
      ast_free(hist);
      return;
   }
   memcpy(hist->event, buf, l);
   if (p->history_entries == MAX_HISTORY_ENTRIES) {
      struct sip_history *oldest;
      oldest = AST_LIST_REMOVE_HEAD(p->history, list);
      p->history_entries--;
      ast_free(oldest);
   }
   AST_LIST_INSERT_TAIL(p->history, hist, list);
   p->history_entries++;
}

static int apply_directmedia_acl	(	struct sip_pvt *	p,
		struct ast_acl_list *	directmediaacl,
		const char *	op
	)		[static]

Definition at line 32580 of file chan_sip.c.

References ast_apply_acl(), ast_debug, ast_rtp_instance_get_local_address(), ast_rtp_instance_get_remote_address(), AST_SENSE_ALLOW, AST_SENSE_DENY, and ast_sockaddr_stringify().

Referenced by sip_allow_anyrtp_remote().

{
   struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
   int res = AST_SENSE_ALLOW;

   ast_rtp_instance_get_remote_address(p->rtp, &them);
   ast_rtp_instance_get_local_address(p->rtp, &us);

   if ((res = ast_apply_acl(directmediaacl, &them, "SIP Direct Media ACL: ")) == AST_SENSE_DENY) {
      const char *us_addr = ast_strdupa(ast_sockaddr_stringify(&us));
      const char *them_addr = ast_strdupa(ast_sockaddr_stringify(&them));

      ast_debug(3, "Reinvite %s to %s denied by directmedia ACL on %s\n",
         op, them_addr, us_addr);
   }

   return res;
}

AST_DATA_STRUCTURE	(	sip_peer	,
		DATA_EXPORT_SIP_PEER
	)

static void ast_quiet_chan

(

struct ast_channel *

chan

)

[static]

Turn off generator data XXX Does this function belong in the SIP channel?

Definition at line 24458 of file chan_sip.c.

References ast_channel_flags(), ast_channel_generatordata(), ast_deactivate_generator(), AST_FLAG_MOH, ast_moh_stop(), AST_STATE_UP, and ast_test_flag.

Referenced by attempt_transfer(), and handle_invite_replaces().

{
   if (chan && ast_channel_state(chan) == AST_STATE_UP) {
      if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
         ast_moh_stop(chan);
      else if (ast_channel_generatordata(chan))
         ast_deactivate_generator(chan);
   }
}

static void ast_sip_ouraddrfor	(	const struct ast_sockaddr *	them,
		struct ast_sockaddr *	us,
		struct sip_pvt *	p
	)		[static]

NAT fix - decide which IP address to use for Asterisk server?

Using the localaddr structure built up with localnet statements in sip.conf apply it to their address to see if we need to substitute our externaddr or can get away with our internal bindaddr 'us' is always overwritten.

Definition at line 3933 of file chan_sip.c.

References ast_apply_ha(), ast_debug, ast_log(), ast_ouraddrfor(), AST_SENSE_ALLOW, ast_sockaddr_copy(), ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_stringify(), bindaddr, externaddr, externrefresh, internip, ast_tcptls_session_args::local_address, LOG_NOTICE, LOG_WARNING, sip_cfg, and sip_get_transport().

Referenced by __sip_subscribe_mwi_do(), sip_alloc(), sip_cc_monitor_request_cc(), sip_cli_notify(), sip_msg_send(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_publish(), transmit_register(), and transmit_response_using_temp().

{
   struct ast_sockaddr theirs;

   /* Set want_remap to non-zero if we want to remap 'us' to an externally
    * reachable IP address and port. This is done if:
    * 1. we have a localaddr list (containing 'internal' addresses marked
    *    as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
    *    and AST_SENSE_ALLOW on 'external' ones);
    * 2. externaddr is set, so we know what to use as the
    *    externally visible address;
    * 3. the remote address, 'them', is external;
    * 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
    *    when passed to ast_apply_ha() so it does need to be remapped.
    *    This fourth condition is checked later.
    */
   int want_remap = 0;

   ast_sockaddr_copy(us, &internip); /* starting guess for the internal address */
   /* now ask the system what would it use to talk to 'them' */
   ast_ouraddrfor(them, us);
   ast_sockaddr_copy(&theirs, them);

   if (ast_sockaddr_is_ipv6(&theirs)) {
      if (localaddr && !ast_sockaddr_isnull(&externaddr) && !ast_sockaddr_is_any(&bindaddr)) {
         ast_log(LOG_WARNING, "Address remapping activated in sip.conf "
            "but we're using IPv6, which doesn't need it. Please "
            "remove \"localnet\" and/or \"externaddr\" settings.\n");
      }
   } else {
      want_remap = localaddr &&
         !ast_sockaddr_isnull(&externaddr) &&
         ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;
   }

   if (want_remap &&
       (!sip_cfg.matchexternaddrlocally || !ast_apply_ha(localaddr, us)) ) {
      /* if we used externhost, see if it is time to refresh the info */
      if (externexpire && time(NULL) >= externexpire) {
         if (ast_sockaddr_resolve_first(&externaddr, externhost, 0)) {
            ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
         }
         externexpire = time(NULL) + externrefresh;
      }
      if (!ast_sockaddr_isnull(&externaddr)) {
         ast_sockaddr_copy(us, &externaddr);
         switch (p->socket.type) {
         case SIP_TRANSPORT_TCP:
            if (!externtcpport && ast_sockaddr_port(&externaddr)) {
               /* for consistency, default to the externaddr port */
               externtcpport = ast_sockaddr_port(&externaddr);
            }
            ast_sockaddr_set_port(us, externtcpport);
            break;
         case SIP_TRANSPORT_TLS:
            ast_sockaddr_set_port(us, externtlsport);
            break;
         case SIP_TRANSPORT_UDP:
            if (!ast_sockaddr_port(&externaddr)) {
               ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
            }
            break;
         default:
            break;
         }
      }
      ast_debug(1, "Target address %s is not local, substituting externaddr\n",
           ast_sockaddr_stringify(them));
   } else {
      /* no remapping, but we bind to a specific address, so use it. */
      switch (p->socket.type) {
      case SIP_TRANSPORT_TCP:
         if (!ast_sockaddr_is_any(&sip_tcp_desc.local_address)) {
            ast_sockaddr_copy(us,
                    &sip_tcp_desc.local_address);
         } else {
            ast_sockaddr_set_port(us,
                        ast_sockaddr_port(&sip_tcp_desc.local_address));
         }
         break;
      case SIP_TRANSPORT_TLS:
         if (!ast_sockaddr_is_any(&sip_tls_desc.local_address)) {
            ast_sockaddr_copy(us,
                    &sip_tls_desc.local_address);
         } else {
            ast_sockaddr_set_port(us,
                        ast_sockaddr_port(&sip_tls_desc.local_address));
         }
         break;
      case SIP_TRANSPORT_UDP:
         /* fall through on purpose */
      default:
         if (!ast_sockaddr_is_any(&bindaddr)) {
            ast_sockaddr_copy(us, &bindaddr);
         }
         if (!ast_sockaddr_port(us)) {
            ast_sockaddr_set_port(us, ast_sockaddr_port(&bindaddr));
         }
      }
   }
   ast_debug(3, "Setting SIP_TRANSPORT_%s with address %s\n", sip_get_transport(p->socket.type), ast_sockaddr_stringify(us));
}

static int ast_sockaddr_resolve_first	(	struct ast_sockaddr *	addr,
		const char *	name,
		int	flag
	)		[static]

Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.

Using this function probably means you have a faulty design.

Definition at line 33457 of file chan_sip.c.

References ast_sockaddr_resolve_first_af(), and get_address_family_filter().

Referenced by ast_sip_ouraddrfor(), check_via(), and reload_config().

{
   return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(SIP_TRANSPORT_UDP));
}

static int ast_sockaddr_resolve_first_af	(	struct ast_sockaddr *	addr,
		const char *	name,
		int	flag,
		int	family
	)		[static]

Return the first entry from ast_sockaddr_resolve filtered by address family.

Using this function probably means you have a faulty design.

Definition at line 33433 of file chan_sip.c.

References ast_debug, ast_free, ast_sockaddr_copy(), and ast_sockaddr_resolve().

Referenced by ast_sockaddr_resolve_first(), ast_sockaddr_resolve_first_transport(), get_ip_and_port_from_sdp(), process_sdp_c(), and sip_do_debug_ip().

{
   struct ast_sockaddr *addrs;
   int addrs_cnt;

   addrs_cnt = ast_sockaddr_resolve(&addrs, name, flag, family);
   if (addrs_cnt <= 0) {
      return 1;
   }
   if (addrs_cnt > 1) {
      ast_debug(1, "Multiple addresses, using the first one only\n");
   }

   ast_sockaddr_copy(addr, &addrs[0]);

   ast_free(addrs);
   return 0;
}

static int ast_sockaddr_resolve_first_transport	(	struct ast_sockaddr *	addr,
		const char *	name,
		int	flag,
		unsigned int	transport
	)		[static]

Return the first entry from ast_sockaddr_resolve filtered by family of binddaddr.

Using this function probably means you have a faulty design.

Definition at line 33467 of file chan_sip.c.

References ast_sockaddr_resolve_first_af(), and get_address_family_filter().

Referenced by __set_address_from_contact(), create_addr(), parse_register_contact(), process_via(), and set_destination().

{
        return ast_sockaddr_resolve_first_af(addr, name, flag, get_address_family_filter(transport));
}

static int attempt_transfer	(	struct sip_dual *	transferer,
		struct sip_dual *	target
	)		[static]

Attempt transfer of SIP call This fix for attended transfers on a local PBX.

Definition at line 24470 of file chan_sip.c.

References ast_channel_masquerade(), ast_channel_name(), ast_debug, ast_log(), ast_quiet_chan(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_state2str(), LOG_NOTICE, and LOG_WARNING.

Referenced by local_attended_transfer().

{
   int res = 0;
   struct ast_channel *peera = NULL,
      *peerb = NULL,
      *peerc = NULL,
      *peerd = NULL;


   /* We will try to connect the transferee with the target and hangup
      all channels to the transferer */
   ast_debug(4, "Sip transfer:--------------------\n");
   if (transferer->chan1)
      ast_debug(4, "-- Transferer to PBX channel: %s State %s\n", ast_channel_name(transferer->chan1), ast_state2str(ast_channel_state(transferer->chan1)));
   else
      ast_debug(4, "-- No transferer first channel - odd??? \n");
   if (target->chan1)
      ast_debug(4, "-- Transferer to PBX second channel (target): %s State %s\n", ast_channel_name(target->chan1), ast_state2str(ast_channel_state(target->chan1)));
   else
      ast_debug(4, "-- No target first channel ---\n");
   if (transferer->chan2)
      ast_debug(4, "-- Bridged call to transferee: %s State %s\n", ast_channel_name(transferer->chan2), ast_state2str(ast_channel_state(transferer->chan2)));
   else
      ast_debug(4, "-- No bridged call to transferee\n");
   if (target->chan2)
      ast_debug(4, "-- Bridged call to transfer target: %s State %s\n", target->chan2 ? ast_channel_name(target->chan2) : "<none>", target->chan2 ? ast_state2str(ast_channel_state(target->chan2)) : "(none)");
   else
      ast_debug(4, "-- No target second channel ---\n");
   ast_debug(4, "-- END Sip transfer:--------------------\n");
   if (transferer->chan2) { /* We have a bridge on the transferer's channel */
      peera = transferer->chan1; /* Transferer - PBX -> transferee channel * the one we hangup */
      peerb = target->chan1;     /* Transferer - PBX -> target channel - This will get lost in masq */
      peerc = transferer->chan2; /* Asterisk to Transferee */
      peerd = target->chan2;     /* Asterisk to Target */
      ast_debug(3, "SIP transfer: Four channels to handle\n");
   } else if (target->chan2) {   /* Transferer has no bridge (IVR), but transferee */
      peera = target->chan1;     /* Transferer to PBX -> target channel */
      peerb = transferer->chan1; /* Transferer to IVR*/
      peerc = target->chan2;     /* Asterisk to Target */
      peerd = transferer->chan2; /* Nothing */
      ast_debug(3, "SIP transfer: Three channels to handle\n");
   }

   if (peera && peerb && peerc && (peerb != peerc)) {
      ast_quiet_chan(peera);     /* Stop generators */
      /* no need to quiet peerb since it should be hungup after the
         transfer and the masquerade needs to be able to see if MOH is
         playing on it */
      ast_quiet_chan(peerc);
      if (peerd)
         ast_quiet_chan(peerd);

      ast_debug(4, "SIP transfer: trying to masquerade %s into %s\n", ast_channel_name(peerc), ast_channel_name(peerb));
      if (ast_channel_masquerade(peerb, peerc)) {
         ast_log(LOG_WARNING, "Failed to masquerade %s into %s\n", ast_channel_name(peerb), ast_channel_name(peerc));
         res = -1;
      } else
         ast_debug(4, "SIP transfer: Succeeded to masquerade channels.\n");
      return res;
   } else {
      ast_log(LOG_NOTICE, "SIP Transfer attempted with no appropriate bridged calls to transfer\n");
      if (transferer->chan1)
         ast_softhangup_nolock(transferer->chan1, AST_SOFTHANGUP_DEV);
      if (target->chan1)
         ast_softhangup_nolock(target->chan1, AST_SOFTHANGUP_DEV);
      return -1;
   }
   return 0;
}

static int auto_congest

(

const void *

arg

)

[static]

Scheduled congestion on a call. Only called by the scheduler, must return the reference when done.

Definition at line 6230 of file chan_sip.c.

References append_history, ast_channel_trylock, ast_channel_unlock, AST_CONTROL_CONGESTION, ast_queue_control(), sip_pvt_lock, sip_pvt_unlock, and sip_scheddestroy().

Referenced by sip_call(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *)arg;

   sip_pvt_lock(p);
   p->initid = -1;   /* event gone, will not be rescheduled */
   if (p->owner) {
      /* XXX fails on possible deadlock */
      if (!ast_channel_trylock(p->owner)) {
         append_history(p, "Cong", "Auto-congesting (timer)");
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         ast_channel_unlock(p->owner);
      }

      /* Give the channel a chance to act before we proceed with destruction */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }
   sip_pvt_unlock(p);
   dialog_unref(p, "unreffing arg passed into auto_congest callback (p->initid)");
   return 0;
}

static const char* autocreatepeer2str

(

enum autocreatepeer_mode

r

)

[static]

Definition at line 18873 of file chan_sip.c.

References map_x_s().

Referenced by sip_show_settings().

{
   return map_x_s(autopeermodes, r, "Unknown");
}

static int block_msg_header

(

const char *

header_name

)

[static]

Definition at line 26964 of file chan_sip.c.

References ARRAY_LEN.

Referenced by sip_msg_send().

{
   int idx;

   /*
    * Don't block Content-Type or Max-Forwards headers because the
    * user can override them.
    */
   static const char *hdr[] = {
      "To",
      "From",
      "Via",
      "Route",
      "Contact",
      "Call-ID",
      "CSeq",
      "Allow",
      "Content-Length",
      "Request-URI",
   };

   for (idx = 0; idx < ARRAY_LEN(hdr); ++idx) {
      if (!strcasecmp(header_name, hdr[idx])) {
         /* Block addition of this header. */
         return 1;
      }
   }
   return 0;
}

static void build_callid_pvt

(

struct sip_pvt *

pvt

)

[static]

Build SIP Call-ID value for a non-REGISTER transaction.

Note:

The passed in pvt must not be in a dialogs container since this function changes the hash key used by the container.

Definition at line 8514 of file chan_sip.c.

References ast_sockaddr_stringify_remote(), ast_string_field_build, generate_random_string(), and S_OR.

Referenced by change_callid_pvt(), and sip_alloc().

{
   char buf[33];
   const char *host = S_OR(pvt->fromdomain, ast_sockaddr_stringify_remote(&pvt->ourip));

   ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}

static void build_callid_registry	(	struct sip_registry *	reg,
		const struct ast_sockaddr *	ourip,
		const char *	fromdomain
	)		[static]

Build SIP Call-ID value for a REGISTER transaction.

Definition at line 8577 of file chan_sip.c.

References ast_sockaddr_stringify_host_remote(), ast_string_field_build, generate_random_string(), and S_OR.

Referenced by transmit_register().

{
   char buf[33];

   const char *host = S_OR(fromdomain, ast_sockaddr_stringify_host_remote(ourip));

   ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}

static void build_contact

(

struct sip_pvt *

p

)

[static]

Build contact header - the contact header we send out.

Definition at line 13793 of file chan_sip.c.

References ast_sockaddr_stringify_remote(), ast_string_field_build, ast_strlen_zero(), ast_uri_encode(), ast_uri_sip_user, and sip_get_transport().

Referenced by __sip_subscribe_mwi_do(), check_user_full(), handle_request_invite(), handle_request_options(), handle_request_subscribe(), initreqprep(), register_verify(), and transmit_register().

{
   char tmp[SIPBUFSIZE];
   char *user = ast_uri_encode(p->exten, tmp, sizeof(tmp), ast_uri_sip_user);

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      ast_string_field_build(p, our_contact, "<sip:%s%s%s>", user,
         ast_strlen_zero(user) ? "" : "@", ast_sockaddr_stringify_remote(&p->ourip));
   } else {
      ast_string_field_build(p, our_contact, "<sip:%s%s%s;transport=%s>", user,
         ast_strlen_zero(user) ? "" : "@", ast_sockaddr_stringify_remote(&p->ourip),
         sip_get_transport(p->socket.type));
   }
}

static void build_localtag_registry

(

struct sip_registry *

reg

)

[static]

Build SIP From tag value for REGISTER.

Definition at line 8587 of file chan_sip.c.

References ast_random(), and ast_string_field_build.

Referenced by transmit_register().

{
   ast_string_field_build(reg, localtag, "as%08lx", ast_random());
}

static void build_nonce	(	struct sip_pvt *	p,
		int	forceupdate
	)		[static]

builds the sip_pvt's nonce field which is used for the authentication challenge. When forceupdate is not set, the nonce is only updated if the current one is stale. In this case, a stalenonce is one which has already received a response, if a nonce has not received a response it is not always necessary or beneficial to create a new one.

Definition at line 16360 of file chan_sip.c.

References ast_random(), ast_string_field_build, and ast_strlen_zero().

Referenced by check_auth(), and transmit_fake_auth_response().

{
   if (p->stalenonce || forceupdate || ast_strlen_zero(p->nonce)) {
      ast_string_field_build(p, nonce, "%08lx", ast_random()); /* Create nonce for challenge */
      p->stalenonce = 0;
   }
}

static struct sip_peer * build_peer	(	const char *	name,
		struct ast_variable *	v,
		struct ast_variable *	alt,
		int	realtime,
		int	devstate_only
	)		[static, read]

Build peer from configuration (file or realtime static/dynamic)

< The first transport listed should be default outbound

Definition at line 30621 of file chan_sip.c.

References __set_address_from_contact(), accountcode, acl_change_event_subscribe(), add_peer_mailboxes(), add_peer_mwi_subs(), add_realm_authentication(), add_var(), ao2_lock, ao2_t_alloc, ao2_t_find, ao2_t_ref, ao2_t_unlink, ao2_unlock, AST_AES_CM_128_HMAC_SHA1_32, AST_AES_CM_128_HMAC_SHA1_80, ast_append_acl(), ast_asprintf, ast_atomic_fetchadd_int(), ast_callerid_split(), AST_CC_AGENT_NATIVE, AST_CC_AGENT_NEVER, ast_cc_config_params_init, ast_cc_is_config_param(), ast_cc_set_param(), ast_cdr_amaflags2int(), ast_copy_flags, ast_copy_string(), ast_debug, ast_dnsmgr_lookup_cb(), ast_dnsmgr_refresh(), ast_format_cap_alloc_nolock(), ast_free, ast_free_acl_list(), ast_get_cc_agent_policy(), ast_get_group(), ast_get_indication_zone(), ast_get_ip(), ast_get_namedgroups(), ast_get_time_t(), AST_LIST_EMPTY, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, ast_log(), ast_parse_allow_disallow(), ast_parse_caller_presentation(), ast_rtp_dtls_cfg_parse(), AST_SCHED_DEL_UNREF, ast_set2_flag, ast_set_cc_agent_policy(), ast_set_flag, ast_skip_blanks(), ast_sockaddr_isnull(), ast_sockaddr_parse(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_setnull(), ast_sockaddr_stringify_addr(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_reset(), ast_str_set(), ast_str_strlen(), ast_string_field_init, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_tone_zone_unref(), ast_true(), ast_unref_namedgroups(), ast_variables_destroy(), cid_name, cid_num, context, default_maxcallbitrate, DEFAULT_MAXMS, default_primary_transport, default_transports, destroy_association(), destroy_mailbox(), ast_tls_config::enabled, FALSE, ast_flags::flags, format, get_address_family_filter(), get_srv_protocol(), get_srv_service(), global_max_se, global_min_se, global_qualifyfreq, global_rtpholdtimeout, global_rtpkeepalive, global_rtptimeout, global_st_mode, global_st_refresher, global_t1min, global_timer_b, handle_common_options(), handle_t38_options(), language, ast_variable::lineno, LOG_ERROR, LOG_NOTICE, LOG_WARNING, mailbox, mark_parsed_methods(), mohinterpret, mohsuggest, ast_variable::name, ast_variable::next, OBJ_POINTER, OBJ_UNLINK, on_dns_update_peer(), parkinglot, PARSE_PORT_FORBID, port_str2int(), proxy_from_config(), reg_source_db(), secret, set_peer_defaults(), set_socket_transport(), sip_cfg, sip_destroy_peer_fn(), sip_poke_peer(), sip_ref_peer(), sip_register(), sip_send_mwi_to_peer(), sip_unref_peer(), srvlookup, str2stmode(), str2strefresherparam(), TRUE, ast_variable::value, and vmexten.

Referenced by realtime_peer(), register_realtime_peers_with_callbackextens(), and reload_config().

{
   struct sip_peer *peer = NULL;
   struct ast_acl_list *oldacl = NULL;
   struct ast_acl_list *olddirectmediaacl = NULL;
   int found = 0;
   int firstpass = 1;
   uint16_t port = 0;
   int format = 0;      /* Ama flags */
   int timerb_set = 0, timert1_set = 0;
   time_t regseconds = 0;
   struct ast_flags peerflags[3] = {{(0)}};
   struct ast_flags mask[3] = {{(0)}};
   struct sip_peer tmp_peer;
   const char *srvlookup = NULL;
   static int deprecation_warning = 1;
   int alt_fullcontact = alt ? 1 : 0, headercount = 0;
   struct ast_str *fullcontact = ast_str_alloca(512);
   int acl_change_subscription_needed = 0;

   if (!realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      /* Note we do NOT use sip_find_peer here, to avoid realtime recursion */
      /* We also use a case-sensitive comparison (unlike sip_find_peer) so
         that case changes made to the peer name will be properly handled
         during reload
      */
      ast_copy_string(tmp_peer.name, name, sizeof(tmp_peer.name));
      peer = ao2_t_find(peers, &tmp_peer, OBJ_POINTER | OBJ_UNLINK, "find and unlink peer from peers table");
   }

   if (peer) {
      /* Already in the list, remove it and it will be added back (or FREE'd)  */
      found++;
      /* we've unlinked the peer from the peers container but not unlinked from the peers_by_ip container yet
        this leads to a wrong refcounter and the peer object is never destroyed */
      if (!ast_sockaddr_isnull(&peer->addr)) {
         ao2_t_unlink(peers_by_ip, peer, "ao2_unlink peer from peers_by_ip table");
      }
      if (!(peer->the_mark))
         firstpass = 0;
   } else {
      if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct"))) {
         return NULL;
      }
      if (!(peer->caps = ast_format_cap_alloc_nolock())) {
         ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer");
         return NULL;
      }
      if (ast_string_field_init(peer, 512)) {
         ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
         return NULL;
      }

      if (!(peer->cc_params = ast_cc_config_params_init())) {
         ao2_t_ref(peer, -1, "failed to allocate cc_params for peer");
         return NULL;
      }

      if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
         ast_atomic_fetchadd_int(&rpeerobjs, 1);
         ast_debug(3, "-REALTIME- peer built. Name: %s. Peer objects: %d\n", name, rpeerobjs);
      } else
         ast_atomic_fetchadd_int(&speerobjs, 1);
   }

   /* Note that our peer HAS had its reference count increased */
   if (firstpass) {
      oldacl = peer->acl;
      peer->acl = NULL;
      olddirectmediaacl = peer->directmediaacl;
      peer->directmediaacl = NULL;
      set_peer_defaults(peer);   /* Set peer defaults */
      peer->type = 0;
   }

   /* in case the case of the peer name has changed, update the name */
   ast_copy_string(peer->name, name, sizeof(peer->name));

   /* If we have channel variables, remove them (reload) */
   if (peer->chanvars) {
      ast_variables_destroy(peer->chanvars);
      peer->chanvars = NULL;
      /* XXX should unregister ? */
   }

   if (found)
      peer->portinuri = 0;

   /* If we have realm authentication information, remove them (reload) */
   ao2_lock(peer);
   if (peer->auth) {
      ao2_t_ref(peer->auth, -1, "Removing old peer authentication");
      peer->auth = NULL;
   }
   ao2_unlock(peer);

   /* clear the transport information.  We will detect if a default value is required after parsing the config */
   peer->default_outbound_transport = 0;
   peer->transports = 0;

   if (!devstate_only) {
      struct sip_mailbox *mailbox;
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         mailbox->delme = 1;
      }
   }

   /* clear named callgroup and named pickup group container */
   peer->named_callgroups = ast_unref_namedgroups(peer->named_callgroups);
   peer->named_pickupgroups = ast_unref_namedgroups(peer->named_pickupgroups);

   for (; v || ((v = alt) && !(alt=NULL)); v = v->next) {
      if (!devstate_only) {
         if (handle_common_options(&peerflags[0], &mask[0], v)) {
            continue;
         }
         if (handle_t38_options(&peerflags[0], &mask[0], v, &peer->t38_maxdatagram)) {
            continue;
         }
         if (!strcasecmp(v->name, "transport")) {
            char *val = ast_strdupa(v->value);
            char *trans;

            peer->transports = peer->default_outbound_transport = 0;
            while ((trans = strsep(&val, ","))) {
               trans = ast_skip_blanks(trans);

               if (!strncasecmp(trans, "udp", 3)) {
                  peer->transports |= SIP_TRANSPORT_UDP;
               } else if (!strncasecmp(trans, "wss", 3)) {
                  peer->transports |= SIP_TRANSPORT_WSS;
               } else if (!strncasecmp(trans, "ws", 2)) {
                  peer->transports |= SIP_TRANSPORT_WS;
               } else if (sip_cfg.tcp_enabled && !strncasecmp(trans, "tcp", 3)) {
                  peer->transports |= SIP_TRANSPORT_TCP;
               } else if (default_tls_cfg.enabled && !strncasecmp(trans, "tls", 3)) {
                  peer->transports |= SIP_TRANSPORT_TLS;
               } else if (!strncasecmp(trans, "tcp", 3) || !strncasecmp(trans, "tls", 3)) {
                  ast_log(LOG_WARNING, "'%.3s' is not a valid transport type when %.3senable=no. If no other is specified, the defaults from general will be used.\n", trans, trans);
               } else {
                  ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, the defaults from general will be used.\n", trans);
               }

               if (!peer->default_outbound_transport) { /*!< The first transport listed should be default outbound */
                  peer->default_outbound_transport = peer->transports;
               }
            }
         } else if (realtime && !strcasecmp(v->name, "regseconds")) {
            ast_get_time_t(v->value, &regseconds, 0, NULL);
         } else if (realtime && !strcasecmp(v->name, "name")) {
            ast_copy_string(peer->name, v->value, sizeof(peer->name));
         } else if (realtime && !strcasecmp(v->name, "useragent")) {
            ast_string_field_set(peer, useragent, v->value);
         } else if (!strcasecmp(v->name, "type")) {
            if (!strcasecmp(v->value, "peer")) {
               peer->type |= SIP_TYPE_PEER;
            } else if (!strcasecmp(v->value, "user")) {
               peer->type |= SIP_TYPE_USER;
            } else if (!strcasecmp(v->value, "friend")) {
               peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
            }
         } else if (!strcasecmp(v->name, "remotesecret")) {
            ast_string_field_set(peer, remotesecret, v->value);
         } else if (!strcasecmp(v->name, "secret")) {
            ast_string_field_set(peer, secret, v->value);
         } else if (!strcasecmp(v->name, "description")) {
            ast_string_field_set(peer, description, v->value);
         } else if (!strcasecmp(v->name, "md5secret")) {
            ast_string_field_set(peer, md5secret, v->value);
         } else if (!strcasecmp(v->name, "auth")) {
            add_realm_authentication(&peer->auth, v->value, v->lineno);
         } else if (!strcasecmp(v->name, "callerid")) {
            char cid_name[80] = { '\0' }, cid_num[80] = { '\0' };

            ast_callerid_split(v->value, cid_name, sizeof(cid_name), cid_num, sizeof(cid_num));
            ast_string_field_set(peer, cid_name, cid_name);
            ast_string_field_set(peer, cid_num, cid_num);
         } else if (!strcasecmp(v->name, "mwi_from")) {
            ast_string_field_set(peer, mwi_from, v->value);
         } else if (!strcasecmp(v->name, "fullname")) {
            ast_string_field_set(peer, cid_name, v->value);
         } else if (!strcasecmp(v->name, "trunkname")) {
            /* This is actually for a trunk, so we don't want to override callerid */
            ast_string_field_set(peer, cid_name, "");
         } else if (!strcasecmp(v->name, "cid_number")) {
            ast_string_field_set(peer, cid_num, v->value);
         } else if (!strcasecmp(v->name, "cid_tag")) {
            ast_string_field_set(peer, cid_tag, v->value);
         } else if (!strcasecmp(v->name, "context")) {
            ast_string_field_set(peer, context, v->value);
            ast_set_flag(&peer->flags[1], SIP_PAGE2_HAVEPEERCONTEXT);
         } else if (!strcasecmp(v->name, "recordonfeature")) {
            ast_string_field_set(peer, record_on_feature, v->value);
         } else if (!strcasecmp(v->name, "recordofffeature")) {
            ast_string_field_set(peer, record_off_feature, v->value);
         } else if (!strcasecmp(v->name, "outofcall_message_context")) {
            ast_string_field_set(peer, messagecontext, v->value);
         } else if (!strcasecmp(v->name, "subscribecontext")) {
            ast_string_field_set(peer, subscribecontext, v->value);
         } else if (!strcasecmp(v->name, "fromdomain")) {
            char *fromdomainport;
            ast_string_field_set(peer, fromdomain, v->value);
            if ((fromdomainport = strchr(peer->fromdomain, ':'))) {
               *fromdomainport++ = '\0';
               if (!(peer->fromdomainport = port_str2int(fromdomainport, 0))) {
                  ast_log(LOG_NOTICE, "'%s' is not a valid port number for fromdomain.\n",fromdomainport);
               }
            } else {
               peer->fromdomainport = STANDARD_SIP_PORT;
            }
         } else if (!strcasecmp(v->name, "usereqphone")) {
            ast_set2_flag(&peer->flags[0], ast_true(v->value), SIP_USEREQPHONE);
         } else if (!strcasecmp(v->name, "fromuser")) {
            ast_string_field_set(peer, fromuser, v->value);
         } else if (!strcasecmp(v->name, "outboundproxy")) {
            struct sip_proxy *proxy;
            if (ast_strlen_zero(v->value)) {
               ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf\n", v->lineno);
               continue;
            }
            proxy = proxy_from_config(v->value, v->lineno, peer->outboundproxy);
            if (!proxy) {
               ast_log(LOG_WARNING, "failure parsing the outbound proxy on line %d of sip.conf.\n", v->lineno);
               continue;
            }
            peer->outboundproxy = proxy;
         } else if (!strcasecmp(v->name, "host")) {
            if (!strcasecmp(v->value, "dynamic")) {
               /* They'll register with us */
               if ((!found && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)) || !peer->host_dynamic) {
                  /* Initialize stuff if this is a new peer, or if it used to
                   * not be dynamic before the reload. */
                  ast_sockaddr_setnull(&peer->addr);
               }
               peer->host_dynamic = TRUE;
            } else {
               /* Non-dynamic.  Make sure we become that way if we're not */
               AST_SCHED_DEL_UNREF(sched, peer->expire,
                     sip_unref_peer(peer, "removing register expire ref"));
               peer->host_dynamic = FALSE;
               srvlookup = v->value;
            }
         } else if (!strcasecmp(v->name, "defaultip")) {
            if (!ast_strlen_zero(v->value) && ast_get_ip(&peer->defaddr, v->value)) {
               sip_unref_peer(peer, "sip_unref_peer: from build_peer defaultip");
               return NULL;
            }
         } else if (!strcasecmp(v->name, "permit") || !strcasecmp(v->name, "deny") || !strcasecmp(v->name, "acl")) {
            int ha_error = 0;
            if (!ast_strlen_zero(v->value)) {
               ast_append_acl(v->name, v->value, &peer->acl, &ha_error, &acl_change_subscription_needed);
            }
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny") || !strcasecmp(v->name, "contactacl")) {
            int ha_error = 0;
            if (!ast_strlen_zero(v->value)) {
               ast_append_acl(v->name + 7, v->value, &peer->contactacl, &ha_error, &acl_change_subscription_needed);
            }
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "directmediapermit") || !strcasecmp(v->name, "directmediadeny") || !strcasecmp(v->name, "directmediaacl")) {
            int ha_error = 0;
            ast_append_acl(v->name + 11, v->value, &peer->directmediaacl, &ha_error, &acl_change_subscription_needed);
            if (ha_error) {
               ast_log(LOG_ERROR, "Bad directmedia ACL entry in configuration line %d : %s\n", v->lineno, v->value);
            }
         } else if (!strcasecmp(v->name, "port")) {
            peer->portinuri = 1;
            if (!(port = port_str2int(v->value, 0))) {
               if (realtime) {
                  /* If stored as integer, could be 0 for some DBs (notably MySQL) */
                  peer->portinuri = 0;
               } else {
                  ast_log(LOG_WARNING, "Invalid peer port configuration at line %d : %s\n", v->lineno, v->value);
               }
            }
         } else if (!strcasecmp(v->name, "callingpres")) {
            peer->callingpres = ast_parse_caller_presentation(v->value);
            if (peer->callingpres == -1) {
               peer->callingpres = atoi(v->value);
            }
         } else if (!strcasecmp(v->name, "username") || !strcmp(v->name, "defaultuser")) {   /* "username" is deprecated */
            ast_string_field_set(peer, username, v->value);
            if (!strcasecmp(v->name, "username")) {
               if (deprecation_warning) {
                  ast_log(LOG_NOTICE, "The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser'\n");
                  deprecation_warning = 0;
               }
               peer->deprecated_username = 1;
            }
         } else if (!strcasecmp(v->name, "tonezone")) {
            struct ast_tone_zone *new_zone;
            if (!(new_zone = ast_get_indication_zone(v->value))) {
               ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone in device [%s] at line %d. Check indications.conf for available country codes.\n", v->value, peer->name, v->lineno);
            } else {
               ast_tone_zone_unref(new_zone);
               ast_string_field_set(peer, zone, v->value);
            }
         } else if (!strcasecmp(v->name, "language")) {
            ast_string_field_set(peer, language, v->value);
         } else if (!strcasecmp(v->name, "regexten")) {
            ast_string_field_set(peer, regexten, v->value);
         } else if (!strcasecmp(v->name, "callbackextension")) {
            ast_string_field_set(peer, callback, v->value);
         } else if (!strcasecmp(v->name, "amaflags")) {
            format = ast_cdr_amaflags2int(v->value);
            if (format < 0) {
               ast_log(LOG_WARNING, "Invalid AMA Flags for peer: %s at line %d\n", v->value, v->lineno);
            } else {
               peer->amaflags = format;
            }
         } else if (!strcasecmp(v->name, "maxforwards")) {
            if (sscanf(v->value, "%30d", &peer->maxforwards) != 1
               || peer->maxforwards < 1 || 255 < peer->maxforwards) {
               ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d.  Using default.\n", v->value, v->lineno);
               peer->maxforwards = sip_cfg.default_max_forwards;
            }
         } else if (!strcasecmp(v->name, "accountcode")) {
            ast_string_field_set(peer, accountcode, v->value);
         } else if (!strcasecmp(v->name, "mohinterpret")) {
            ast_string_field_set(peer, mohinterpret, v->value);
         } else if (!strcasecmp(v->name, "mohsuggest")) {
            ast_string_field_set(peer, mohsuggest, v->value);
         } else if (!strcasecmp(v->name, "parkinglot")) {
            ast_string_field_set(peer, parkinglot, v->value);
         } else if (!strcasecmp(v->name, "rtp_engine")) {
            ast_string_field_set(peer, engine, v->value);
         } else if (!strcasecmp(v->name, "mailbox")) {
            add_peer_mailboxes(peer, v->value);
         } else if (!strcasecmp(v->name, "hasvoicemail")) {
            /* People expect that if 'hasvoicemail' is set, that the mailbox will
             * be also set, even if not explicitly specified. */
            if (ast_true(v->value) && AST_LIST_EMPTY(&peer->mailboxes)) {
               add_peer_mailboxes(peer, name);
            }
         } else if (!strcasecmp(v->name, "subscribemwi")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_SUBSCRIBEMWIONLY);
         } else if (!strcasecmp(v->name, "vmexten")) {
            ast_string_field_set(peer, vmexten, v->value);
         } else if (!strcasecmp(v->name, "callgroup")) {
            peer->callgroup = ast_get_group(v->value);
         } else if (!strcasecmp(v->name, "allowtransfer")) {
            peer->allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
         } else if (!strcasecmp(v->name, "pickupgroup")) {
            peer->pickupgroup = ast_get_group(v->value);
         } else if (!strcasecmp(v->name, "namedcallgroup")) {
            peer->named_callgroups = ast_get_namedgroups(v->value);
         } else if (!strcasecmp(v->name, "namedpickupgroup")) {
            peer->named_pickupgroups = ast_get_namedgroups(v->value);
         } else if (!strcasecmp(v->name, "allow")) {
            int error = ast_parse_allow_disallow(&peer->prefs, peer->caps, v->value, TRUE);
            if (error) {
               ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
            }
         } else if (!strcasecmp(v->name, "disallow")) {
            int error =  ast_parse_allow_disallow(&peer->prefs, peer->caps, v->value, FALSE);
            if (error) {
               ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
            }
         } else if (!strcasecmp(v->name, "preferred_codec_only")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
         } else if (!strcasecmp(v->name, "autoframing")) {
            peer->autoframing = ast_true(v->value);
         } else if (!strcasecmp(v->name, "rtptimeout")) {
            if ((sscanf(v->value, "%30d", &peer->rtptimeout) != 1) || (peer->rtptimeout < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtptimeout = global_rtptimeout;
            }
         } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
            if ((sscanf(v->value, "%30d", &peer->rtpholdtimeout) != 1) || (peer->rtpholdtimeout < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtpholdtimeout = global_rtpholdtimeout;
            }
         } else if (!strcasecmp(v->name, "rtpkeepalive")) {
            if ((sscanf(v->value, "%30d", &peer->rtpkeepalive) != 1) || (peer->rtpkeepalive < 0)) {
               ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
               peer->rtpkeepalive = global_rtpkeepalive;
            }
         } else if (!strcasecmp(v->name, "timert1")) {
            if ((sscanf(v->value, "%30d", &peer->timer_t1) != 1) || (peer->timer_t1 < 200) || (peer->timer_t1 < global_t1min)) {
               ast_log(LOG_WARNING, "'%s' is not a valid T1 time at line %d.  Using default.\n", v->value, v->lineno);
               peer->timer_t1 = global_t1min;
            }
            timert1_set = 1;
         } else if (!strcasecmp(v->name, "timerb")) {
            if ((sscanf(v->value, "%30d", &peer->timer_b) != 1) || (peer->timer_b < 200)) {
               ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d.  Using default.\n", v->value, v->lineno);
               peer->timer_b = global_timer_b;
            }
            timerb_set = 1;
         } else if (!strcasecmp(v->name, "setvar")) {
            peer->chanvars = add_var(v->value, peer->chanvars);
         } else if (!strcasecmp(v->name, "header")) {
            char tmp[4096];
            snprintf(tmp, sizeof(tmp), "__SIPADDHEADERpre%2d=%s", ++headercount, v->value);
            peer->chanvars = add_var(tmp, peer->chanvars);
         } else if (!strcasecmp(v->name, "qualifyfreq")) {
            int i;
            if (sscanf(v->value, "%30d", &i) == 1) {
               peer->qualifyfreq = i * 1000;
            } else {
               ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->qualifyfreq = global_qualifyfreq;
            }
         } else if (!strcasecmp(v->name, "maxcallbitrate")) {
            peer->maxcallbitrate = atoi(v->value);
            if (peer->maxcallbitrate < 0) {
               peer->maxcallbitrate = default_maxcallbitrate;
            }
         } else if (!strcasecmp(v->name, "session-timers")) {
            int i = (int) str2stmode(v->value);
            if (i < 0) {
               ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_mode_oper = global_st_mode;
            } else {
               peer->stimer.st_mode_oper = i;
            }
         } else if (!strcasecmp(v->name, "session-expires")) {
            if (sscanf(v->value, "%30d", &peer->stimer.st_max_se) != 1) {
               ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_max_se = global_max_se;
            }
         } else if (!strcasecmp(v->name, "session-minse")) {
            if (sscanf(v->value, "%30d", &peer->stimer.st_min_se) != 1) {
               ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_min_se = global_min_se;
            }
            if (peer->stimer.st_min_se < DEFAULT_MIN_SE) {
               ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < %d secs\n", v->value, v->lineno, config, DEFAULT_MIN_SE);
               peer->stimer.st_min_se = global_min_se;
            }
         } else if (!strcasecmp(v->name, "session-refresher")) {
            int i = (int) str2strefresherparam(v->value);
            if (i < 0) {
               ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
               peer->stimer.st_ref = global_st_refresher;
            } else {
               peer->stimer.st_ref = i;
            }
         } else if (!strcasecmp(v->name, "disallowed_methods")) {
            char *disallow = ast_strdupa(v->value);
            mark_parsed_methods(&peer->disallowed_methods, disallow);
         } else if (!strcasecmp(v->name, "unsolicited_mailbox")) {
            ast_string_field_set(peer, unsolicited_mailbox, v->value);
         } else if (!strcasecmp(v->name, "use_q850_reason")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
         } else if (!strcasecmp(v->name, "encryption")) {
            ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_USE_SRTP);
         } else if (!strcasecmp(v->name, "encryption_taglen")) {
            ast_set2_flag(&peer->flags[2], !strcasecmp(v->value, "32"), SIP_PAGE3_SRTP_TAG_32);
         } else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
            ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
         } else if (!strcasecmp(v->name, "avpf")) {
            ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_USE_AVPF);
         } else if (!strcasecmp(v->name, "icesupport")) {
            ast_set2_flag(&peer->flags[2], ast_true(v->value), SIP_PAGE3_ICE_SUPPORT);
         } else {
            ast_rtp_dtls_cfg_parse(&peer->dtls_cfg, v->name, v->value);
         }
      }

      /* Apply the encryption tag length to the DTLS configuration, in case DTLS is in use */
      peer->dtls_cfg.suite = (ast_test_flag(&peer->flags[2], SIP_PAGE3_SRTP_TAG_32) ? AST_AES_CM_128_HMAC_SHA1_32 : AST_AES_CM_128_HMAC_SHA1_80);

      /* These apply to devstate lookups */
      if (realtime && !strcasecmp(v->name, "lastms")) {
         sscanf(v->value, "%30d", &peer->lastms);
      } else if (realtime && !strcasecmp(v->name, "ipaddr") && !ast_strlen_zero(v->value) ) {
         ast_sockaddr_parse(&peer->addr, v->value, PARSE_PORT_FORBID);
      } else if (realtime && !strcasecmp(v->name, "fullcontact")) {
         if (alt_fullcontact && !alt) {
            /* Reset, because the alternate also has a fullcontact and we
             * do NOT want the field value to be doubled. It might be
             * tempting to skip this, but the first table might not have
             * fullcontact and since we're here, we know that the alternate
             * absolutely does. */
            alt_fullcontact = 0;
            ast_str_reset(fullcontact);
         }
         /* Reconstruct field, because realtime separates our value at the ';' */
         if (ast_str_strlen(fullcontact) > 0) {
            ast_str_append(&fullcontact, 0, ";%s", v->value);
         } else {
            ast_str_set(&fullcontact, 0, "%s", v->value);
         }
      } else if (!strcasecmp(v->name, "qualify")) {
         if (!strcasecmp(v->value, "no")) {
            peer->maxms = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            peer->maxms = default_qualify ? default_qualify : DEFAULT_MAXMS;
         } else if (sscanf(v->value, "%30d", &peer->maxms) != 1) {
            ast_log(LOG_WARNING, "Qualification of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
            peer->maxms = 0;
         }
         if (realtime && !ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->maxms > 0) {
            /* This would otherwise cause a network storm, where the
             * qualify response refreshes the peer from the database,
             * which in turn causes another qualify to be sent, ad
             * infinitum. */
            ast_log(LOG_WARNING, "Qualify is incompatible with dynamic uncached realtime.  Please either turn rtcachefriends on or turn qualify off on peer '%s'\n", peer->name);
            peer->maxms = 0;
         }
      } else if (!strcasecmp(v->name, "keepalive")) {
         if (!strcasecmp(v->value, "no")) {
            peer->keepalive = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            peer->keepalive = DEFAULT_KEEPALIVE_INTERVAL;
         } else if (sscanf(v->value, "%30d", &peer->keepalive) != 1) {
            ast_log(LOG_WARNING, "Keep alive of peer '%s' should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", peer->name, v->lineno);
            peer->keepalive = 0;
         }
      } else if (!strcasecmp(v->name, "callcounter")) {
         peer->call_limit = ast_true(v->value) ? INT_MAX : 0;
      } else if (!strcasecmp(v->name, "call-limit")) {
         peer->call_limit = atoi(v->value);
         if (peer->call_limit < 0) {
            peer->call_limit = 0;
         }
      } else if (!strcasecmp(v->name, "busylevel")) {
         peer->busy_level = atoi(v->value);
         if (peer->busy_level < 0) {
            peer->busy_level = 0;
         }
      } else if (ast_cc_is_config_param(v->name)) {
         ast_cc_set_param(peer->cc_params, v->name, v->value);
      }
   }

   if (!devstate_only) {
      struct sip_mailbox *mailbox;
      AST_LIST_TRAVERSE_SAFE_BEGIN(&peer->mailboxes, mailbox, entry) {
         if (mailbox->delme) {
            AST_LIST_REMOVE_CURRENT(entry);
            destroy_mailbox(mailbox);
         }
      }
      AST_LIST_TRAVERSE_SAFE_END;
   }

   if (!can_parse_xml && (ast_get_cc_agent_policy(peer->cc_params) == AST_CC_AGENT_NATIVE)) {
      ast_log(LOG_WARNING, "Peer %s has a cc_agent_policy of 'native' but required libxml2 dependency is not installed. Changing policy to 'never'\n", peer->name);
      ast_set_cc_agent_policy(peer->cc_params, AST_CC_AGENT_NEVER);
   }

   /* Note that Timer B is dependent upon T1 and MUST NOT be lower
    * than T1 * 64, according to RFC 3261, Section 17.1.1.2 */
   if (peer->timer_b < peer->timer_t1 * 64) {
      if (timerb_set && timert1_set) {
         ast_log(LOG_WARNING, "Timer B has been set lower than recommended for peer %s (%d < 64 * Timer-T1=%d)\n", peer->name, peer->timer_b, peer->timer_t1);
      } else if (timerb_set) {
         if ((peer->timer_t1 = peer->timer_b / 64) < global_t1min) {
            ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", peer->timer_b, peer->timer_t1);
            peer->timer_t1 = global_t1min;
            peer->timer_b = peer->timer_t1 * 64;
         }
         peer->timer_t1 = peer->timer_b / 64;
      } else {
         peer->timer_b = peer->timer_t1 * 64;
      }
   }

   if (!peer->default_outbound_transport) {
      /* Set default set of transports */
      peer->transports = default_transports;
      /* Set default primary transport */
      peer->default_outbound_transport = default_primary_transport;
   }

   /* The default transport type set during build_peer should only replace the socket.type when...
    * 1. Registration is not present and the socket.type and default transport types are different.
    * 2. The socket.type is not an acceptable transport type after rebuilding peer.
    * 3. The socket.type is not set yet. */
   if (((peer->socket.type != peer->default_outbound_transport) && (peer->expire == -1)) ||
      !(peer->socket.type & peer->transports) || !(peer->socket.type)) {
      set_socket_transport(&peer->socket, peer->default_outbound_transport);
   }

   ast_copy_flags(&peer->flags[0], &peerflags[0], mask[0].flags);
   ast_copy_flags(&peer->flags[1], &peerflags[1], mask[1].flags);
   ast_copy_flags(&peer->flags[2], &peerflags[2], mask[2].flags);

   if (ast_str_strlen(fullcontact)) {
      ast_string_field_set(peer, fullcontact, ast_str_buffer(fullcontact));
      peer->rt_fromcontact = TRUE;
      /* We have a hostname in the fullcontact, but if we don't have an
       * address listed on the entry (or if it's 'dynamic'), then we need to
       * parse the entry to obtain the IP address, so a dynamic host can be
       * contacted immediately after reload (as opposed to waiting for it to
       * register once again). But if we have an address for this peer and NAT was
       * specified, use that address instead. */
      /* XXX May need to revisit the final argument; does the realtime DB store whether
       * the original contact was over TLS or not? XXX */
      if ((!ast_test_flag(&peer->flags[2],  SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT))
          || ast_sockaddr_isnull(&peer->addr)) {
         __set_address_from_contact(ast_str_buffer(fullcontact), &peer->addr, 0);
      }
   }

   if (srvlookup && peer->dnsmgr == NULL) {
      char transport[MAXHOSTNAMELEN];
      char _srvlookup[MAXHOSTNAMELEN];
      char *params;

      ast_copy_string(_srvlookup, srvlookup, sizeof(_srvlookup));
      if ((params = strchr(_srvlookup, ';'))) {
         *params++ = '\0';
      }

      snprintf(transport, sizeof(transport), "_%s._%s", get_srv_service(peer->socket.type), get_srv_protocol(peer->socket.type));

      peer->addr.ss.ss_family = get_address_family_filter(peer->socket.type); /* Filter address family */
      if (ast_dnsmgr_lookup_cb(_srvlookup, &peer->addr, &peer->dnsmgr, sip_cfg.srvlookup && !peer->portinuri ? transport : NULL,
               on_dns_update_peer, sip_ref_peer(peer, "Store peer on dnsmgr"))) {
         ast_log(LOG_ERROR, "srvlookup failed for host: %s, on peer %s, removing peer\n", _srvlookup, peer->name);
         sip_unref_peer(peer, "dnsmgr lookup failed, getting rid of peer dnsmgr ref");
         sip_unref_peer(peer, "getting rid of a peer pointer");
         return NULL;
      }
      if (!peer->dnsmgr) {
         /* dnsmgr refresh disabeld, release reference */
         sip_unref_peer(peer, "dnsmgr disabled, unref peer");
      }

      ast_string_field_set(peer, tohost, srvlookup);

      if (global_dynamic_exclude_static && !ast_sockaddr_isnull(&peer->addr)) {
         int ha_error = 0;

         ast_append_acl("deny", ast_sockaddr_stringify_addr(&peer->addr), &sip_cfg.contact_acl, &ha_error, NULL);
         if (ha_error) {
            ast_log(LOG_ERROR, "Bad or unresolved host/IP entry in configuration for peer %s, cannot add to contact ACL\n", peer->name);
         }
      }
   } else if (peer->dnsmgr && !peer->host_dynamic) {
      /* force a refresh here on reload if dnsmgr already exists and host is set. */
      ast_dnsmgr_refresh(peer->dnsmgr);
   }

   if (port && !realtime && peer->host_dynamic) {
      ast_sockaddr_set_port(&peer->defaddr, port);
   } else if (port) {
      ast_sockaddr_set_port(&peer->addr, port);
   }

   if (ast_sockaddr_port(&peer->addr) == 0) {
      ast_sockaddr_set_port(&peer->addr,
                  (peer->socket.type & SIP_TRANSPORT_TLS) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }
   if (ast_sockaddr_port(&peer->defaddr) == 0) {
      ast_sockaddr_set_port(&peer->defaddr,
                  (peer->socket.type & SIP_TRANSPORT_TLS) ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }
   if (!peer->socket.port) {
      peer->socket.port = htons(((peer->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));
   }

   if (realtime) {
      int enablepoke = 1;

      if (!sip_cfg.ignore_regexpire && peer->host_dynamic) {
         time_t nowtime = time(NULL);

         if ((nowtime - regseconds) > 0) {
            destroy_association(peer);
            memset(&peer->addr, 0, sizeof(peer->addr));
            peer->lastms = -1;
            enablepoke = 0;
            ast_debug(1, "Bah, we're expired (%d/%d/%d)!\n", (int)(nowtime - regseconds), (int)regseconds, (int)nowtime);
         }
      }

      /* Startup regular pokes */
      if (!devstate_only && enablepoke) {
         sip_ref_peer(peer, "schedule qualify");
         sip_poke_peer(peer, 0);
      }
   }

   if (ast_test_flag(&peer->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
      sip_cfg.allowsubscribe = TRUE;   /* No global ban any more */
   }
   /* If read-only RT backend, then refresh from local DB cache */
   if (peer->host_dynamic && (!peer->is_realtime || !sip_cfg.peer_rtupdate)) {
      reg_source_db(peer);
   }

   /* If they didn't request that MWI is sent *only* on subscribe, go ahead and
    * subscribe to it now. */
   if (!devstate_only && !ast_test_flag(&peer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY) &&
      !AST_LIST_EMPTY(&peer->mailboxes)) {
      add_peer_mwi_subs(peer);
      /* Send MWI from the event cache only.  This is so we can send initial
       * MWI if app_voicemail got loaded before chan_sip.  If it is the other
       * way, then we will get events when app_voicemail gets loaded. */
      sip_send_mwi_to_peer(peer, 1);
   }

   peer->the_mark = 0;

   oldacl = ast_free_acl_list(oldacl);
   olddirectmediaacl = ast_free_acl_list(olddirectmediaacl);
   if (!ast_strlen_zero(peer->callback)) { /* build string from peer info */
      char *reg_string;
      if (ast_asprintf(&reg_string, "%s?%s:%s@%s/%s", peer->name, peer->username, !ast_strlen_zero(peer->remotesecret) ? peer->remotesecret : peer->secret, peer->tohost, peer->callback) >= 0) {
         sip_register(reg_string, 0); /* XXX TODO: count in registry_count */
         ast_free(reg_string);
      }
   }

   /* If an ACL change subscription is needed and doesn't exist, we need one. */
   if (acl_change_subscription_needed) {
      acl_change_event_subscribe();
   }

   return peer;
}

static int build_reply_digest	(	struct sip_pvt *	p,
		int	method,
		char *	digest,
		int	digest_len
	)		[static]

Build reply digest.

Returns:

Returns -1 if we have no auth

Note:

Build digest challenge for authentication of registrations and calls Also used for authentication of BYE

Definition at line 21917 of file chan_sip.c.

References ao2_lock, ao2_t_ref, ao2_unlock, append_history, ast_copy_string(), ast_debug, ast_md5_hash(), ast_mutex_lock, ast_mutex_unlock, ast_random(), ast_sockaddr_stringify_host_remote(), ast_strlen_zero(), authl, authl_lock, find_realm_authentication(), secret, sip_methods, and text.

Referenced by reply_digest(), transmit_register(), and transmit_request_with_auth().

{
   char a1[256];
   char a2[256];
   char a1_hash[256];
   char a2_hash[256];
   char resp[256];
   char resp_hash[256];
   char uri[256];
   char opaque[256] = "";
   char cnonce[80];
   const char *username;
   const char *secret;
   const char *md5secret;
   struct sip_auth *auth;  /* Realm authentication credential */
   struct sip_auth_container *credentials;

   if (!ast_strlen_zero(p->domain))
      snprintf(uri, sizeof(uri), "%s:%s", p->socket.type == SIP_TRANSPORT_TLS ? "sips" : "sip", p->domain);
   else if (!ast_strlen_zero(p->uri))
      ast_copy_string(uri, p->uri, sizeof(uri));
   else
      snprintf(uri, sizeof(uri), "%s:%s@%s", p->socket.type == SIP_TRANSPORT_TLS ? "sips" : "sip", p->username, ast_sockaddr_stringify_host_remote(&p->sa));

   snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());

   /* Check if we have peer credentials */
   ao2_lock(p);
   credentials = p->peerauth;
   if (credentials) {
      ao2_t_ref(credentials, +1, "Ref peer auth for digest");
   }
   ao2_unlock(p);
   auth = find_realm_authentication(credentials, p->realm);
   if (!auth) {
      /* If not, check global credentials */
      if (credentials) {
         ao2_t_ref(credentials, -1, "Unref peer auth for digest");
      }
      ast_mutex_lock(&authl_lock);
      credentials = authl;
      if (credentials) {
         ao2_t_ref(credentials, +1, "Ref global auth for digest");
      }
      ast_mutex_unlock(&authl_lock);
      auth = find_realm_authentication(credentials, p->realm);
   }

   if (auth) {
      ast_debug(3, "use realm [%s] from peer [%s][%s]\n", auth->username, p->peername, p->username);
      username = auth->username;
      secret = auth->secret;
      md5secret = auth->md5secret;
      if (sipdebug)
         ast_debug(1, "Using realm %s authentication for call %s\n", p->realm, p->callid);
   } else {
      /* No authentication, use peer or register= config */
      username = p->authname;
      secret = p->relatedpeer 
         && !ast_strlen_zero(p->relatedpeer->remotesecret)
            ? p->relatedpeer->remotesecret : p->peersecret;
      md5secret = p->peermd5secret;
   }
   if (ast_strlen_zero(username)) {
      /* We have no authentication */
      if (credentials) {
         ao2_t_ref(credentials, -1, "Unref auth for digest");
      }
      return -1;
   }

   /* Calculate SIP digest response */
   snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
   snprintf(a2, sizeof(a2), "%s:%s", sip_methods[method].text, uri);
   if (!ast_strlen_zero(md5secret))
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   else
      ast_md5_hash(a1_hash, a1);
   ast_md5_hash(a2_hash, a2);

   p->noncecount++;
   if (!ast_strlen_zero(p->qop))
      snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash);
   else
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
   ast_md5_hash(resp_hash, resp);

   /* only include the opaque string if it's set */
   if (!ast_strlen_zero(p->opaque)) {
      snprintf(opaque, sizeof(opaque), ", opaque=\"%s\"", p->opaque);
   }

   /* XXX We hard code our qop to "auth" for now.  XXX */
   if (!ast_strlen_zero(p->qop))
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount);
   else
      snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);

   append_history(p, "AuthResp", "Auth response sent for %s in realm %s - nc %d", username, p->realm, p->noncecount);

   if (credentials) {
      ao2_t_ref(credentials, -1, "Unref auth for digest");
   }
   return 0;
}

static void build_route	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	backwards,
		int	resp
	)		[static]

Build route list from Record-Route header.

Parameters:

resp	the SIP response code or 0 for a request

Definition at line 16236 of file chan_sip.c.

References __get_header(), ast_copy_string(), ast_debug, ast_malloc, ast_strlen_zero(), free_old_route(), get_in_brackets(), get_in_brackets_const(), len(), list_route(), sip_debug_test_pvt(), and sip_get_header().

Referenced by forked_invite_init(), handle_request_invite(), handle_request_subscribe(), and handle_response_invite().

{
   struct sip_route *thishop, *head, *tail;
   int start = 0;
   int len;
   const char *rr, *c;

   /* Once a persistent route is set, don't fool with it */
   if (p->route && p->route_persistent) {
      ast_debug(1, "build_route: Retaining previous route: <%s>\n", p->route->hop);
      return;
   }

   if (p->route) {
      free_old_route(p->route);
      p->route = NULL;
   }

   /* We only want to create the route set the first time this is called except
      it is called from a provisional response.*/
   if ((resp < 100) || (resp > 199)) {
      p->route_persistent = 1;
   }

   /* Build a tailq, then assign it to p->route when done.
    * If backwards, we add entries from the head so they end up
    * in reverse order. However, we do need to maintain a correct
    * tail pointer because the contact is always at the end.
    */
   head = NULL;
   tail = head;
   /* 1st we pass through all the hops in any Record-Route headers */
   for (;;) {
      /* Each Record-Route header */
      int len = 0;
      const char *uri;
      rr = __get_header(req, "Record-Route", &start);
      if (*rr == '\0') {
         break;
      }
      while (!get_in_brackets_const(rr, &uri, &len)) {
         len++;
         rr = strchr(rr, ',');
         if(rr >= uri && rr < (uri + len)) {
            /* comma inside brackets*/
            const char *next_br = strchr(rr, '<');
            if (next_br && next_br < (uri + len)) {
               rr++;
               continue;
            }
            continue;
         }
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            ast_copy_string(thishop->hop, uri, len);
            ast_debug(2, "build_route: Record-Route hop: <%s>\n", thishop->hop);
            /* Link in */
            if (backwards) {
               /* Link in at head so they end up in reverse order */
               thishop->next = head;
               head = thishop;
               /* If this was the first then it'll be the tail */
               if (!tail) {
                  tail = thishop;
               }
            } else {
               thishop->next = NULL;
               /* Link in at the end */
               if (tail) {
                  tail->next = thishop;
               } else {
                  head = thishop;
               }
               tail = thishop;
            }
         }
         rr = strchr(uri + len, ',');
         if (rr == NULL) {
            /* No more field-values, we're done with this header */
            break;
         }
         /* Advance past comma */
         rr++;
      }
   }

   /* Only append the contact if we are dealing with a strict router */
   if (!head || (!ast_strlen_zero(head->hop) && strstr(head->hop, ";lr") == NULL) ) {
      /* 2nd append the Contact: if there is one */
      /* Can be multiple Contact headers, comma separated values - we just take the first */
      char *contact = ast_strdupa(sip_get_header(req, "Contact"));
      if (!ast_strlen_zero(contact)) {
         ast_debug(2, "build_route: Contact hop: %s\n", contact);
         /* Look for <: delimited address */
         c = get_in_brackets(contact);
         len = strlen(c) + 1;
         if ((thishop = ast_malloc(sizeof(*thishop) + len))) {
            /* ast_calloc is not needed because all fields are initialized in this block */
            ast_copy_string(thishop->hop, c, len);
            thishop->next = NULL;
            /* Goes at the end */
            if (tail) {
               tail->next = thishop;
            } else {
               head = thishop;
            }
         }
      }
   }

   /* Store as new route */
   p->route = head;

   /* For debugging dump what we ended up with */
   if (sip_debug_test_pvt(p)) {
      list_route(p->route);
   }
}

static void build_via

(

struct sip_pvt *

p

)

[static]

Build a Via header for a request.

Definition at line 3914 of file chan_sip.c.

References ast_sockaddr_stringify_remote(), ast_test_flag, and get_transport_pvt().

Referenced by __sip_subscribe_mwi_do(), reqprep(), sip_alloc(), sip_cli_notify(), sip_msg_send(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_invite(), transmit_register(), and transmit_response_using_temp().

{
   /* Work around buggy UNIDEN UIP200 firmware */
   const char *rport = (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)) ? ";rport" : "";

   /* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
   snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s;branch=z9hG4bK%08x%s",
       get_transport_pvt(p),
       ast_sockaddr_stringify_remote(&p->ourip),
       (int) p->branch, rport);
}

static int cb_extensionstate	(	char *	context,
		char *	exten,
		struct ast_state_cb_info *	info,
		void *	data
	)		[static]

Callback for the devicestate notification (SUBSCRIBE) support subsystem.

Note:

If you add an "hint" priority to the extension in the dial plan, you will get notifications on device state changes

Definition at line 16734 of file chan_sip.c.

References allow_notify_user_presence(), AST_HINT_UPDATE_PRESENCE, ast_state_cb_info::device_state_info, ast_state_cb_info::exten_state, extensionstate_update(), FALSE, ast_state_cb_info::presence_message, ast_state_cb_info::presence_state, ast_state_cb_info::presence_subtype, ast_state_cb_info::reason, and state_notify_data::state.

Referenced by dialog_unlink_all(), and handle_request_subscribe().

{
   struct sip_pvt *p = data;
   struct state_notify_data notify_data = {
      .state = info->exten_state,
      .device_state_info = info->device_state_info,
      .presence_state = info->presence_state,
      .presence_subtype = info->presence_subtype,
      .presence_message = info->presence_message,
   };

   if ((info->reason == AST_HINT_UPDATE_PRESENCE) && !(allow_notify_user_presence(p))) {
      /* ignore a presence triggered update if we know the useragent doesn't care */
      return 0;
   }

   return extensionstate_update(context, exten, &notify_data, p, FALSE);
}

static void cb_extensionstate_destroy	(	int	id,
		void *	data
	)		[static]

Definition at line 16642 of file chan_sip.c.

Referenced by handle_request_subscribe().

{
   struct sip_pvt *p = data;

   dialog_unref(p, "the extensionstate containing this dialog ptr was destroyed");
}

static void cc_epa_destructor

(

void *

data

)

[static]

Definition at line 950 of file chan_sip.c.

References ast_free.

{
   struct sip_epa_entry *epa_entry = data;
   struct cc_epa_entry *cc_entry = epa_entry->instance_data;
   ast_free(cc_entry);
}

static int cc_esc_publish_handler	(	struct sip_pvt *	pvt,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		struct sip_esc_entry *	esc_entry
	)		[static]

Definition at line 27316 of file chan_sip.c.

References ao2_ref, ast_cc_agent_caller_available(), ast_cc_agent_caller_busy(), ast_log(), ast_strlen_zero(), ast_xml_close(), ast_xml_find_element(), ast_xml_free_text(), ast_xml_get_root(), ast_xml_get_text(), ast_xml_node_get_children(), ast_cc_agent::core_id, ast_cc_agent::device_name, FALSE, find_sip_cc_agent_by_notify_uri(), find_sip_cc_agent_by_subscribe_uri(), LOG_NOTICE, LOG_WARNING, ast_cc_agent::private_data, sip_pidf_validate(), transmit_response(), and TRUE.

{
   const char *uri = REQ_OFFSET_TO_STR(req, rlpart2);
   struct ast_cc_agent *agent;
   struct sip_cc_agent_pvt *agent_pvt;
   struct ast_xml_doc *pidf_doc = NULL;
   const char *basic_status = NULL;
   struct ast_xml_node *presence_node;
   struct ast_xml_node *presence_children;
   struct ast_xml_node *tuple_node;
   struct ast_xml_node *tuple_children;
   struct ast_xml_node *status_node;
   struct ast_xml_node *status_children;
   struct ast_xml_node *basic_node;
   int res = 0;

   if (!((agent = find_sip_cc_agent_by_notify_uri(uri)) || (agent = find_sip_cc_agent_by_subscribe_uri(uri)))) {
      ast_log(LOG_WARNING, "Could not find agent using uri '%s'\n", uri);
      transmit_response(pvt, "412 Conditional Request Failed", req);
      return -1;
   }

   agent_pvt = agent->private_data;

   if (sip_pidf_validate(req, &pidf_doc) == FALSE) {
      res = -1;
      goto cc_publish_cleanup;
   }

   /* It's important to note that the PIDF validation routine has no knowledge
    * of what we specifically want in this instance. A valid PIDF document could
    * have no tuples, or it could have tuples whose status element has no basic
    * element contained within. While not violating the PIDF spec, these are
    * insufficient for our needs in this situation
    */
   presence_node = ast_xml_get_root(pidf_doc);
   if (!(presence_children = ast_xml_node_get_children(presence_node))) {
      ast_log(LOG_WARNING, "No tuples within presence element.\n");
      res = -1;
      goto cc_publish_cleanup;
   }

   if (!(tuple_node = ast_xml_find_element(presence_children, "tuple", NULL, NULL))) {
      ast_log(LOG_NOTICE, "Couldn't find tuple node?\n");
      res = -1;
      goto cc_publish_cleanup;
   }

   /* We already made sure that the tuple has a status node when we validated the PIDF
    * document earlier. So there's no need to enclose this operation in an if statement.
    */
   tuple_children = ast_xml_node_get_children(tuple_node);
   /* coverity[null_returns: FALSE] */
   status_node = ast_xml_find_element(tuple_children, "status", NULL, NULL);

   if (!(status_children = ast_xml_node_get_children(status_node))) {
      ast_log(LOG_WARNING, "No basic elements within status element.\n");
      res = -1;
      goto cc_publish_cleanup;
   }

   if (!(basic_node = ast_xml_find_element(status_children, "basic", NULL, NULL))) {
      ast_log(LOG_WARNING, "Couldn't find basic node?\n");
      res = -1;
      goto cc_publish_cleanup;
   }

   basic_status = ast_xml_get_text(basic_node);

   if (ast_strlen_zero(basic_status)) {
      ast_log(LOG_NOTICE, "NOthing in basic node?\n");
      res = -1;
      goto cc_publish_cleanup;
   }

   if (!strcmp(basic_status, "open")) {
      agent_pvt->is_available = TRUE;
      ast_cc_agent_caller_available(agent->core_id, "Received PUBLISH stating SIP caller %s is available",
            agent->device_name);
   } else if (!strcmp(basic_status, "closed")) {
      agent_pvt->is_available = FALSE;
      ast_cc_agent_caller_busy(agent->core_id, "Received PUBLISH stating SIP caller %s is busy",
            agent->device_name);
   } else {
      ast_log(LOG_NOTICE, "Invalid content in basic element: %s\n", basic_status);
   }

cc_publish_cleanup:
   if (basic_status) {
      ast_xml_free_text(basic_status);
   }
   if (pidf_doc) {
      ast_xml_close(pidf_doc);
   }
   ao2_ref(agent, -1);
   if (res) {
      transmit_response(pvt, "400 Bad Request", req);
   }
   return res;
}

static void cc_handle_publish_error	(	struct sip_pvt *	pvt,
		const int	resp,
		struct sip_request *	req,
		struct sip_epa_entry *	epa_entry
	)		[static]

Definition at line 22565 of file chan_sip.c.

References ao2_callback, ao2_ref, ast_cc_monitor_failed(), ast_log(), ast_strlen_zero(), FALSE, find_sip_monitor_instance_by_suspension_entry(), LOG_WARNING, sip_get_header(), and transmit_invite().

{
   struct cc_epa_entry *cc_entry = epa_entry->instance_data;
   struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
         find_sip_monitor_instance_by_suspension_entry, epa_entry);
   const char *min_expires;

   if (!monitor_instance) {
      ast_log(LOG_WARNING, "Can't find monitor_instance corresponding to epa_entry %p.\n", epa_entry);
      return;
   }

   if (resp != 423) {
      ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
            "Received error response to our PUBLISH");
      ao2_ref(monitor_instance, -1);
      return;
   }

   /* Allrighty, the other end doesn't like our Expires value. They think it's
    * too small, so let's see if they've provided a more sensible value. If they
    * haven't, then we'll just double our Expires value and see if they like that
    * instead.
    *
    * XXX Ideally this logic could be placed into its own function so that SUBSCRIBE,
    * PUBLISH, and REGISTER could all benefit from the same shared code.
    */
   min_expires = sip_get_header(req, "Min-Expires");
   if (ast_strlen_zero(min_expires)) {
      pvt->expiry *= 2;
      if (pvt->expiry < 0) {
         /* You dork! You overflowed! */
         ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
               "PUBLISH expiry overflowed");
         ao2_ref(monitor_instance, -1);
         return;
      }
   } else if (sscanf(min_expires, "%30d", &pvt->expiry) != 1) {
      ast_cc_monitor_failed(cc_entry->core_id, monitor_instance->device_name,
            "Min-Expires has non-numeric value");
      ao2_ref(monitor_instance, -1);
      return;
   }
   /* At this point, we have most certainly changed pvt->expiry, so try transmitting
    * again
    */
   transmit_invite(pvt, SIP_PUBLISH, FALSE, 0, NULL);
   ao2_ref(monitor_instance, -1);
}

static void change_callid_pvt	(	struct sip_pvt *	pvt,
		const char *	callid
	)		[static]

Definition at line 8545 of file chan_sip.c.

References ao2_lock, ao2_t_link, ao2_unlock, ast_debug, ast_string_field_set, build_callid_pvt(), and CONTAINER_UNLINK.

Referenced by __sip_subscribe_mwi_do(), create_addr_from_peer(), sip_cli_notify(), sip_poke_peer(), sip_request_call(), and sip_send_mwi_to_peer().

{
   int in_dialog_container;
   int in_rtp_container;
   char *oldid = ast_strdupa(pvt->callid);

   ao2_lock(dialogs);
   ao2_lock(dialogs_rtpcheck);
   in_dialog_container = CONTAINER_UNLINK(dialogs, pvt,
      "About to change the callid -- remove the old name");
   in_rtp_container = CONTAINER_UNLINK(dialogs_rtpcheck, pvt,
      "About to change the callid -- remove the old name");
   if (callid) {
      ast_string_field_set(pvt, callid, callid);
   } else {
      build_callid_pvt(pvt);
   }
   if (in_dialog_container) {
      ao2_t_link(dialogs, pvt, "New dialog callid -- inserted back into table");
   }
   if (in_rtp_container) {
      ao2_t_link(dialogs_rtpcheck, pvt, "New dialog callid -- inserted back into table");
   }
   ao2_unlock(dialogs_rtpcheck);
   ao2_unlock(dialogs);

   if (strcmp(oldid, pvt->callid)) {
      ast_debug(1, "SIP call-id changed from '%s' to '%s'\n", oldid, pvt->callid);
   }
}

static void change_hold_state	(	struct sip_pvt *	dialog,
		struct sip_request *	req,
		int	holdstate,
		int	sendonly
	)		[static]

Change hold state for a call.

Definition at line 9791 of file chan_sip.c.

References append_history, ast_channel_name(), ast_channel_uniqueid(), ast_clear_flag, ast_set_flag, ast_str_buffer(), ast_test_flag, EVENT_FLAG_CALL, manager_event, sip_cfg, and sip_peer_hold().

Referenced by handle_request_invite(), and process_sdp().

{
   if (sip_cfg.notifyhold && (!holdstate || !ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD)))
      sip_peer_hold(dialog, holdstate);
   if (sip_cfg.callevents)
      manager_event(EVENT_FLAG_CALL, "Hold",
               "Status: %s\r\n"
               "Channel: %s\r\n"
               "Uniqueid: %s\r\n",
               holdstate ? "On" : "Off",
               ast_channel_name(dialog->owner),
               ast_channel_uniqueid(dialog->owner));
   append_history(dialog, holdstate ? "Hold" : "Unhold", "%s", ast_str_buffer(req->data));
   if (!holdstate) { /* Put off remote hold */
      ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD);   /* Clear both flags */
      return;
   }
   /* No address for RTP, we're on hold */

   /* Ensure hold flags are cleared so that overlapping flags do not conflict */
   ast_clear_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD);

   if (sendonly == 1)   /* One directional hold (sendonly/recvonly) */
      ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ONEDIR);
   else if (sendonly == 2) /* Inactive stream */
      ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_INACTIVE);
   else
      ast_set_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD_ACTIVE);
   return;
}

static void change_redirecting_information	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_party_redirecting *	redirecting,
		struct ast_set_party_redirecting *	update_redirecting,
		int	set_call_forward
	)		[static]

update redirecting information for a channel based on headers

Definition at line 22291 of file chan_sip.c.

References ast_debug, ast_free, AST_REDIRECTING_REASON_UNCONDITIONAL, ast_strdup, ast_strlen_zero(), ast_party_redirecting::from, ast_set_party_redirecting::from, get_name_and_number(), get_rdnis(), ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_moved_contact(), ast_party_redirecting::reason, sip_get_header(), ast_party_name::str, ast_party_number::str, ast_party_id::tag, ast_party_redirecting::to, ast_set_party_redirecting::to, ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_request_invite(), handle_request_refer(), handle_response(), and handle_response_invite().

{
   char *redirecting_from_name = NULL;
   char *redirecting_from_number = NULL;
   char *redirecting_to_name = NULL;
   char *redirecting_to_number = NULL;
   int reason = AST_REDIRECTING_REASON_UNCONDITIONAL;
   int is_response = req->method == SIP_RESPONSE;
   int res = 0;

   res = get_rdnis(p, req, &redirecting_from_name, &redirecting_from_number, &reason);
   if (res == -1) {
      if (is_response) {
         get_name_and_number(sip_get_header(req, "TO"), &redirecting_from_name, &redirecting_from_number);
      } else {
         return;
      }
   }

   /* At this point, all redirecting "from" info should be filled in appropriately
    * on to the "to" info
    */

   if (is_response) {
      parse_moved_contact(p, req, &redirecting_to_name, &redirecting_to_number, set_call_forward);
   } else {
      get_name_and_number(sip_get_header(req, "TO"), &redirecting_to_name, &redirecting_to_number);
   }

   if (!ast_strlen_zero(redirecting_from_number)) {
      ast_debug(3, "Got redirecting from number %s\n", redirecting_from_number);
      update_redirecting->from.number = 1;
      redirecting->from.number.valid = 1;
      ast_free(redirecting->from.number.str);
      redirecting->from.number.str = redirecting_from_number;
   }
   if (!ast_strlen_zero(redirecting_from_name)) {
      ast_debug(3, "Got redirecting from name %s\n", redirecting_from_name);
      update_redirecting->from.name = 1;
      redirecting->from.name.valid = 1;
      ast_free(redirecting->from.name.str);
      redirecting->from.name.str = redirecting_from_name;
   }
   if (!ast_strlen_zero(p->cid_tag)) {
      ast_free(redirecting->from.tag);
      redirecting->from.tag = ast_strdup(p->cid_tag);
      ast_free(redirecting->to.tag);
      redirecting->to.tag = ast_strdup(p->cid_tag);
   }
   if (!ast_strlen_zero(redirecting_to_number)) {
      ast_debug(3, "Got redirecting to number %s\n", redirecting_to_number);
      update_redirecting->to.number = 1;
      redirecting->to.number.valid = 1;
      ast_free(redirecting->to.number.str);
      redirecting->to.number.str = redirecting_to_number;
   }
   if (!ast_strlen_zero(redirecting_to_name)) {
      ast_debug(3, "Got redirecting to name %s\n", redirecting_from_number);
      update_redirecting->to.name = 1;
      redirecting->to.name.valid = 1;
      ast_free(redirecting->to.name.str);
      redirecting->to.name.str = redirecting_to_name;
   }
   redirecting->reason = reason;
}

static void change_t38_state	(	struct sip_pvt *	p,
		int	state
	)		[static]

Change the T38 state on a SIP dialog.

Definition at line 5756 of file chan_sip.c.

References ast_channel_name(), AST_CONTROL_T38_PARAMETERS, ast_debug, ast_queue_control_data(), AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_TERMINATED, ast_udptl_get_far_max_ifp(), ast_udptl_set_tag(), ast_control_t38_parameters::max_ifp, ast_control_t38_parameters::request_response, and state.

Referenced by handle_response_invite(), interpret_t38_parameters(), process_sdp(), and sip_t38_abort().

{
   int old = p->t38.state;
   struct ast_channel *chan = p->owner;
   struct ast_control_t38_parameters parameters = { .request_response = 0 };

   /* Don't bother changing if we are already in the state wanted */
   if (old == state)
      return;

   p->t38.state = state;
   ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? ast_channel_name(chan) : "<none>");

   /* If no channel was provided we can't send off a control frame */
   if (!chan)
      return;

   /* Given the state requested and old state determine what control frame we want to queue up */
   switch (state) {
   case T38_PEER_REINVITE:
      parameters = p->t38.their_parms;
      parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
      parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
      ast_udptl_set_tag(p->udptl, "%s", ast_channel_name(chan));
      break;
   case T38_ENABLED:
      parameters = p->t38.their_parms;
      parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
      parameters.request_response = AST_T38_NEGOTIATED;
      ast_udptl_set_tag(p->udptl, "%s", ast_channel_name(chan));
      break;
   case T38_REJECTED:
   case T38_DISABLED:
      if (old == T38_ENABLED) {
         parameters.request_response = AST_T38_TERMINATED;
      } else if (old == T38_LOCAL_REINVITE) {
         parameters.request_response = AST_T38_REFUSED;
      }
      break;
   case T38_LOCAL_REINVITE:
      /* wait until we get a peer response before responding to local reinvite */
      break;
   }

   /* Woot we got a message, create a control frame and send it on! */
   if (parameters.request_response)
      ast_queue_control_data(chan, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
}

static enum check_auth_result check_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const char *	username,
		const char *	secret,
		const char *	md5secret,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable
	)		[static]

Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set)

Returns:

0 on success, non-zero on error

XXX

Todo:

need a better return code here

XXX

Todo:

need a better return code here

Definition at line 16401 of file chan_sip.c.

References append_history, ast_copy_string(), AST_DYNSTR_BUILD_FAILED, ast_log(), ast_md5_hash(), ast_str_set(), ast_str_thread_get(), ast_strlen_zero(), BOGUS_PEER_MD5SECRET, build_nonce(), FALSE, LOG_NOTICE, LOG_WARNING, S_OR, sip_auth_headers(), sip_digest_parser(), sip_get_header(), sip_methods, sip_scheddestroy(), text, transmit_response_with_auth(), and TRUE.

Referenced by check_peer_ok(), and register_verify().

{
   const char *response;
   char *reqheader, *respheader;
   const char *authtoken;
   char a1_hash[256];
   char resp_hash[256]="";
   char *c;
   int is_bogus_peer = 0;
   int  wrongnonce = FALSE;
   int  good_response;
   const char *usednonce = p->nonce;
   struct ast_str *buf;
   int res;

   /* table of recognised keywords, and their value in the digest */
   struct digestkeys keys[] = {
      [K_RESP] = { "response=", "" },
      [K_URI] = { "uri=", "" },
      [K_USER] = { "username=", "" },
      [K_NONCE] = { "nonce=", "" },
      [K_LAST] = { NULL, NULL}
   };

   /* Always OK if no secret */
   if (ast_strlen_zero(secret) && ast_strlen_zero(md5secret)) {
      return AUTH_SUCCESSFUL;
   }

   /* Always auth with WWW-auth since we're NOT a proxy */
   /* Using proxy-auth in a B2BUA may block proxy authorization in the same transaction */
   response = "401 Unauthorized";

   /*
    * Note the apparent swap of arguments below, compared to other
    * usages of sip_auth_headers().
    */
   sip_auth_headers(WWW_AUTH, &respheader, &reqheader);

   authtoken = sip_get_header(req, reqheader);
   if (req->ignore && !ast_strlen_zero(p->nonce) && ast_strlen_zero(authtoken)) {
      /* This is a retransmitted invite/register/etc, don't reconstruct authentication
         information */
      if (!reliable) {
         /* Resend message if this was NOT a reliable delivery.   Otherwise the
            retransmission should get it */
         transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
         /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
      return AUTH_CHALLENGE_SENT;
   } else if (ast_strlen_zero(p->nonce) || ast_strlen_zero(authtoken)) {
      /* We have no auth, so issue challenge and request authentication */
      build_nonce(p, 1); /* Create nonce for challenge */
      transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   }

   /* --- We have auth, so check it */

   /* Whoever came up with the authentication section of SIP can suck my %&#$&* for not putting
      an example in the spec of just what it is you're doing a hash on. */

   if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
   }

   /* Make a copy of the response and parse it */
   res = ast_str_set(&buf, 0, "%s", authtoken);

   if (res == AST_DYNSTR_BUILD_FAILED) {
      return AUTH_SECRET_FAILED; /*! XXX \todo need a better return code here */
   }

   c = buf->str;

   sip_digest_parser(c, keys);

   /* We cannot rely on the bogus_peer having a bad md5 value. Someone could
    * use it to construct valid auth. */
   if (md5secret && strcmp(md5secret, BOGUS_PEER_MD5SECRET) == 0) {
      is_bogus_peer = 1;
   }

   /* Verify that digest username matches  the username we auth as */
   if (strcmp(username, keys[K_USER].s) && !is_bogus_peer) {
      ast_log(LOG_WARNING, "username mismatch, have <%s>, digest has <%s>\n",
         username, keys[K_USER].s);
      /* Oops, we're trying something here */
      return AUTH_USERNAME_MISMATCH;
   }

   /* Verify nonce from request matches our nonce, and the nonce has not already been responded to.
    * If this check fails, send 401 with new nonce */
   if (strcasecmp(p->nonce, keys[K_NONCE].s) || p->stalenonce) { /* XXX it was 'n'casecmp ? */
      wrongnonce = TRUE;
      usednonce = keys[K_NONCE].s;
   } else {
      p->stalenonce = 1; /* now, since the nonce has a response, mark it as stale so it can't be sent or responded to again */
   }

   if (!ast_strlen_zero(md5secret)) {
      ast_copy_string(a1_hash, md5secret, sizeof(a1_hash));
   } else {
      char a1[256];

      snprintf(a1, sizeof(a1), "%s:%s:%s", username, p->realm, secret);
      ast_md5_hash(a1_hash, a1);
   }

   /* compute the expected response to compare with what we received */
   {
      char a2[256];
      char a2_hash[256];
      char resp[256];

      snprintf(a2, sizeof(a2), "%s:%s", sip_methods[sipmethod].text,
            S_OR(keys[K_URI].s, uri));
      ast_md5_hash(a2_hash, a2);
      snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, usednonce, a2_hash);
      ast_md5_hash(resp_hash, resp);
   }

   good_response = keys[K_RESP].s &&
         !strncasecmp(keys[K_RESP].s, resp_hash, strlen(resp_hash)) &&
         !is_bogus_peer; /* lastly, check that the peer isn't the fake peer */
   if (wrongnonce) {
      if (good_response) {
         if (sipdebug)
            ast_log(LOG_NOTICE, "Correct auth, but based on stale nonce received from '%s'\n", sip_get_header(req, "From"));
         /* We got working auth token, based on stale nonce . */
         build_nonce(p, 0);
         transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, TRUE);
      } else {
         /* Everything was wrong, so give the device one more try with a new challenge */
         if (!req->ignore) {
            if (sipdebug) {
               ast_log(LOG_NOTICE, "Bad authentication received from '%s'\n", sip_get_header(req, "To"));
            }
            build_nonce(p, 1);
         } else {
            if (sipdebug) {
               ast_log(LOG_NOTICE, "Duplicate authentication received from '%s'\n", sip_get_header(req, "To"));
            }
         }
         transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, FALSE);
      }

      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return AUTH_CHALLENGE_SENT;
   }
   if (good_response) {
      append_history(p, "AuthOK", "Auth challenge successful for %s", username);
      return AUTH_SUCCESSFUL;
   }

   /* Ok, we have a bad username/secret pair */
   /* Tell the UAS not to re-send this authentication data, because
      it will continue to fail
   */

   return AUTH_SECRET_FAILED;
}

static void check_for_nat	(	const struct ast_sockaddr *	addr,
		struct sip_pvt *	p
	)		[static]

Check and see if the requesting UA is likely to be behind a NAT.

If the requesting NAT is behind NAT, set the * natdetected flag so that later, peers with nat=auto_* can use the value. Also, set the flags so that Asterisk responds identically whether or not a peer exists so as not to leak peer name information.

Definition at line 18057 of file chan_sip.c.

References ast_clear_flag, ast_debug, ast_set_flag, ast_sockaddr_cmp_addr(), ast_sockaddr_stringify_addr(), and ast_test_flag.

Referenced by check_via(), and sip_request_call().

{

   if (!addr || !p) {
      return;
   }

   if (ast_sockaddr_cmp_addr(addr, &p->recv)) {
      char *tmp_str = ast_strdupa(ast_sockaddr_stringify_addr(addr));
      ast_debug(3, "NAT detected for %s / %s\n", tmp_str, ast_sockaddr_stringify_addr(&p->recv));
      p->natdetected = 1;
      if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
         ast_set_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
      }
      if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
         ast_set_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
      }
   } else {
      p->natdetected = 0;
      if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
         ast_clear_flag(&p->flags[0], SIP_NAT_FORCE_RPORT);
      }
      if (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
         ast_clear_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
      }
   }

}

static enum message_integrity check_message_integrity	(	struct ast_str **	request,
		struct ast_str **	overflow
	)		[static]

Check that a message received over TCP is a full message.

This will take the information read in and then determine if 1) The message is a full SIP request 2) The message is a partial SIP request 3) The message contains a full SIP request along with another partial request

Parameters:

data	The unparsed incoming SIP message.
request	The resulting request with extra fragments removed.
overflow	If the message contains more than a full request, this is the remainder of the message

Returns:

The resulting integrity of the message

Definition at line 2901 of file chan_sip.c.

References ast_str_append(), ast_str_buffer(), ast_str_strlen(), ast_str_truncate(), MESSAGE_COMPLETE, MESSAGE_FRAGMENT, MESSAGE_FRAGMENT_COMPLETE, MESSAGE_INVALID, and read_raw_content_length().

Referenced by sip_tcp_read().

{
   char *message = ast_str_buffer(*request);
   char *body;
   int content_length;
   int message_len = ast_str_strlen(*request);
   int body_len;

   /* Important pieces to search for in a SIP request are \r\n\r\n. This
    * marks either
    * 1) The division between the headers and body
    * 2) The end of the SIP request
    */
   body = strstr(message, "\r\n\r\n");
   if (!body) {
      /* This is clearly a partial message since we haven't reached an end
       * yet.
       */
      return MESSAGE_FRAGMENT;
   }
   body += sizeof("\r\n\r\n") - 1;
   body_len = message_len - (body - message);

   body[-1] = '\0';
   content_length = read_raw_content_length(message);
   body[-1] = '\n';

   if (content_length < 0) {
      return MESSAGE_INVALID;
   } else if (content_length == 0) {
      /* We've definitely received an entire message. We need
       * to check if there's also a fragment of another message
       * in addition.
       */
      if (body_len == 0) {
         return MESSAGE_COMPLETE;
      } else {
         ast_str_append(overflow, 0, "%s", body);
         ast_str_truncate(*request, message_len - body_len);
         return MESSAGE_FRAGMENT_COMPLETE;
      }
   }
   /* Positive content length. Let's see what sort of
    * message body we're dealing with.
    */
   if (body_len < content_length) {
      /* We don't have the full message body yet */
      return MESSAGE_FRAGMENT;
   } else if (body_len > content_length) {
      /* We have the full message plus a fragment of a further
       * message
       */
      ast_str_append(overflow, 0, "%s", body + content_length);
      ast_str_truncate(*request, message_len - (body_len - content_length));
      return MESSAGE_FRAGMENT_COMPLETE;
   } else {
      /* Yay! Full message with no extra content */
      return MESSAGE_COMPLETE;
   }
}

static enum check_auth_result check_peer_ok	(	struct sip_pvt *	p,
		char *	of,
		struct sip_request *	req,
		int	sipmethod,
		struct ast_sockaddr *	addr,
		struct sip_peer **	authpeer,
		enum xmittype	reliable,
		char *	calleridname,
		char *	uri2
	)		[static]

Validate device authentication.

Definition at line 18160 of file chan_sip.c.

References accountcode, ao2_t_ref, ast_apply_acl(), ast_cc_copy_config_params(), ast_copy_flags, ast_copy_string(), ast_debug, ast_format_cap_copy(), ast_format_cap_destroy(), ast_format_cap_is_empty(), ast_format_cap_joint(), ast_is_shrinkable_phonenumber(), ast_ref_namedgroups(), ast_rtp_codecs_packetization_set(), ast_rtp_dtls_cfg_copy(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_set_flag, ast_shrink_phone_number(), ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_unref_namedgroups(), ast_variables_destroy(), ast_verbose(), bogus_peer, check_auth(), cid_name, cid_num, context, copy_vars(), debug, dialog_initialize_rtp(), do_setnat(), dummy(), FALSE, get_rpid(), language, mohinterpret, mohsuggest, parkinglot, parse_uri(), set_peer_nat(), set_pvt_allowed_methods(), set_t38_capabilities(), sip_cfg, sip_debug_test_addr(), sip_find_peer(), sip_find_peer_by_ip_and_exten(), sip_ref_peer(), sip_unref_peer(), and TRUE.

Referenced by check_user_full().

{
   enum check_auth_result res;
   int debug = sip_debug_test_addr(addr);
   struct sip_peer *peer;

   if (sipmethod == SIP_SUBSCRIBE) {
      /* For subscribes, match on device name only; for other methods,
      * match on IP address-port of the incoming request.
      */
      peer = sip_find_peer(of, NULL, TRUE, FINDALLDEVICES, FALSE, 0);
   } else {
      /* First find devices based on username (avoid all type=peer's) */
      peer = sip_find_peer(of, NULL, TRUE, FINDUSERS, FALSE, 0);

      /* Then find devices based on IP */
      if (!peer) {
         char *uri_tmp, *callback = NULL, *dummy;
         uri_tmp = ast_strdupa(uri2);
         parse_uri(uri_tmp, "sip:,sips:", &callback, &dummy, &dummy, &dummy);
         if (!ast_strlen_zero(callback) && (peer = sip_find_peer_by_ip_and_exten(&p->recv, callback, p->socket.type))) {
            ; /* found, fall through */
         } else {
            peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);
         }
      }
   }

   if (!peer) {
      if (debug) {
         ast_verbose("No matching peer for '%s' from '%s'\n",
            of, ast_sockaddr_stringify(&p->recv));
      }

      /* If you don't mind, we can return 404s for devices that do
       * not exist: username disclosure. If we allow guests, there
       * is no way around that. */
      if (sip_cfg.allowguest || !sip_cfg.alwaysauthreject) {
         return AUTH_DONT_KNOW;
      }

      /* If you do mind, we use a peer that will never authenticate.
       * This ensures that we follow the same code path as regular
       * auth: less chance for username disclosure. */
      peer = bogus_peer;
      sip_ref_peer(peer, "sip_ref_peer: check_peer_ok: must ref bogus_peer so unreffing it does not fail");
   }

   /*  build_peer, called through sip_find_peer, is not able to check the
    *  sip_pvt->natdetected flag in order to determine if the peer is behind
    *  NAT or not when SIP_PAGE3_NAT_AUTO_RPORT or SIP_PAGE3_NAT_AUTO_COMEDIA
    *  are set on the peer.  So we check for that here and set the peer's
    *  address accordingly.
    */
   set_peer_nat(p, peer);

   if (p->natdetected && ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
      ast_sockaddr_copy(&peer->addr, &p->recv);
   }

   if (!ast_apply_acl(peer->acl, addr, "SIP Peer ACL: ")) {
      ast_debug(2, "Found peer '%s' for '%s', but fails host access\n", peer->name, of);
      sip_unref_peer(peer, "sip_unref_peer: check_peer_ok: from sip_find_peer call, early return of AUTH_ACL_FAILED");
      return AUTH_ACL_FAILED;
   }
   if (debug && peer != bogus_peer) {
      ast_verbose("Found peer '%s' for '%s' from %s\n",
         peer->name, of, ast_sockaddr_stringify(&p->recv));
   }

   /* XXX what about p->prefs = peer->prefs; ? */
   /* Set Frame packetization */
   if (p->rtp) {
      ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
      p->autoframing = peer->autoframing;
   }

   /* Take the peer */
   ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) && p->udptl) {
      p->t38_maxdatagram = peer->t38_maxdatagram;
      set_t38_capabilities(p);
   }

   ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &p->dtls_cfg);

   /* Copy SIP extensions profile to peer */
   /* XXX is this correct before a successful auth ? */
   if (p->sipoptions)
      peer->sipoptions = p->sipoptions;

   do_setnat(p);

   ast_string_field_set(p, peersecret, peer->secret);
   ast_string_field_set(p, peermd5secret, peer->md5secret);
   ast_string_field_set(p, subscribecontext, peer->subscribecontext);
   ast_string_field_set(p, mohinterpret, peer->mohinterpret);
   ast_string_field_set(p, mohsuggest, peer->mohsuggest);
   if (!ast_strlen_zero(peer->parkinglot)) {
      ast_string_field_set(p, parkinglot, peer->parkinglot);
   }
   ast_string_field_set(p, engine, peer->engine);
   p->disallowed_methods = peer->disallowed_methods;
   set_pvt_allowed_methods(p, req);
   ast_cc_copy_config_params(p->cc_params, peer->cc_params);
   if (peer->callingpres)  /* Peer calling pres setting will override RPID */
      p->callingpres = peer->callingpres;
   if (peer->maxms && peer->lastms)
      p->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      p->timer_t1 = peer->timer_t1;

   /* Set timer B to control transaction timeouts */
   if (peer->timer_b)
      p->timer_b = peer->timer_b;
   else
      p->timer_b = 64 * p->timer_t1;

   p->allowtransfer = peer->allowtransfer;

   if (ast_test_flag(&peer->flags[0], SIP_INSECURE_INVITE)) {
      /* Pretend there is no required authentication */
      ast_string_field_set(p, peersecret, NULL);
      ast_string_field_set(p, peermd5secret, NULL);
   }
   if (!(res = check_auth(p, req, peer->name, p->peersecret, p->peermd5secret, sipmethod, uri2, reliable))) {
      /* If we have a call limit, set flag */
      if (peer->call_limit)
         ast_set_flag(&p->flags[0], SIP_CALL_LIMIT);
      ast_string_field_set(p, peername, peer->name);
      ast_string_field_set(p, authname, peer->name);

      ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &p->dtls_cfg);

      if (sipmethod == SIP_INVITE) {
         /* destroy old channel vars and copy in new ones. */
         ast_variables_destroy(p->chanvars);
         p->chanvars = copy_vars(peer->chanvars);
      }

      if (authpeer) {
         ao2_t_ref(peer, 1, "copy pointer into (*authpeer)");
         (*authpeer) = peer;  /* Add a ref to the object here, to keep it in memory a bit longer if it is realtime */
      }

      if (!ast_strlen_zero(peer->username)) {
         ast_string_field_set(p, username, peer->username);
         /* Use the default username for authentication on outbound calls */
         /* XXX this takes the name from the caller... can we override ? */
         ast_string_field_set(p, authname, peer->username);
      }
      if (!get_rpid(p, req)) {
         if (!ast_strlen_zero(peer->cid_num)) {
            char *tmp = ast_strdupa(peer->cid_num);
            if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp))
               ast_shrink_phone_number(tmp);
            ast_string_field_set(p, cid_num, tmp);
         }
         if (!ast_strlen_zero(peer->cid_name))
            ast_string_field_set(p, cid_name, peer->cid_name);
         if (peer->callingpres)
            p->callingpres = peer->callingpres;
      }
      if (!ast_strlen_zero(peer->cid_tag)) {
         ast_string_field_set(p, cid_tag, peer->cid_tag);
      }
      ast_string_field_set(p, fullcontact, peer->fullcontact);
      if (!ast_strlen_zero(peer->context)) {
         ast_string_field_set(p, context, peer->context);
      }
      if (!ast_strlen_zero(peer->messagecontext)) {
         ast_string_field_set(p, messagecontext, peer->messagecontext);
      }
      if (!ast_strlen_zero(peer->mwi_from)) {
         ast_string_field_set(p, mwi_from, peer->mwi_from);
      }
      ast_string_field_set(p, peersecret, peer->secret);
      ast_string_field_set(p, peermd5secret, peer->md5secret);
      ast_string_field_set(p, language, peer->language);
      ast_string_field_set(p, accountcode, peer->accountcode);
      p->amaflags = peer->amaflags;
      p->callgroup = peer->callgroup;
      p->pickupgroup = peer->pickupgroup;
      ast_unref_namedgroups(p->named_callgroups);
      p->named_callgroups = ast_ref_namedgroups(peer->named_callgroups);
      ast_unref_namedgroups(p->named_pickupgroups);
      p->named_pickupgroups = ast_ref_namedgroups(peer->named_pickupgroups);
      ast_format_cap_copy(p->caps, peer->caps);
      ast_format_cap_copy(p->jointcaps, peer->caps);
      p->prefs = peer->prefs;
      ast_copy_string(p->zone, peer->zone, sizeof(p->zone));
      if (peer->maxforwards > 0) {
         p->maxforwards = peer->maxforwards;
      }
      if (!(ast_format_cap_is_empty(p->peercaps))) {
         struct ast_format_cap *tmp = ast_format_cap_joint(p->jointcaps, p->peercaps);
         struct ast_format_cap *tmp2;
         if (tmp) {
            tmp2 = p->jointcaps;
            p->jointcaps = tmp;
            ast_format_cap_destroy(tmp2);
         }
      }
      p->maxcallbitrate = peer->maxcallbitrate;
      if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
          (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
         p->noncodeccapability |= AST_RTP_DTMF;
      else
         p->noncodeccapability &= ~AST_RTP_DTMF;
      p->jointnoncodeccapability = p->noncodeccapability;
      p->rtptimeout = peer->rtptimeout;
      p->rtpholdtimeout = peer->rtpholdtimeout;
      p->rtpkeepalive = peer->rtpkeepalive;
      if (!dialog_initialize_rtp(p)) {
         if (p->rtp) {
            ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &peer->prefs);
            p->autoframing = peer->autoframing;
         }
      } else {
         res = AUTH_RTP_FAILED;
      }
   }
   sip_unref_peer(peer, "check_peer_ok: sip_unref_peer: tossing temp ptr to peer from sip_find_peer");
   return res;
}

static void check_pendings

(

struct sip_pvt *

p

)

[static]

Check pending actions on SIP call.

Note:

both sip_pvt and sip_pvt's owner channel (if present) must be locked for this function.

Definition at line 22480 of file chan_sip.c.

References ast_clear_flag, ast_debug, AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), ast_test_flag, FALSE, sip_scheddestroy(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), and TRUE.

Referenced by handle_incoming(), handle_response_invite(), reinvite_timeout(), and sip_reinvite_retry().

{
   if (ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
      if (p->reinviteid > -1) {
         /* Outstanding p->reinviteid timeout, so wait... */
         return;
      } else if (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA) {
         /* if we can't BYE, then this is really a pending CANCEL */
         p->invitestate = INV_CANCELLED;
         transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
         /* If the cancel occurred on an initial invite, cancel the pending BYE */
         if (!ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
            ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
         }
         /* Actually don't destroy us yet, wait for the 487 on our original
            INVITE, but do set an autodestruct just in case we never get it. */
      } else {
         /* We have a pending outbound invite, don't send something
          * new in-transaction, unless it is a pending reinvite, then
          * by the time we are called here, we should probably just hang up. */
         if (p->pendinginvite && !p->ongoing_reinvite)
            return;

         if (p->owner) {
            ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
         }
         /* Perhaps there is an SD change INVITE outstanding */
         transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);
         ast_clear_flag(&p->flags[0], SIP_PENDINGBYE);
      }
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else if (ast_test_flag(&p->flags[0], SIP_NEEDREINVITE)) {
      /* if we can't REINVITE, hold it for later */
      if (p->pendinginvite || p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA || p->waitid > 0) {
         ast_debug(2, "NOT Sending pending reinvite (yet) on '%s'\n", p->callid);
      } else {
         ast_debug(2, "Sending pending reinvite on '%s'\n", p->callid);
         /* Didn't get to reinvite yet, so do it now */
         transmit_reinvite_with_sdp(p, (p->t38.state == T38_LOCAL_REINVITE ? TRUE : FALSE), FALSE);
         ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   }
}

static int check_rtp_timeout	(	struct sip_pvt *	dialog,
		time_t	t
	)		[static]

helper function for the monitoring thread -- seems to be called with the assumption that the dialog is locked

Returns:

CMP_MATCH for items to be unlinked from dialogs_rtpcheck.

Todo:

Check video RTP keepalives

Do we need to move the lastrtptx to the RTP structure to have one for audio and one for video? It really does belong to the RTP structure.

Definition at line 28956 of file chan_sip.c.

References ast_channel_name(), ast_channel_trylock, ast_channel_uniqueid(), ast_channel_unlock, ast_log(), ast_rtp_instance_get_hold_timeout(), ast_rtp_instance_get_keepalive(), ast_rtp_instance_get_timeout(), ast_rtp_instance_sendcng(), ast_rtp_instance_set_hold_timeout(), ast_rtp_instance_set_timeout(), ast_sockaddr_isnull(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, ast_test_flag, CMP_MATCH, EVENT_FLAG_CALL, keepalive, LOG_NOTICE, and manager_event.

Referenced by dialog_checkrtp_cb().

{
   int timeout;
   int hold_timeout;
   int keepalive;

   if (!dialog->rtp) {
      /*
       * We have no RTP.  Since we don't do much with video RTP for
       * now, stop checking this dialog.
       */
      return CMP_MATCH;
   }

   /* If we have no active owner, no need to check timers */
   if (!dialog->owner) {
      return CMP_MATCH;
   }

   /* If the call is redirected outside Asterisk, no need to check timers */
   if (!ast_sockaddr_isnull(&dialog->redirip)) {
      return CMP_MATCH;
   }

   /* If the call is involved in a T38 fax session do not check RTP timeout */
   if (dialog->t38.state == T38_ENABLED) {
      return CMP_MATCH;
   }
   /* If the call is not in UP state return for later check. */
   if (ast_channel_state(dialog->owner) != AST_STATE_UP) {
      return 0;
   }

   /* Store these values locally to avoid multiple function calls */
   timeout = ast_rtp_instance_get_timeout(dialog->rtp);
   hold_timeout = ast_rtp_instance_get_hold_timeout(dialog->rtp);
   keepalive = ast_rtp_instance_get_keepalive(dialog->rtp);

   /* If we have no timers set, return now */
   if (!keepalive && !timeout && !hold_timeout) {
      return CMP_MATCH;
   }

   /* Check AUDIO RTP keepalives */
   if (dialog->lastrtptx && keepalive && (t > dialog->lastrtptx + keepalive)) {
      /* Need to send an empty RTP packet */
      dialog->lastrtptx = time(NULL);
      ast_rtp_instance_sendcng(dialog->rtp, 0);
   }

   /*! \todo Check video RTP keepalives

      Do we need to move the lastrtptx to the RTP structure to have one for audio and one
      for video? It really does belong to the RTP structure.
   */

   /* Check AUDIO RTP timers */
   if (dialog->lastrtprx && (timeout || hold_timeout) && (t > dialog->lastrtprx + timeout)) {
      if (!ast_test_flag(&dialog->flags[1], SIP_PAGE2_CALL_ONHOLD) || (hold_timeout && (t > dialog->lastrtprx + hold_timeout))) {
         /* Needs a hangup */
         if (timeout) {
            if (!dialog->owner || ast_channel_trylock(dialog->owner)) {
               /*
                * Don't block, just try again later.
                * If there was no owner, the call is dead already.
                */
               return 0;
            }
            ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
               ast_channel_name(dialog->owner), (long) (t - dialog->lastrtprx));
            manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: RTPTimeout\r\n"
                  "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(dialog->owner), ast_channel_uniqueid(dialog->owner));
            /* Issue a softhangup */
            ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
            ast_channel_unlock(dialog->owner);
            /* forget the timeouts for this call, since a hangup
               has already been requested and we don't want to
               repeatedly request hangups
            */
            ast_rtp_instance_set_timeout(dialog->rtp, 0);
            ast_rtp_instance_set_hold_timeout(dialog->rtp, 0);
            if (dialog->vrtp) {
               ast_rtp_instance_set_timeout(dialog->vrtp, 0);
               ast_rtp_instance_set_hold_timeout(dialog->vrtp, 0);
            }
            /* finally unlink the dialog from dialogs_rtpcheck. */
            return CMP_MATCH;
         }
      }
   }
   return 0;
}

static int check_sip_domain	(	const char *	domain,
		char *	context,
		size_t	len
	)		[static]

check_sip_domain: Check if domain part of uri is local to our server

Definition at line 30305 of file chan_sip.c.

References ast_copy_string(), AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, and ast_strlen_zero().

Referenced by func_check_sipdomain(), get_destination(), get_realm(), handle_request_refer(), and register_verify().

{
   struct domain *d;
   int result = 0;

   AST_LIST_LOCK(&domain_list);
   AST_LIST_TRAVERSE(&domain_list, d, list) {
      if (strcasecmp(d->domain, domain)) {
         continue;
      }

      if (len && !ast_strlen_zero(d->context))
         ast_copy_string(context, d->context, len);

      result = 1;
      break;
   }
   AST_LIST_UNLOCK(&domain_list);

   return result;
}

static int check_user	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable,
		struct ast_sockaddr *	addr
	)		[static]

Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.

Definition at line 18528 of file chan_sip.c.

References check_user_full().

Referenced by handle_request_options(), handle_request_publish(), and receive_message().

{
   return check_user_full(p, req, sipmethod, uri, reliable, addr, NULL);
}

static enum check_auth_result check_user_full	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	sipmethod,
		const char *	uri,
		enum xmittype	reliable,
		struct ast_sockaddr *	addr,
		struct sip_peer **	authpeer
	)		[static]

Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.

Returns:

0 on success, non-zero on failure

Definition at line 18398 of file chan_sip.c.

References ast_copy_string(), ast_is_shrinkable_phonenumber(), ast_log(), ast_set_flag, ast_shrink_phone_number(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_contact(), check_peer_ok(), cid_name, cid_num, dialog_initialize_rtp(), exten, extract_host_from_hostport(), get_calleridname(), get_in_brackets(), get_rpid(), global_rtpholdtimeout, global_rtpkeepalive, global_rtptimeout, LOG_ERROR, LOG_NOTICE, name, parse_uri_legacy_check(), sip_cfg, sip_get_header(), SIP_PEDANTIC_DECODE, and terminate_uri().

Referenced by check_user(), handle_request_invite(), and handle_request_subscribe().

{
   char from[256], *of, *name, *unused_password, *domain;
   enum check_auth_result res = AUTH_DONT_KNOW;
   char calleridname[256];
   char *uri2 = ast_strdupa(uri);

   terminate_uri(uri2); /* trim extra stuff */

   ast_copy_string(from, sip_get_header(req, "From"), sizeof(from));
   /* XXX here tries to map the username for invite things */

   /* strip the display-name portion off the beginning of the FROM header. */
   if (!(of = (char *) get_calleridname(from, calleridname, sizeof(calleridname)))) {
      ast_log(LOG_ERROR, "FROM header can not be parsed \n");
      return res;
   }

   if (calleridname[0]) {
      ast_string_field_set(p, cid_name, calleridname);
   }

   if (ast_strlen_zero(p->exten)) {
      char *t = uri2;
      if (!strncasecmp(t, "sip:", 4))
         t+= 4;
      else if (!strncasecmp(t, "sips:", 5))
         t += 5;
      ast_string_field_set(p, exten, t);
      t = strchr(p->exten, '@');
      if (t)
         *t = '\0';

      if (ast_strlen_zero(p->our_contact))
         build_contact(p);
   }

   of = get_in_brackets(of);

   /* save the URI part of the From header */
   ast_string_field_set(p, from, of);

   if (parse_uri_legacy_check(of, "sip:,sips:", &name, &unused_password, &domain, NULL)) {
      ast_log(LOG_NOTICE, "From address missing 'sip:', using it anyway\n");
   }

   SIP_PEDANTIC_DECODE(name);
   SIP_PEDANTIC_DECODE(domain);

   extract_host_from_hostport(&domain);

   if (ast_strlen_zero(domain)) {
      /* <sip:name@[EMPTY]>, never good */
      ast_log(LOG_ERROR, "Empty domain name in FROM header\n");
      return res;
   }

   if (ast_strlen_zero(name)) {
      /* <sip:[EMPTY][@]hostport>. Asterisk 1.4 and 1.6 have always
       * treated that as a username, so we continue the tradition:
       * uri is now <sip:host@hostport>. */
      name = domain;
   } else {
      /* Non-empty name, try to get caller id from it */
      char *tmp = ast_strdupa(name);
      /* We need to be able to handle from-headers looking like
         <sip:8164444422;phone-context=+1@1.2.3.4:5060;user=phone;tag=SDadkoa01-gK0c3bdb43>
      */
      tmp = strsep(&tmp, ";");
      if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(tmp)) {
         ast_shrink_phone_number(tmp);
      }
      ast_string_field_set(p, cid_num, tmp);
   }

   if (global_match_auth_username) {
      /*
       * XXX This is experimental code to grab the search key from the
       * Auth header's username instead of the 'From' name, if available.
       * Do not enable this block unless you understand the side effects (if any!)
       * Note, the search for "username" should be done in a more robust way.
       * Note2, at the moment we check both fields, though maybe we should
       * pick one or another depending on the request ? XXX
       */
      const char *hdr = sip_get_header(req, "Authorization");
      if (ast_strlen_zero(hdr)) {
         hdr = sip_get_header(req, "Proxy-Authorization");
      }

      if (!ast_strlen_zero(hdr) && (hdr = strstr(hdr, "username=\""))) {
         ast_copy_string(from, hdr + strlen("username=\""), sizeof(from));
         name = from;
         name = strsep(&name, "\"");
      }
   }

   res = check_peer_ok(p, name, req, sipmethod, addr,
         authpeer, reliable, calleridname, uri2);
   if (res != AUTH_DONT_KNOW) {
      return res;
   }

   /* Finally, apply the guest policy */
   if (sip_cfg.allowguest) {
      /* Ignore check_return warning from Coverity for get_rpid below. */
      get_rpid(p, req);
      p->rtptimeout = global_rtptimeout;
      p->rtpholdtimeout = global_rtpholdtimeout;
      p->rtpkeepalive = global_rtpkeepalive;
      if (!dialog_initialize_rtp(p)) {
         res = AUTH_SUCCESSFUL;
      } else {
         res = AUTH_RTP_FAILED;
      }
   } else {
      res = AUTH_SECRET_FAILED; /* we don't want any guests, authentication will fail */
   }

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT)) {
      ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
   }

   return res;
}

static void check_via	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)		[static]

check Via: header for hostname, port and rport request/answer

Definition at line 18087 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_set_flag, ast_skip_blanks(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_verbose(), check_for_nat(), LOG_WARNING, sip_debug_test_pvt(), sip_get_header(), sip_nat_mode(), and sip_real_dst().

Referenced by handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_register(), handle_request_subscribe(), sip_alloc(), and transmit_response_using_temp().

{
   char via[512];
   char *c, *maddr;
   struct ast_sockaddr tmp = { { 0, } };
   uint16_t port;

   ast_copy_string(via, sip_get_header(req, "Via"), sizeof(via));

   /* If this is via WebSocket we don't use the Via header contents at all */
   if (!strncasecmp(via, "SIP/2.0/WS", 10)) {
      return;
   }

   /* Work on the leftmost value of the topmost Via header */
   c = strchr(via, ',');
   if (c)
      *c = '\0';

   /* Check for rport */
   c = strstr(via, ";rport");
   if (c && (c[6] != '=')) { /* rport query, not answer */
      ast_set_flag(&p->flags[1], SIP_PAGE2_RPORT_PRESENT);
      ast_set_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT);
   }

   /* Check for maddr */
   maddr = strstr(via, "maddr=");
   if (maddr) {
      maddr += 6;
      c = maddr + strspn(maddr, "abcdefghijklmnopqrstuvwxyz"
                      "ABCDEFGHIJKLMNOPQRSTUVWXYZ0123456789-.:[]");
      *c = '\0';
   }

   c = strchr(via, ';');
   if (c)
      *c = '\0';

   c = strchr(via, ' ');
   if (c) {
      *c = '\0';
      c = ast_skip_blanks(c+1);
      if (strcasecmp(via, "SIP/2.0/UDP") && strcasecmp(via, "SIP/2.0/TCP") && strcasecmp(via, "SIP/2.0/TLS")) {
         ast_log(LOG_WARNING, "Don't know how to respond via '%s'\n", via);
         return;
      }

      if (maddr && ast_sockaddr_resolve_first(&p->sa, maddr, 0)) {
         p->sa = p->recv;
      }

      if (ast_sockaddr_resolve_first(&tmp, c, 0)) {
         ast_log(LOG_WARNING, "Could not resolve socket address for '%s'\n", c);
         port = STANDARD_SIP_PORT;
      } else if (!(port = ast_sockaddr_port(&tmp))) {
         port = STANDARD_SIP_PORT;
         ast_sockaddr_set_port(&tmp, port);
      }

      ast_sockaddr_set_port(&p->sa, port);

      check_for_nat(&tmp, p);

      if (sip_debug_test_pvt(p)) {
         ast_verbose("Sending to %s (%s)\n",
                ast_sockaddr_stringify(sip_real_dst(p)),
                sip_nat_mode(p));
      }
   }
}

static void cleanup_all_regs

(

void

)

[static]

Definition at line 31389 of file chan_sip.c.

References ast_debug, ast_dnsmgr_release(), AST_SCHED_DEL_UNREF, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_UNLOCK, ASTOBJ_WRLOCK, dialog_unlink_all(), registry_unref(), and regl.

Referenced by reload_config(), and unload_module().

{
      /* First, destroy all outstanding registry calls */
      /* This is needed, since otherwise active registry entries will not be destroyed */
      ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {  /* regl is locked */
            ASTOBJ_WRLOCK(iterator); /* now regl is locked, and the object is also locked */
            if (iterator->call) {
               ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", iterator->username, iterator->hostname);
               /* This will also remove references to the registry */
               dialog_unlink_all(iterator->call);
               iterator->call = dialog_unref(iterator->call, "remove iterator->call from registry traversal");
            }
            if (iterator->expire > -1) {
               AST_SCHED_DEL_UNREF(sched, iterator->expire, registry_unref(iterator, "reg ptr unref from reload config"));
            }
            if (iterator->timeout > -1) {
               AST_SCHED_DEL_UNREF(sched, iterator->timeout, registry_unref(iterator, "reg ptr unref from reload config"));
            }
            if (iterator->dnsmgr) {
               ast_dnsmgr_release(iterator->dnsmgr);
               iterator->dnsmgr = NULL;
               registry_unref(iterator, "reg ptr unref from dnsmgr");
            }
            ASTOBJ_UNLOCK(iterator);
      } while(0));
}

static void cleanup_stale_contexts	(	char *	new,
		char *	old
	)		[static]

Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.

Definition at line 19472 of file chan_sip.c.

References ast_context_destroy(), ast_context_find(), ast_copy_string(), and AST_MAX_CONTEXT.

Referenced by reload_config().

{
   char *oldcontext, *newcontext, *stalecontext, *stringp, newlist[AST_MAX_CONTEXT];

   while ((oldcontext = strsep(&old, "&"))) {
      stalecontext = '\0';
      ast_copy_string(newlist, new, sizeof(newlist));
      stringp = newlist;
      while ((newcontext = strsep(&stringp, "&"))) {
         if (!strcmp(newcontext, oldcontext)) {
            /* This is not the context you're looking for */
            stalecontext = '\0';
            break;
         } else if (strcmp(newcontext, oldcontext)) {
            stalecontext = oldcontext;
         }
         
      }
      if (stalecontext)
         ast_context_destroy(ast_context_find(stalecontext), "SIP");
   }
}

static void clear_peer_mailboxes

(

struct sip_peer *

peer

)

[static]

Destroy all peer-related mailbox subscriptions

Definition at line 5172 of file chan_sip.c.

References AST_LIST_REMOVE_HEAD, destroy_mailbox(), and mailbox.

Referenced by set_peer_defaults(), and sip_destroy_peer().

{
   struct sip_mailbox *mailbox;

   while ((mailbox = AST_LIST_REMOVE_HEAD(&peer->mailboxes, entry)))
      destroy_mailbox(mailbox);
}

static void clear_sip_domains

(

void

)

[static]

Clear our domain list (at reload)

Definition at line 30328 of file chan_sip.c.

References ast_free, AST_LIST_LOCK, AST_LIST_REMOVE_HEAD, and AST_LIST_UNLOCK.

Referenced by reload_config(), and unload_module().

{
   struct domain *d;

   AST_LIST_LOCK(&domain_list);
   while ((d = AST_LIST_REMOVE_HEAD(&domain_list, list)))
      ast_free(d);
   AST_LIST_UNLOCK(&domain_list);
}

static char * complete_sip_notify	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip notify' CLI.

Definition at line 21149 of file chan_sip.c.

References ast_category_browse(), ast_strdup, and complete_sip_peer().

Referenced by sip_cli_notify().

{
   char *c = NULL;

   if (pos == 2) {
      int which = 0;
      char *cat = NULL;
      int wordlen = strlen(word);

      /* do completion for notify type */

      if (!notify_types)
         return NULL;
      
      while ( (cat = ast_category_browse(notify_types, cat)) ) {
         if (!strncasecmp(word, cat, wordlen) && ++which > state) {
            c = ast_strdup(cat);
            break;
         }
      }
      return c;
   }

   if (pos > 2)
      return complete_sip_peer(word, state, 0);

   return NULL;
}

static char * complete_sip_peer	(	const char *	word,
		int	state,
		int	flags2
	)		[static]

Do completion on peer name.

Definition at line 21072 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, and sip_unref_peer().

Referenced by complete_sip_notify(), complete_sip_show_peer(), sip_do_debug(), and sip_prune_realtime().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator i = ao2_iterator_init(peers, 0);
   struct sip_peer *peer;

   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, peer->name, wordlen) &&
            (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
            ++which > state)
         result = ast_strdup(peer->name);
      sip_unref_peer(peer, "toss iterator peer ptr before break");
      if (result) {
         break;
      }
   }
   ao2_iterator_destroy(&i);
   return result;
}

static char * complete_sip_registered_peer	(	const char *	word,
		int	state,
		int	flags2
	)		[static]

Do completion on registered peer name.

Definition at line 21096 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, ast_test_flag, and sip_unref_peer().

Referenced by complete_sip_unregister().

{
       char *result = NULL;
       int wordlen = strlen(word);
       int which = 0;
       struct ao2_iterator i;
       struct sip_peer *peer;
       
       i = ao2_iterator_init(peers, 0);
       while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
          if (!strncasecmp(word, peer->name, wordlen) &&
         (!flags2 || ast_test_flag(&peer->flags[1], flags2)) &&
         ++which > state && peer->expire > 0)
             result = ast_strdup(peer->name);
          if (result) {
             sip_unref_peer(peer, "toss iterator peer ptr before break");
             break;
          }
          sip_unref_peer(peer, "toss iterator peer ptr");
       }
       ao2_iterator_destroy(&i);
       return result;
}

static char * complete_sip_show_history	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip show history' CLI.

Definition at line 21121 of file chan_sip.c.

References complete_sipch().

Referenced by sip_show_history().

{
   if (pos == 3)
      return complete_sipch(line, word, pos, state);

   return NULL;
}

static char * complete_sip_show_peer	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip show peer' CLI.

Definition at line 21130 of file chan_sip.c.

References complete_sip_peer().

Referenced by sip_qualify_peer(), and sip_show_peer().

{
   if (pos == 3) {
      return complete_sip_peer(word, state, 0);
   }

   return NULL;
}

static char* complete_sip_show_user	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip show user' CLI.

Definition at line 20317 of file chan_sip.c.

References complete_sip_user().

Referenced by sip_show_user().

{
   if (pos == 3)
      return complete_sip_user(word, state);

   return NULL;
}

static char * complete_sip_unregister	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip unregister' CLI.

Definition at line 21140 of file chan_sip.c.

References complete_sip_registered_peer().

Referenced by sip_unregister().

{
       if (pos == 2)
               return complete_sip_registered_peer(word, state, 0);

       return NULL;
}

static char* complete_sip_user	(	const char *	word,
		int	state
	)		[static]

Do completion on user name.

Definition at line 20287 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_strdup, sip_unref_peer(), and user.

Referenced by complete_sip_show_user().

{
   char *result = NULL;
   int wordlen = strlen(word);
   int which = 0;
   struct ao2_iterator user_iter;
   struct sip_peer *user;

   user_iter = ao2_iterator_init(peers, 0);
   while ((user = ao2_t_iterator_next(&user_iter, "iterate thru peers table"))) {
      ao2_lock(user);
      if (!(user->type & SIP_TYPE_USER)) {
         ao2_unlock(user);
         sip_unref_peer(user, "complete sip user");
         continue;
      }
      /* locking of the object is not required because only the name and flags are being compared */
      if (!strncasecmp(word, user->name, wordlen) && ++which > state) {
         result = ast_strdup(user->name);
      }
      ao2_unlock(user);
      sip_unref_peer(user, "complete sip user");
      if (result) {
         break;
      }
   }
   ao2_iterator_destroy(&user_iter);
   return result;
}

static char* complete_sipch	(	const char *	line,
		const char *	word,
		int	pos,
		int	state
	)		[static]

Support routine for 'sip show channel' and 'sip show history' CLI This is in charge of generating all strings that match a prefix in the given position. As many functions of this kind, each invokation has O(state) time complexity so be careful in using it.

Definition at line 21042 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ast_strdup, sip_pvt_lock, and sip_pvt_unlock.

Referenced by complete_sip_show_history(), and sip_show_channel().

{
   int which=0;
   struct sip_pvt *cur;
   char *c = NULL;
   int wordlen = strlen(word);
   struct ao2_iterator i;

   if (pos != 3) {
      return NULL;
   }

   i = ao2_iterator_init(dialogs, 0);
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);
      if (!strncasecmp(word, cur->callid, wordlen) && ++which > state) {
         c = ast_strdup(cur->callid);
         sip_pvt_unlock(cur);
         dialog_unref(cur, "drop ref in iterator loop break");
         break;
      }
      sip_pvt_unlock(cur);
      dialog_unref(cur, "drop ref in iterator loop");
   }
   ao2_iterator_destroy(&i);
   return c;
}

static int construct_pidf_body	(	enum sip_cc_publish_state	state,
		char *	pidf_body,
		size_t	size,
		const char *	presentity
	)		[static]

Definition at line 2056 of file chan_sip.c.

References ast_copy_string(), ast_str_alloca, ast_str_append(), ast_str_buffer(), and generate_random_string().

Referenced by handle_cc_notify(), sip_cc_monitor_suspend(), and sip_cc_monitor_unsuspend().

{
   struct ast_str *body = ast_str_alloca(size);
   char tuple_id[32];

   generate_random_string(tuple_id, sizeof(tuple_id));

   /* We'll make this a bare-bones pidf body. In state_notify_build_xml, the PIDF
    * body gets a lot more extra junk that isn't necessary, so we'll leave it out here.
    */
   ast_str_append(&body, 0, "<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n");
   /* XXX The entity attribute is currently set to the peer name associated with the
    * dialog. This is because we currently only call this function for call-completion
    * PUBLISH bodies. In such cases, the entity is completely disregarded. For other
    * event packages, it may be crucial to have a proper URI as the presentity so this
    * should be revisited as support is expanded.
    */
   ast_str_append(&body, 0, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" entity=\"%s\">\n", presentity);
   ast_str_append(&body, 0, "<tuple id=\"%s\">\n", tuple_id);
   ast_str_append(&body, 0, "<status><basic>%s</basic></status>\n", state == CC_OPEN ? "open" : "closed");
   ast_str_append(&body, 0, "</tuple>\n");
   ast_str_append(&body, 0, "</presence>\n");
   ast_copy_string(pidf_body, ast_str_buffer(body), size);
   return 0;
}

static int copy_all_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)		[static]

Copy all headers from one request to another.

Definition at line 11474 of file chan_sip.c.

References __get_header(), add_header(), and ast_strlen_zero().

Referenced by respprep().

{
   int start = 0;
   int copied = 0;
   for (;;) {
      const char *tmp = __get_header(orig, field, &start);

      if (ast_strlen_zero(tmp))
         break;
      /* Add what we're responding to */
      add_header(req, field, tmp);
      copied++;
   }
   return copied ? 0 : -1;
}

static int copy_header	(	struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)		[static]

Copy one header field from one request to another.

Definition at line 11463 of file chan_sip.c.

References add_header(), ast_log(), ast_strlen_zero(), LOG_NOTICE, and sip_get_header().

Referenced by reqprep(), and respprep().

{
   const char *tmp = sip_get_header(orig, field);

   if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
      return add_header(req, field, tmp);
   ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
   return -1;
}

static void copy_request	(	struct sip_request *	dst,
		const struct sip_request *	src
	)		[static]

copy SIP request (mostly used to save request for responses)

Definition at line 13568 of file chan_sip.c.

References ast_str_copy_string(), ast_str_create(), and ast_str_strlen().

Referenced by forked_invite_init(), handle_request_bye(), handle_request_invite(), handle_request_options(), handle_request_register(), handle_request_subscribe(), initialize_initreq(), parse_copy(), receive_message(), sip_park(), and sip_tls_read().

{
   /* XXX this function can encounter memory allocation errors, perhaps it
    * should return a value */

   struct ast_str *duplicate = dst->data;
   struct ast_str *duplicate_content = dst->content;

   /* copy the entire request then restore the original data and content
    * members from the dst request */
   *dst = *src;
   dst->data = duplicate;
   dst->content = duplicate_content;

   /* copy the data into the dst request */
   if (!dst->data && !(dst->data = ast_str_create(ast_str_strlen(src->data) + 1))) {
      return;
   }
   ast_str_copy_string(&dst->data, src->data);

   /* copy the content into the dst request (if it exists) */
   if (src->content) {
      if (!dst->content && !(dst->content = ast_str_create(ast_str_strlen(src->content) + 1))) {
         return;
      }
      ast_str_copy_string(&dst->content, src->content);
   }
}

static void copy_socket_data	(	struct sip_socket *	to_sock,
		const struct sip_socket *	from_sock
	)		[static]

Definition at line 5819 of file chan_sip.c.

References ao2_ref, ast_websocket_ref(), and ast_websocket_unref().

Referenced by create_addr_from_peer(), handle_request_do(), parse_register_contact(), sip_poke_peer(), and transmit_response_using_temp().

{
   if (to_sock->tcptls_session) {
      ao2_ref(to_sock->tcptls_session, -1);
      to_sock->tcptls_session = NULL;
   } else if (to_sock->ws_session) {
      ast_websocket_unref(to_sock->ws_session);
      to_sock->ws_session = NULL;
   }

   if (from_sock->tcptls_session) {
      ao2_ref(from_sock->tcptls_session, +1);
   } else if (from_sock->ws_session) {
      ast_websocket_ref(from_sock->ws_session);
   }

   *to_sock = *from_sock;
}

static struct ast_variable * copy_vars

(

struct ast_variable *

src

)

[static, read]

duplicate a list of channel variables,

Returns:

the copy.

Definition at line 2439 of file chan_sip.c.

References ast_variable_new(), and ast_variable::next.

Referenced by check_peer_ok(), and create_addr_from_peer().

{
   struct ast_variable *res = NULL, *tmp, *v = NULL;

   for (v = src ; v ; v = v->next) {
      if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
         tmp->next = res;
         res = tmp;
      }
   }
   return res;
}

static int copy_via_headers	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const struct sip_request *	orig,
		const char *	field
	)		[static]

Copy SIP VIA Headers from the request to the response.

Note:

If the client indicates that it wishes to know the port we received from, it adds ;rport without an argument to the topmost via header. We need to add the port number (from our point of view) to that parameter.

	We always add ;received=<ip address> to the topmost via header.

Received: RFC 3261, rport RFC 3581

Definition at line 11498 of file chan_sip.c.

References __get_header(), add_header(), ast_copy_string(), ast_log(), ast_sockaddr_port, ast_sockaddr_stringify_addr_remote(), ast_strlen_zero(), ast_test_flag, and LOG_NOTICE.

Referenced by respprep().

{
   int copied = 0;
   int start = 0;

   for (;;) {
      char new[512];
      const char *oh = __get_header(orig, field, &start);

      if (ast_strlen_zero(oh))
         break;

      if (!copied) { /* Only check for empty rport in topmost via header */
         char leftmost[512], *others, *rport;

         /* Only work on leftmost value */
         ast_copy_string(leftmost, oh, sizeof(leftmost));
         others = strchr(leftmost, ',');
         if (others)
             *others++ = '\0';

         /* Find ;rport;  (empty request) */
         rport = strstr(leftmost, ";rport");
         if (rport && *(rport+6) == '=')
            rport = NULL;     /* We already have a parameter to rport */

         if (((ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) || (rport && ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)))) {
            /* We need to add received port - rport */
            char *end;

            rport = strstr(leftmost, ";rport");

            if (rport) {
               end = strchr(rport + 1, ';');
               if (end)
                  memmove(rport, end, strlen(end) + 1);
               else
                  *rport = '\0';
            }

            /* Add rport to first VIA header if requested */
            snprintf(new, sizeof(new), "%s;received=%s;rport=%d%s%s",
               leftmost, ast_sockaddr_stringify_addr_remote(&p->recv),
               ast_sockaddr_port(&p->recv),
               others ? "," : "", others ? others : "");
         } else {
            /* We should *always* add a received to the topmost via */
            snprintf(new, sizeof(new), "%s;received=%s%s%s",
               leftmost, ast_sockaddr_stringify_addr_remote(&p->recv),
               others ? "," : "", others ? others : "");
         }
         oh = new;   /* the header to copy */
      }  /* else add the following via headers untouched */
      add_header(req, field, oh);
      copied++;
   }
   if (!copied) {
      ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
      return -1;
   }
   return 0;
}

static int create_addr	(	struct sip_pvt *	dialog,
		const char *	opeer,
		struct ast_sockaddr *	addr,
		int	newdialog
	)		[static]

create address structure from device name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success

Todo:

Fix this function. When we ask for SRV, we should check all transports In the future, we should first check NAPTR to find out transport preference

Definition at line 6125 of file chan_sip.c.

References AST_APP_ARG, ast_check_digits(), AST_DECLARE_APP_ARGS, ast_get_srv(), ast_log(), AST_NONSTANDARD_RAW_ARGS, ast_sockaddr_copy(), ast_sockaddr_port, ast_sockaddr_resolve_first_transport(), ast_sockaddr_set_port, ast_string_field_set, bindaddr, create_addr_from_peer(), default_sip_port(), dialog_initialize_rtp(), FALSE, get_srv_protocol(), get_srv_service(), global_rtpholdtimeout, global_rtpkeepalive, global_rtptimeout, global_t1, global_timer_b, LOG_WARNING, obproxy_get(), ref_proxy(), service, set_socket_transport(), sip_cfg, sip_find_peer(), sip_ref_peer(), sip_unref_peer(), and TRUE.

Referenced by __sip_subscribe_mwi_do(), manager_sipnotify(), sip_cc_monitor_request_cc(), sip_cli_notify(), sip_msg_send(), sip_request_call(), transmit_publish(), and transmit_register().

{
   struct sip_peer *peer;
   char *peername, *peername2, *hostn;
   char host[MAXHOSTNAMELEN];
   char service[MAXHOSTNAMELEN];
   int srv_ret = 0;
   int tportno;

   AST_DECLARE_APP_ARGS(hostport,
      AST_APP_ARG(host);
      AST_APP_ARG(port);
   );

   peername = ast_strdupa(opeer);
   peername2 = ast_strdupa(opeer);
   AST_NONSTANDARD_RAW_ARGS(hostport, peername2, ':');

   if (hostport.port)
      dialog->portinuri = 1;

   dialog->timer_t1 = global_t1; /* Default SIP retransmission timer T1 (RFC 3261) */
   dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
   peer = sip_find_peer(peername, NULL, TRUE, FINDPEERS, FALSE, 0);

   if (peer) {
      int res;
      if (newdialog) {
         set_socket_transport(&dialog->socket, 0);
      }
      res = create_addr_from_peer(dialog, peer);
      dialog->relatedpeer = sip_ref_peer(peer, "create_addr: setting dialog's relatedpeer pointer");
      sip_unref_peer(peer, "create_addr: unref peer from sip_find_peer hashtab lookup");
      return res;
   } else if (ast_check_digits(peername)) {
      /* Although an IPv4 hostname *could* be represented as a 32-bit integer, it is uncommon and
       * it makes dialing SIP/${EXTEN} for a peer that isn't defined resolve to an IP that is
       * almost certainly not intended. It is much better to just reject purely numeric hostnames */
      ast_log(LOG_WARNING, "Purely numeric hostname (%s), and not a peer--rejecting!\n", peername);
      return -1;
   } else {
      dialog->rtptimeout = global_rtptimeout;
      dialog->rtpholdtimeout = global_rtpholdtimeout;
      dialog->rtpkeepalive = global_rtpkeepalive;
      if (dialog_initialize_rtp(dialog)) {
         return -1;
      }
   }

   ast_string_field_set(dialog, tohost, hostport.host);
   dialog->allowed_methods &= ~sip_cfg.disallowed_methods;

   /* Get the outbound proxy information */
   ref_proxy(dialog, obproxy_get(dialog, NULL));

   if (addr) {
      /* This address should be updated using dnsmgr */
      ast_sockaddr_copy(&dialog->sa, addr);
   } else {

      /* Let's see if we can find the host in DNS. First try DNS SRV records,
         then hostname lookup */
      /*! \todo Fix this function. When we ask for SRV, we should check all transports
           In the future, we should first check NAPTR to find out transport preference
       */
      hostn = peername;
      /* Section 4.2 of RFC 3263 specifies that if a port number is specified, then
       * an A record lookup should be used instead of SRV.
       */
      if (!hostport.port && sip_cfg.srvlookup) {
         snprintf(service, sizeof(service), "_%s._%s.%s", 
             get_srv_service(dialog->socket.type),
             get_srv_protocol(dialog->socket.type), peername);
         if ((srv_ret = ast_get_srv(NULL, host, sizeof(host), &tportno,
                     service)) > 0) {
            hostn = host;
         }
      }

      if (ast_sockaddr_resolve_first_transport(&dialog->sa, hostn, 0, dialog->socket.type ? dialog->socket.type : SIP_TRANSPORT_UDP)) {
         ast_log(LOG_WARNING, "No such host: %s\n", peername);
         return -1;
      }

      if (srv_ret > 0) {
         ast_sockaddr_set_port(&dialog->sa, tportno);
      }
   }

   if (!dialog->socket.type)
      set_socket_transport(&dialog->socket, SIP_TRANSPORT_UDP);
   if (!dialog->socket.port) {
      dialog->socket.port = htons(ast_sockaddr_port(&bindaddr));
   }

   if (!ast_sockaddr_port(&dialog->sa)) {
      ast_sockaddr_set_port(&dialog->sa, default_sip_port(dialog->socket.type));
   }
   ast_sockaddr_copy(&dialog->recv, &dialog->sa);
   return 0;
}

static int create_addr_from_peer	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)		[static]

Create address structure from peer reference. This function copies data from peer to the dialog, so we don't have to look up the peer again from memory or database during the life time of the dialog.

Returns:

-1 on error, 0 on success.

Definition at line 5961 of file chan_sip.c.

References accountcode, ao2_lock, ao2_t_ref, ao2_unlock, ast_alloca, ast_cc_copy_config_params(), ast_copy_flags, ast_copy_string(), ast_duplicate_acl_list(), ast_format_cap_copy(), ast_ref_namedgroups(), ast_rtp_codecs_packetization_set(), ast_rtp_dtls_cfg_copy(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify_host_remote(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_unref_namedgroups(), change_callid_pvt(), check_request_transport, cid_name, cid_num, context, copy_socket_data(), copy_vars(), dialog_initialize_rtp(), language, mohinterpret, mohsuggest, obproxy_get(), parkinglot, and ref_proxy().

Referenced by create_addr(), and sip_send_mwi_to_peer().

{
   struct sip_auth_container *credentials;

   /* this checks that the dialog is contacting the peer on a valid
    * transport type based on the peers transport configuration,
    * otherwise, this function bails out */
   if (dialog->socket.type && check_request_transport(peer, dialog))
      return -1;
   copy_socket_data(&dialog->socket, &peer->socket);

   if (!(ast_sockaddr_isnull(&peer->addr) && ast_sockaddr_isnull(&peer->defaddr)) &&
       (!peer->maxms || ((peer->lastms >= 0)  && (peer->lastms <= peer->maxms)))) {
      dialog->sa = ast_sockaddr_isnull(&peer->addr) ? peer->defaddr : peer->addr;
      dialog->recv = dialog->sa;
   } else
      return -1;

   /* XXX TODO: get flags directly from peer only as they are needed using dialog->relatedpeer */
   ast_copy_flags(&dialog->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&dialog->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);
   ast_format_cap_copy(dialog->caps, peer->caps);
   dialog->prefs = peer->prefs;
   dialog->amaflags = peer->amaflags;

   ast_string_field_set(dialog, engine, peer->engine);

   ast_rtp_dtls_cfg_copy(&peer->dtls_cfg, &dialog->dtls_cfg);

   dialog->rtptimeout = peer->rtptimeout;
   dialog->rtpholdtimeout = peer->rtpholdtimeout;
   dialog->rtpkeepalive = peer->rtpkeepalive;
   if (dialog_initialize_rtp(dialog)) {
      return -1;
   }

   if (dialog->rtp) { /* Audio */
      ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
      ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      /* Set Frame packetization */
      ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(dialog->rtp), dialog->rtp, &dialog->prefs);
      dialog->autoframing = peer->autoframing;
   }

   /* XXX TODO: get fields directly from peer only as they are needed using dialog->relatedpeer */
   ast_string_field_set(dialog, peername, peer->name);
   ast_string_field_set(dialog, authname, peer->username);
   ast_string_field_set(dialog, username, peer->username);
   ast_string_field_set(dialog, peersecret, peer->secret);
   ast_string_field_set(dialog, peermd5secret, peer->md5secret);
   ast_string_field_set(dialog, mohsuggest, peer->mohsuggest);
   ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
   ast_string_field_set(dialog, tohost, peer->tohost);
   ast_string_field_set(dialog, fullcontact, peer->fullcontact);
   ast_string_field_set(dialog, accountcode, peer->accountcode);
   ast_string_field_set(dialog, context, peer->context);
   ast_string_field_set(dialog, cid_num, peer->cid_num);
   ast_string_field_set(dialog, cid_name, peer->cid_name);
   ast_string_field_set(dialog, cid_tag, peer->cid_tag);
   ast_string_field_set(dialog, mwi_from, peer->mwi_from);
   if (!ast_strlen_zero(peer->parkinglot)) {
      ast_string_field_set(dialog, parkinglot, peer->parkinglot);
   }
   ast_string_field_set(dialog, engine, peer->engine);
   ref_proxy(dialog, obproxy_get(dialog, peer));
   dialog->callgroup = peer->callgroup;
   dialog->pickupgroup = peer->pickupgroup;
   ast_unref_namedgroups(dialog->named_callgroups);
   dialog->named_callgroups = ast_ref_namedgroups(peer->named_callgroups);
   ast_unref_namedgroups(dialog->named_pickupgroups);
   dialog->named_pickupgroups = ast_ref_namedgroups(peer->named_pickupgroups);
   ast_copy_string(dialog->zone, peer->zone, sizeof(dialog->zone));
   dialog->allowtransfer = peer->allowtransfer;
   dialog->jointnoncodeccapability = dialog->noncodeccapability;

   /* Update dialog authorization credentials */
   ao2_lock(peer);
   credentials = peer->auth;
   if (credentials) {
      ao2_t_ref(credentials, +1, "Ref peer auth for dialog");
   }
   ao2_unlock(peer);
   ao2_lock(dialog);
   if (dialog->peerauth) {
      ao2_t_ref(dialog->peerauth, -1, "Unref old dialog peer auth");
   }
   dialog->peerauth = credentials;
   ao2_unlock(dialog);

   dialog->maxcallbitrate = peer->maxcallbitrate;
   dialog->disallowed_methods = peer->disallowed_methods;
   ast_cc_copy_config_params(dialog->cc_params, peer->cc_params);
   if (ast_strlen_zero(dialog->tohost))
      ast_string_field_set(dialog, tohost, ast_sockaddr_stringify_host_remote(&dialog->sa));
   if (!ast_strlen_zero(peer->fromdomain)) {
      ast_string_field_set(dialog, fromdomain, peer->fromdomain);
      if (!dialog->initreq.headers) {
         char *new_callid;
         char *tmpcall = ast_strdupa(dialog->callid);
         /* this sure looks to me like we are going to change the callid on this dialog!! */
         new_callid = strchr(tmpcall, '@');
         if (new_callid) {
            int callid_size;

            *new_callid = '\0';

            /* Change the dialog callid. */
            callid_size = strlen(tmpcall) + strlen(peer->fromdomain) + 2;
            new_callid = ast_alloca(callid_size);
            snprintf(new_callid, callid_size, "%s@%s", tmpcall, peer->fromdomain);
            change_callid_pvt(dialog, new_callid);
         }
      }
   }
   if (!ast_strlen_zero(peer->fromuser))
      ast_string_field_set(dialog, fromuser, peer->fromuser);
   if (!ast_strlen_zero(peer->language))
      ast_string_field_set(dialog, language, peer->language);
   /* Set timer T1 to RTT for this peer (if known by qualify=) */
   /* Minimum is settable or default to 100 ms */
   /* If there is a maxms and lastms from a qualify use that over a manual T1
      value. Otherwise, use the peer's T1 value. */
   if (peer->maxms && peer->lastms)
      dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
   else
      dialog->timer_t1 = peer->timer_t1;

   /* Set timer B to control transaction timeouts, the peer setting is the default and overrides
      the known timer */
   if (peer->timer_b)
      dialog->timer_b = peer->timer_b;
   else
      dialog->timer_b = 64 * dialog->timer_t1;

   if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
      dialog->noncodeccapability |= AST_RTP_DTMF;
   else
      dialog->noncodeccapability &= ~AST_RTP_DTMF;

   dialog->directmediaacl = ast_duplicate_acl_list(peer->directmediaacl);

   if (peer->call_limit)
      ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
   if (!dialog->portinuri)
      dialog->portinuri = peer->portinuri;
   dialog->chanvars = copy_vars(peer->chanvars);
   if (peer->fromdomainport)
      dialog->fromdomainport = peer->fromdomainport;
   dialog->callingpres = peer->callingpres;

   return 0;
}

static struct sip_epa_entry* create_epa_entry	(	const char *const	event_package,
		const char *const	destination
	)		[static, read]

Definition at line 978 of file chan_sip.c.

References ao2_t_alloc, ast_copy_string(), and find_static_data().

Referenced by sip_cc_monitor_suspend().

{
   struct sip_epa_entry *epa_entry;
   const struct epa_static_data *static_data;

   if (!(static_data = find_static_data(event_package))) {
      return NULL;
   }

   if (!(epa_entry = ao2_t_alloc(sizeof(*epa_entry), static_data->destructor, "Allocate new EPA entry"))) {
      return NULL;
   }

   epa_entry->static_data = static_data;
   ast_copy_string(epa_entry->destination, destination, sizeof(epa_entry->destination));
   return epa_entry;
}

static struct sip_esc_entry* create_esc_entry	(	struct event_state_compositor *	esc,
		struct sip_request *	req,
		const int	expires
	)		[static, read]

Definition at line 1104 of file chan_sip.c.

References ao2_alloc, ao2_ref, ast_sched_add(), create_new_sip_etag(), esc_entry_destructor(), event_state_compositor::name, and publish_expire().

Referenced by handle_sip_publish_initial().

{
   struct sip_esc_entry *esc_entry;
   int expires_ms;

   if (!(esc_entry = ao2_alloc(sizeof(*esc_entry), esc_entry_destructor))) {
      return NULL;
   }

   esc_entry->event = esc->name;

   expires_ms = expires * 1000;
   /* Bump refcount for scheduler */
   ao2_ref(esc_entry, +1);
   esc_entry->sched_id = ast_sched_add(sched, expires_ms, publish_expire, esc_entry);

   /* Note: This links the esc_entry into the ESC properly */
   create_new_sip_etag(esc_entry, 0);

   return esc_entry;
}

static void create_new_sip_etag	(	struct sip_esc_entry *	esc_entry,
		int	is_linked
	)		[static]

Definition at line 1091 of file chan_sip.c.

References ao2_link, ao2_unlink, ast_assert, ast_atomic_fetchadd_int(), event_state_compositor::compositor, and get_esc().

Referenced by create_esc_entry(), and transmit_response_with_sip_etag().

{
   int new_etag = ast_atomic_fetchadd_int(&esc_etag_counter, +1);
   struct event_state_compositor *esc = get_esc(esc_entry->event);

   ast_assert(esc != NULL);
   if (is_linked) {
      ao2_unlink(esc->compositor, esc_entry);
   }
   snprintf(esc_entry->entity_tag, sizeof(esc_entry->entity_tag), "%d", new_etag);
   ao2_link(esc->compositor, esc_entry);
}

static int default_sip_port

(

enum sip_transport

type

)

[inline, static]

The default sip port for the given transport.

Definition at line 6117 of file chan_sip.c.

Referenced by create_addr(), on_dns_update_peer(), and parse_register_contact().

{
   return type == SIP_TRANSPORT_TLS ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
}

static void deinit_req

(

struct sip_request *

req

)

[static]

Deinitialize SIP response/request.

Definition at line 11725 of file chan_sip.c.

References ast_free.

Referenced by __sip_destroy(), _sip_tcp_helper_thread(), send_request(), send_response(), sip_park(), sip_park_thread(), sip_websocket_callback(), and sipsock_read().

{
   if (req->data) {
      ast_free(req->data);
      req->data = NULL;
   }
   if (req->content) {
      ast_free(req->content);
      req->content = NULL;
   }
}

static void destroy_association

(

struct sip_peer *

peer

)

[static]

Remove registration data from realtime database or AST/DB when registration expires.

Definition at line 15718 of file chan_sip.c.

References ast_check_realtime(), ast_db_del(), ast_update_realtime(), SENTINEL, and sip_cfg.

Referenced by build_peer(), and expire_register().

{
   int realtimeregs = ast_check_realtime("sipregs");
   char *tablename = (realtimeregs) ? "sipregs" : "sippeers";

   if (!sip_cfg.ignore_regexpire) {
      if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
         ast_update_realtime(tablename, "name", peer->name, "fullcontact", "", "ipaddr", "", "port", "", "regseconds", "0", "regserver", "", "useragent", "", "lastms", "0", SENTINEL);
      } else {
         ast_db_del("SIP/Registry", peer->name);
         ast_db_del("SIP/PeerMethods", peer->name);
      }
   }
}

static void destroy_escs

(

void

)

[static]

Definition at line 1138 of file chan_sip.c.

References ao2_ref, ARRAY_LEN, and event_state_compositors.

Referenced by unload_module().

{
   int i;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      ao2_ref(event_state_compositors[i].compositor, -1);
   }
}

static void destroy_mailbox

(

struct sip_mailbox *

mailbox

)

[static]

Destroy mailbox subscriptions

Definition at line 5164 of file chan_sip.c.

References ast_event_unsubscribe(), and ast_free.

Referenced by build_peer(), and clear_peer_mailboxes().

{
   if (mailbox->event_sub)
      ast_event_unsubscribe(mailbox->event_sub);
   ast_free(mailbox);
}

static void destroy_msg_headers

(

struct sip_pvt *

pvt

)

[static]

Definition at line 12401 of file chan_sip.c.

References ast_free, and AST_LIST_REMOVE_HEAD.

Referenced by __sip_destroy(), sip_park_thread(), and sip_sendtext().

{
   struct sip_msg_hdr *doomed;

   while ((doomed = AST_LIST_REMOVE_HEAD(&pvt->msg_headers, next))) {
      ast_free(doomed);
   }
}

static void destroy_realm_authentication

(

void *

obj

)

[static]

Definition at line 30346 of file chan_sip.c.

References ast_free, and AST_LIST_REMOVE_HEAD.

Referenced by add_realm_authentication().

{
   struct sip_auth_container *credentials = obj;
   struct sip_auth *auth;

   while ((auth = AST_LIST_REMOVE_HEAD(&credentials->list, node))) {
      ast_free(auth);
   }
}

static int determine_firstline_parts

(

struct sip_request *

req

)

[static]

Parse first line of incoming SIP request.

Definition at line 13668 of file chan_sip.c.

References ast_debug, ast_skip_blanks(), ast_skip_nonblanks(), and ast_trim_blanks().

Referenced by parse_request().

{
   char *e = ast_skip_blanks(req->data->str);   /* there shouldn't be any */
   char *local_rlpart1;

   if (!*e)
      return -1;
   req->rlpart1 = e - req->data->str;  /* method or protocol */
   local_rlpart1 = e;
   e = ast_skip_nonblanks(e);
   if (*e)
      *e++ = '\0';
   /* Get URI or status code */
   e = ast_skip_blanks(e);
   if ( !*e )
      return -1;
   ast_trim_blanks(e);

   if (!strcasecmp(local_rlpart1, "SIP/2.0") ) { /* We have a response */
      if (strlen(e) < 3)   /* status code is 3 digits */
         return -1;
      req->rlpart2 = e - req->data->str;
   } else { /* We have a request */
      if ( *e == '<' ) { /* XXX the spec says it must not be in <> ! */
         ast_debug(3, "Oops. Bogus uri in <> %s\n", e);
         e++;
         if (!*e)
            return -1;
      }
      req->rlpart2 = e - req->data->str;  /* URI */
      e = ast_skip_nonblanks(e);
      if (*e)
         *e++ = '\0';
      e = ast_skip_blanks(e);
      if (strcasecmp(e, "SIP/2.0") ) {
         ast_debug(3, "Skipping packet - Bad request protocol %s\n", e);
         return -1;
      }
   }
   return 1;
}

static enum sip_publish_type determine_sip_publish_type	(	struct sip_request *	req,
		const char *const	event,
		const char *const	etag,
		const char *const	expires,
		int *	expires_int
	)		[static]

Definition at line 27128 of file chan_sip.c.

References ast_assert, ast_strlen_zero(), and DEFAULT_PUBLISH_EXPIRES.

Referenced by handle_request_publish().

{
   int etag_present = !ast_strlen_zero(etag);
   int body_present = req->lines > 0;

   ast_assert(expires_int != NULL);

   if (ast_strlen_zero(expires)) {
      /* Section 6, item 4, second bullet point of RFC 3903 says to
       * use a locally-configured default expiration if none is provided
       * in the request
       */
      *expires_int = DEFAULT_PUBLISH_EXPIRES;
   } else if (sscanf(expires, "%30d", expires_int) != 1) {
      return SIP_PUBLISH_UNKNOWN;
   }

   if (*expires_int == 0) {
      return SIP_PUBLISH_REMOVE;
   } else if (!etag_present && body_present) {
      return SIP_PUBLISH_INITIAL;
   } else if (etag_present && !body_present) {
      return SIP_PUBLISH_REFRESH;
   } else if (etag_present && body_present) {
      return SIP_PUBLISH_MODIFY;
   }

   return SIP_PUBLISH_UNKNOWN;
}

static int dialog_checkrtp_cb	(	void *	dialogobj,
		void *	arg,
		int	flags
	)		[static]

Check RTP Timeout on dialogs.

This is used with ao2_callback to check rtptimeout rtponholdtimeout and send rtpkeepalive packets.

Returns:

CMP_MATCH for items to be unlinked from dialogs_rtpcheck.

Definition at line 19503 of file chan_sip.c.

References check_rtp_timeout(), CMP_MATCH, sip_pvt_trylock, and sip_pvt_unlock.

Referenced by do_monitor().

{
   struct sip_pvt *dialog = dialogobj;
   time_t *t = arg;
   int match_status;

   if (sip_pvt_trylock(dialog)) {
      return 0;
   }

   if (dialog->rtp || dialog->vrtp) {
      match_status = check_rtp_timeout(dialog, *t);
   } else {
      /* Dialog has no active RTP or VRTP. unlink it from dialogs_rtpcheck. */
      match_status = CMP_MATCH;
   }
   sip_pvt_unlock(dialog);

   return match_status;
}

static int dialog_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Note:

The only member of the dialog used here callid string

Definition at line 33609 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_pvt *pvt = obj, *pvt2 = arg;

   return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static int dialog_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 19351 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and ast_cli_args::fd.

Referenced by sip_show_objects().

{
   struct sip_pvt *pvt = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   struct ast_cli_args *a = (struct ast_cli_args *) arg;
   
   ast_cli(a->fd, "name: %s\ntype: dialog\nobjflags: %d\nrefcount: %d\n\n",
      pvt->callid, 0, refc);
   return 0;
}

static int dialog_find_multiple	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Note:

Same as dialog_cmp_cb, except without the CMP_STOP on match

Definition at line 33599 of file chan_sip.c.

References CMP_MATCH.

Referenced by find_call().

{
   struct sip_pvt *pvt = obj, *pvt2 = arg;

   return !strcasecmp(pvt->callid, pvt2->callid) ? CMP_MATCH : 0;
}

static int dialog_hash_cb	(	const void *	obj,
		const int	flags
	)		[static]

Note:

The only member of the dialog used here callid string

Definition at line 33589 of file chan_sip.c.

References ast_str_case_hash().

Referenced by load_module().

{
   const struct sip_pvt *pvt = obj;

   return ast_str_case_hash(pvt->callid);
}

static int dialog_initialize_dtls_srtp	(	const struct sip_pvt *	dialog,
		struct ast_rtp_instance *	rtp,
		struct sip_srtp **	srtp
	)		[static]

Initialize DTLS-SRTP support on an RTP instance.

Definition at line 5839 of file chan_sip.c.

References ast_log(), ast_rtp_engine_srtp_is_registered(), ast_rtp_instance_get_dtls(), LOG_ERROR, ast_rtp_engine_dtls::set_configuration, and sip_srtp_alloc().

Referenced by dialog_initialize_rtp().

{
   struct ast_rtp_engine_dtls *dtls;

   if (!dialog->dtls_cfg.enabled) {
      return 0;
   }

   if (!ast_rtp_engine_srtp_is_registered()) {
      ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
      return -1;
   }

   if (!(dtls = ast_rtp_instance_get_dtls(rtp))) {
      ast_log(LOG_ERROR, "No DTLS-SRTP support present on engine for RTP instance '%p', was it compiled with support for it?\n",
         rtp);
      return -1;
   }

   if (dtls->set_configuration(rtp, &dialog->dtls_cfg)) {
      ast_log(LOG_ERROR, "Attempted to set an invalid DTLS-SRTP configuration on RTP instance '%p'\n",
         rtp);
      return -1;
   }

   if (!(*srtp = sip_srtp_alloc())) {
      ast_log(LOG_ERROR, "Failed to create required SRTP structure on RTP instance '%p'\n",
         rtp);
      return -1;
   }

   return 0;
}

static int dialog_initialize_rtp

(

struct sip_pvt *

dialog

)

[static]

Initialize RTP portion of a dialog.

Returns:

-1 on failure, 0 on success

Definition at line 5876 of file chan_sip.c.

References ast_format_cap_has_type(), AST_FORMAT_TYPE_VIDEO, ast_rtp_instance_get_ice(), ast_rtp_instance_new(), ast_rtp_instance_set_hold_timeout(), ast_rtp_instance_set_keepalive(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_qos(), ast_rtp_instance_set_timeout(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, AST_RTP_PROPERTY_RTCP, ast_sockaddr_copy(), ast_test_flag, bindaddr, dialog_initialize_dtls_srtp(), do_setnat(), cfsip_methods::need_rtp, sip_methods, and ast_rtp_engine_ice::stop.

Referenced by check_peer_ok(), check_user_full(), create_addr(), and create_addr_from_peer().

{
   struct ast_sockaddr bindaddr_tmp;
   struct ast_rtp_engine_ice *ice;

   if (!sip_methods[dialog->method].need_rtp) {
      return 0;
   }

   ast_sockaddr_copy(&bindaddr_tmp, &bindaddr);
   if (!(dialog->rtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
      return -1;
   }

   if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->rtp))) {
      ice->stop(dialog->rtp);
   }

   if (dialog_initialize_dtls_srtp(dialog, dialog->rtp, &dialog->srtp)) {
      return -1;
   }

   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS) ||
         (ast_test_flag(&dialog->flags[1], SIP_PAGE2_VIDEOSUPPORT) && (ast_format_cap_has_type(dialog->caps, AST_FORMAT_TYPE_VIDEO)))) {
      if (!(dialog->vrtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
         return -1;
      }

      if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->vrtp))) {
         ice->stop(dialog->vrtp);
      }

      if (dialog_initialize_dtls_srtp(dialog, dialog->vrtp, &dialog->vsrtp)) {
         return -1;
      }

      ast_rtp_instance_set_timeout(dialog->vrtp, dialog->rtptimeout);
      ast_rtp_instance_set_hold_timeout(dialog->vrtp, dialog->rtpholdtimeout);
      ast_rtp_instance_set_keepalive(dialog->vrtp, dialog->rtpkeepalive);

      ast_rtp_instance_set_prop(dialog->vrtp, AST_RTP_PROPERTY_RTCP, 1);
      ast_rtp_instance_set_qos(dialog->vrtp, global_tos_video, global_cos_video, "SIP VIDEO");
   }

   if (ast_test_flag(&dialog->flags[1], SIP_PAGE2_TEXTSUPPORT)) {
      if (!(dialog->trtp = ast_rtp_instance_new(dialog->engine, sched, &bindaddr_tmp, NULL))) {
         return -1;
      }

      if (!ast_test_flag(&dialog->flags[2], SIP_PAGE3_ICE_SUPPORT) && (ice = ast_rtp_instance_get_ice(dialog->trtp))) {
         ice->stop(dialog->trtp);
      }

      if (dialog_initialize_dtls_srtp(dialog, dialog->trtp, &dialog->tsrtp)) {
         return -1;
      }

      /* Do not timeout text as its not constant*/
      ast_rtp_instance_set_keepalive(dialog->trtp, dialog->rtpkeepalive);

      ast_rtp_instance_set_prop(dialog->trtp, AST_RTP_PROPERTY_RTCP, 1);
   }

   ast_rtp_instance_set_timeout(dialog->rtp, dialog->rtptimeout);
   ast_rtp_instance_set_hold_timeout(dialog->rtp, dialog->rtpholdtimeout);
   ast_rtp_instance_set_keepalive(dialog->rtp, dialog->rtpkeepalive);

   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_RTCP, 1);
   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
   ast_rtp_instance_set_prop(dialog->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));

   ast_rtp_instance_set_qos(dialog->rtp, global_tos_audio, global_cos_audio, "SIP RTP");

   do_setnat(dialog);

   return 0;
}

static int dialog_needdestroy	(	void *	dialogobj,
		void *	arg,
		int	flags
	)		[static]

Match dialogs that need to be destroyed.

This is used with ao2_callback to unlink/delete all dialogs that are marked needdestroy.

Todo:

Re-work this to improve efficiency. Currently, this function is called on _every_ dialog after processing _every_ incoming SIP/UDP packet, or potentially even more often when the scheduler has entries to run.

Definition at line 19534 of file chan_sip.c.

References ast_debug, ast_rtp_instance_get_bridged(), dialog_unlink_all(), sip_methods, sip_pvt_trylock, sip_pvt_unlock, and cfsip_methods::text.

Referenced by do_monitor().

{
   struct sip_pvt *dialog = dialogobj;

   if (sip_pvt_trylock(dialog)) {
      /* Don't block the monitor thread.  This function is called often enough
       * that we can wait for the next time around. */
      return 0;
   }

   /* If we have sessions that needs to be destroyed, do it now */
   /* Check if we have outstanding requests not responsed to or an active call
      - if that's the case, wait with destruction */
   if (dialog->needdestroy && !dialog->packets && !dialog->owner) {
      /* We absolutely cannot destroy the rtp struct while a bridge is active or we WILL crash */
      if (dialog->rtp && ast_rtp_instance_get_bridged(dialog->rtp)) {
         ast_debug(2, "Bridge still active.  Delaying destruction of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }

      if (dialog->vrtp && ast_rtp_instance_get_bridged(dialog->vrtp)) {
         ast_debug(2, "Bridge still active.  Delaying destroy of SIP dialog '%s' Method: %s\n", dialog->callid, sip_methods[dialog->method].text);
         sip_pvt_unlock(dialog);
         return 0;
      }

      sip_pvt_unlock(dialog);
      /* no, the unlink should handle this: dialog_unref(dialog, "needdestroy: one more refcount decrement to allow dialog to be destroyed"); */
      /* the CMP_MATCH will unlink this dialog from the dialog hash table */
      dialog_unlink_all(dialog);
      return 0; /* the unlink_all should unlink this from the table, so.... no need to return a match */
   }

   sip_pvt_unlock(dialog);

   return 0;
}

struct sip_pvt* dialog_ref_debug	(	struct sip_pvt *	p,
		const char *	tag,
		char *	file,
		int	line,
		const char *	func
	)		[read]

Definition at line 2327 of file chan_sip.c.

References __ao2_ref_debug(), ao2_ref, ast_log(), and LOG_ERROR.

{
   if (p)
#ifdef REF_DEBUG
      __ao2_ref_debug(p, 1, tag, file, line, func);
#else
      ao2_ref(p, 1);
#endif
   else
      ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
   return p;
}

void dialog_unlink_all

(

struct sip_pvt *

dialog

)

Unlink a dialog from the dialogs container, as well as any other places that it may be currently stored.

Note:

A reference to the dialog must be held before calling this function, and this function does not release that reference.

Definition at line 3392 of file chan_sip.c.

References ao2_t_unlink, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_tech_pvt_set(), ast_channel_unlock, ast_channel_unref, ast_debug, ast_extension_state_del(), ast_free, AST_SCHED_DEL, AST_SCHED_DEL_UNREF, cb_extensionstate(), registry_unref(), sip_pvt_lock_full(), sip_pvt_unlock, and stop_session_timer().

Referenced by __sip_autodestruct(), __sip_subscribe_mwi_do(), cleanup_all_regs(), dialog_needdestroy(), handle_request_subscribe(), manager_sipnotify(), sip_cli_notify(), sip_destroy_peer(), sip_msg_send(), sip_poke_noanswer(), sip_poke_peer(), sip_registry_destroy(), sip_request_call(), sip_send_mwi_to_peer(), transmit_publish(), transmit_register(), and unload_module().

{
   struct sip_pkt *cp;
   struct ast_channel *owner;

   dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");

   ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");
   ao2_t_unlink(dialogs_needdestroy, dialog, "unlinking dialog_needdestroy via ao2_unlink");
   ao2_t_unlink(dialogs_rtpcheck, dialog, "unlinking dialog_rtpcheck via ao2_unlink");

   /* Unlink us from the owner (channel) if we have one */
   owner = sip_pvt_lock_full(dialog);
   if (owner) {
      ast_debug(1, "Detaching from channel %s\n", ast_channel_name(owner));
      ast_channel_tech_pvt_set(owner, dialog_unref(ast_channel_tech_pvt(owner), "resetting channel dialog ptr in unlink_all"));
      ast_channel_unlock(owner);
      ast_channel_unref(owner);
      dialog->owner = NULL;
   }
   sip_pvt_unlock(dialog);

   if (dialog->registry) {
      if (dialog->registry->call == dialog) {
         dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
      }
      dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
   }
   if (dialog->stateid != -1) {
      ast_extension_state_del(dialog->stateid, cb_extensionstate);
      dialog->stateid = -1;
   }
   /* Remove link from peer to subscription of MWI */
   if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog) {
      dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
   }
   if (dialog->relatedpeer && dialog->relatedpeer->call == dialog) {
      dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");
   }

   /* remove all current packets in this dialog */
   while((cp = dialog->packets)) {
      dialog->packets = dialog->packets->next;
      AST_SCHED_DEL(sched, cp->retransid);
      dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
      if (cp->data) {
         ast_free(cp->data);
      }
      ast_free(cp);
   }

   AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));

   AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
   
   if (dialog->autokillid > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));
   }

   if (dialog->request_queue_sched_id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id, dialog_unref(dialog, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
   }

   AST_SCHED_DEL_UNREF(sched, dialog->provisional_keepalive_sched_id, dialog_unref(dialog, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));

   if (dialog->t38id > -1) {
      AST_SCHED_DEL_UNREF(sched, dialog->t38id, dialog_unref(dialog, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
   }

   if (dialog->stimer) {
      stop_session_timer(dialog);
   }

   dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
}

struct sip_pvt* dialog_unref_debug	(	struct sip_pvt *	p,
		const char *	tag,
		char *	file,
		int	line,
		const char *	func
	)		[read]

Definition at line 2340 of file chan_sip.c.

References __ao2_ref_debug(), and ao2_ref.

{
   if (p)
#ifdef REF_DEBUG
      __ao2_ref_debug(p, -1, tag, file, line, func);
#else
      ao2_ref(p, -1);
#endif
   return NULL;
}

static void disable_dsp_detect

(

struct sip_pvt *

p

)

[static]

Definition at line 4840 of file chan_sip.c.

References ast_dsp_free().

Referenced by sip_dtmfmode(), sip_hangup(), and sip_setoption().

{
   if (p->dsp) {
      ast_dsp_free(p->dsp);
      p->dsp = NULL;
   }
}

static void display_nat_warning	(	const char *	cat,
		int	reason,
		struct ast_flags *	flags
	)		[static]

Definition at line 31377 of file chan_sip.c.

References AST_CLI_YESNO, ast_log(), ast_test_flag, CHANNEL_MODULE_LOAD, and LOG_WARNING.

Referenced by reload_config().

                                                                                      {
   int global_nat, specific_nat;

   if (reason == CHANNEL_MODULE_LOAD && (specific_nat = ast_test_flag(&flags[0], SIP_NAT_FORCE_RPORT)) != (global_nat = ast_test_flag(&global_flags[0], SIP_NAT_FORCE_RPORT))) {
      ast_log(LOG_WARNING, "!!! PLEASE NOTE: Setting 'nat' for a peer/user that differs from the  global setting can make\n");
      ast_log(LOG_WARNING, "!!! the name of that peer/user discoverable by an attacker. Replies for non-existent peers/users\n");
      ast_log(LOG_WARNING, "!!! will be sent to a different port than replies for an existing peer/user. If at all possible,\n");
      ast_log(LOG_WARNING, "!!! use the global 'nat' setting and do not set 'nat' per peer/user.\n");
      ast_log(LOG_WARNING, "!!! (config category='%s' global force_rport='%s' peer/user force_rport='%s')\n", cat, AST_CLI_YESNO(global_nat), AST_CLI_YESNO(specific_nat));
   }
}

static int do_magic_pickup	(	struct ast_channel *	channel,
		const char *	extension,
		const char *	context
	)		[static]

Note:

No channel or pvt locks should be held while calling this function.

Definition at line 24993 of file chan_sip.c.

References ast_debug, ast_log(), AST_MAX_CONTEXT, AST_MAX_EXTENSION, ast_str_alloca, ast_str_buffer(), ast_str_set(), LOG_ERROR, pbx_exec(), pbx_findapp(), sip_cfg, and str.

Referenced by handle_request_invite().

{
   struct ast_str *str = ast_str_alloca(AST_MAX_EXTENSION + AST_MAX_CONTEXT + 2);
   struct ast_app *pickup = pbx_findapp("Pickup");

   if (!pickup) {
      ast_log(LOG_ERROR, "Unable to perform pickup: Application 'Pickup' not loaded (app_directed_pickup.so).\n");
      return -1;
   }

   ast_str_set(&str, 0, "%s@%s", extension, sip_cfg.notifycid == IGNORE_CONTEXT ? "PICKUPMARK" : context);

   ast_debug(2, "About to call Pickup(%s)\n", ast_str_buffer(str));

   /* There is no point in capturing the return value since pickup_exec
      doesn't return anything meaningful unless the passed data is an empty
      string (which in our case it will not be) */
   pbx_exec(channel, pickup, ast_str_buffer(str));

   return 0;
}

static int do_message_auth	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23709 of file chan_sip.c.

References append_history, ast_debug, ast_log(), ast_sockaddr_stringify(), LOG_NOTICE, reply_digest(), sip_auth_headers(), and transmit_message().

Referenced by handle_response_message().

{
   char *header;
   char *respheader;
   char digest[1024];

   if (p->options) {
      p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
   }

   if (p->authtries == MAX_AUTHTRIES) {
      ast_log(LOG_NOTICE, "Failed to authenticate MESSAGE with host '%s'\n",
         ast_sockaddr_stringify(&p->sa));
      return -1;
   }

   ++p->authtries;
   sip_auth_headers((resp == 401 ? WWW_AUTH : PROXY_AUTH), &header, &respheader);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, SIP_MESSAGE, digest, sizeof(digest))) {
      /* There's nothing to use for authentication */
      ast_debug(1, "Nothing to use for MESSAGE authentication\n");
      return -1;
   }

   if (p->do_history) {
      append_history(p, "MessageAuth", "Try: %d", p->authtries);
   }

   transmit_message(p, 0, 1);
   return 0;
}

static void * do_monitor

(

void *

data

)

[static]

The SIP monitoring thread.

Note:

This thread monitors all the SIP sessions and peers that needs notification of mwi (and thus do not have a separate thread) indefinitely

Definition at line 29053 of file chan_sip.c.

References ao2_t_callback, ast_debug, ast_io_add(), ast_io_change(), AST_IO_IN, ast_io_remove(), ast_io_wait(), ast_mutex_lock, ast_mutex_unlock, ast_sched_runq(), ast_sched_wait(), ast_verb, dialog_checkrtp_cb(), dialog_needdestroy(), FALSE, monlock, OBJ_MULTIPLE, OBJ_NODATA, OBJ_UNLINK, sip_do_reload(), sip_reload_lock, sip_reloading, and sipsock_read().

Referenced by restart_monitor().

{
   int res;
   time_t t;
   int reloading;

   /* Add an I/O event to our SIP UDP socket */
   if (sipsock > -1) {
      sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
   }

   /* From here on out, we die whenever asked */
   for(;;) {
      /* Check for a reload request */
      ast_mutex_lock(&sip_reload_lock);
      reloading = sip_reloading;
      sip_reloading = FALSE;
      ast_mutex_unlock(&sip_reload_lock);
      if (reloading) {
         ast_verb(1, "Reloading SIP\n");
         sip_do_reload(sip_reloadreason);

         /* Change the I/O fd of our UDP socket */
         if (sipsock > -1) {
            if (sipsock_read_id) {
               sipsock_read_id = ast_io_change(io, sipsock_read_id, sipsock, NULL, 0, NULL);
            } else {
               sipsock_read_id = ast_io_add(io, sipsock, sipsock_read, AST_IO_IN, NULL);
            }
         } else if (sipsock_read_id) {
            ast_io_remove(io, sipsock_read_id);
            sipsock_read_id = NULL;
         }
      }

      /* Check for dialogs needing to be killed */
      t = time(NULL);

      /*
       * Check dialogs with rtp and rtptimeout.
       * All dialogs which have rtp are in dialogs_rtpcheck.
       */
      ao2_t_callback(dialogs_rtpcheck, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE,
         dialog_checkrtp_cb, &t,
         "callback to check rtptimeout and hangup calls if necessary");
      /*
       * Check dialogs marked to be destroyed.
       * All dialogs with needdestroy set are in dialogs_needdestroy.
       */
      ao2_t_callback(dialogs_needdestroy, OBJ_NODATA | OBJ_MULTIPLE, dialog_needdestroy,
         NULL, "callback to check dialogs which need to be destroyed");

      /* XXX TODO The scheduler usage in this module does not have sufficient
       * synchronization being done between running the scheduler and places
       * scheduling tasks.  As it is written, any scheduled item may not run
       * any sooner than about  1 second, regardless of whether a sooner time
       * was asked for. */

      pthread_testcancel();
      /* Wait for sched or io */
      res = ast_sched_wait(sched);
      if ((res < 0) || (res > 1000)) {
         res = 1000;
      }
      res = ast_io_wait(io, res);
      if (res > 20) {
         ast_debug(1, "chan_sip: ast_io_wait ran %d all at once\n", res);
      }
      ast_mutex_lock(&monlock);
      res = ast_sched_runq(sched);
      if (res >= 20) {
         ast_debug(1, "chan_sip: ast_sched_runq ran %d all at once\n", res);
      }
      ast_mutex_unlock(&monlock);
   }

   /* Never reached */
   return NULL;
}

static int do_proxy_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code,
		int	sipmethod,
		int	init
	)		[static]

Add authentication on outbound SIP packet.

Definition at line 21817 of file chan_sip.c.

References ast_calloc, ast_debug, reply_digest(), sip_auth_headers(), sip_methods, cfsip_methods::text, and transmit_invite().

Referenced by handle_response(), handle_response_invite(), handle_response_notify(), handle_response_publish(), handle_response_refer(), handle_response_subscribe(), and handle_response_update().

{
   char *header, *respheader;
   char digest[1024];

   if (!p->options && !(p->options = ast_calloc(1, sizeof(*p->options))))
      return -2;

   p->authtries++;
   sip_auth_headers(code, &header, &respheader);
   ast_debug(2, "Auth attempt %d on %s\n", p->authtries, sip_methods[sipmethod].text);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, sipmethod, digest, sizeof(digest) )) {
      /* No way to authenticate */
      return -1;
   }
   /* Now we have a reply digest */
   p->options->auth = digest;
   p->options->authheader = respheader;
   return transmit_invite(p, sipmethod, sipmethod == SIP_INVITE, init, NULL);
}

static int do_register_auth	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum sip_auth_type	code
	)		[static]

Authenticate for outbound registration.

Definition at line 21793 of file chan_sip.c.

References append_history, ast_verbose(), reply_digest(), sip_auth_headers(), sip_debug_test_pvt(), and transmit_register().

Referenced by handle_response_register().

{
   char *header, *respheader;
   char digest[1024];

   p->authtries++;
   sip_auth_headers(code, &header, &respheader);
   memset(digest, 0, sizeof(digest));
   if (reply_digest(p, req, header, SIP_REGISTER, digest, sizeof(digest))) {
      /* There's nothing to use for authentication */
      /* No digest challenge in request */
      if (sip_debug_test_pvt(p) && p->registry)
         ast_verbose("No authentication challenge, sending blank registration to domain/host name %s\n", p->registry->hostname);
         /* No old challenge */
      return -1;
   }
   if (p->do_history)
      append_history(p, "RegistryAuth", "Try: %d", p->authtries);
   if (sip_debug_test_pvt(p) && p->registry)
      ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
   return transmit_register(p->registry, SIP_REGISTER, digest, respheader);
}

static void do_setnat

(

struct sip_pvt *

p

)

[static]

Set nat mode on the various data sockets.

Definition at line 5729 of file chan_sip.c.

References ast_debug, ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_NAT, ast_test_flag, and ast_udptl_setnat().

Referenced by check_peer_ok(), dialog_initialize_rtp(), sip_alloc(), and sip_request_call().

{
   const char *mode;
   int natflags;

   natflags = ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);
   mode = natflags ? "On" : "Off";

   if (p->rtp) {
      ast_debug(1, "Setting NAT on RTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_NAT, natflags);
   }
   if (p->vrtp) {
      ast_debug(1, "Setting NAT on VRTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_NAT, natflags);
   }
   if (p->udptl) {
      ast_debug(1, "Setting NAT on UDPTL to %s\n", mode);
      ast_udptl_setnat(p->udptl, natflags);
   }
   if (p->trtp) {
      ast_debug(1, "Setting NAT on TRTP to %s\n", mode);
      ast_rtp_instance_set_prop(p->trtp, AST_RTP_PROPERTY_NAT, natflags);
   }
}

static const char * domain_mode_to_text

(

const enum domain_mode

mode

)

[static]

Print domain mode to cli.

Definition at line 19733 of file chan_sip.c.

Referenced by sip_show_domains().

{
   switch (mode) {
   case SIP_DOMAIN_AUTO:
      return "[Automatic]";
   case SIP_DOMAIN_CONFIG:
      return "[Configured]";
   }

   return "";
}

static const char * dtmfmode2str

(

int

mode

)

[static]

Convert DTMF mode to printable string.

Definition at line 19419 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), sip_show_channel(), and sip_show_settings().

{
   return map_x_s(dtmfstr, mode, "<error>");
}

static void enable_dsp_detect

(

struct sip_pvt *

p

)

[static]

Definition at line 4806 of file chan_sip.c.

References ast_dsp_new(), ast_dsp_set_digitmode(), ast_dsp_set_features(), AST_RTP_DTMF_MODE_INBAND, ast_rtp_instance_dtmf_mode_set(), ast_test_flag, DSP_DIGITMODE_DTMF, DSP_DIGITMODE_RELAXDTMF, DSP_FEATURE_DIGIT_DETECT, and DSP_FEATURE_FAX_DETECT.

Referenced by sip_dtmfmode(), sip_new(), and sip_setoption().

{
   int features = 0;

   if (p->dsp) {
      return;
   }

   if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
       (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      if (p->rtp) {
         ast_rtp_instance_dtmf_mode_set(p->rtp, AST_RTP_DTMF_MODE_INBAND);
      }
      features |= DSP_FEATURE_DIGIT_DETECT;
   }

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
      features |= DSP_FEATURE_FAX_DETECT;
   }

   if (!features) {
      return;
   }

   if (!(p->dsp = ast_dsp_new())) {
      return;
   }

   ast_dsp_set_features(p->dsp, features);
   if (global_relaxdtmf) {
      ast_dsp_set_digitmode(p->dsp, DSP_DIGITMODE_DTMF | DSP_DIGITMODE_RELAXDTMF);
   }
}

static int esc_cmp_fn	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1050 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by initialize_escs().

{
   struct sip_esc_entry *entry1 = obj;
   struct sip_esc_entry *entry2 = arg;

   return (!strcmp(entry1->entity_tag, entry2->entity_tag)) ? (CMP_MATCH | CMP_STOP) : 0;
}

static void esc_entry_destructor

(

void *

obj

)

[static]

Definition at line 1036 of file chan_sip.c.

References AST_SCHED_DEL.

Referenced by create_esc_entry().

{
   struct sip_esc_entry *esc_entry = obj;
   if (esc_entry->sched_id > -1) {
      AST_SCHED_DEL(sched, esc_entry->sched_id);
   }
}

static int esc_hash_fn	(	const void *	obj,
		const int	flags
	)		[static]

Definition at line 1044 of file chan_sip.c.

References ast_str_hash().

Referenced by initialize_escs().

{
   const struct sip_esc_entry *entry = obj;
   return ast_str_hash(entry->entity_tag);
}

static int expire_register

(

const void *

data

)

[static]

Expire registration of SIP peer.

Definition at line 15750 of file chan_sip.c.

References ao2_ref, ao2_t_unlink, ast_debug, AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), ast_sockaddr_isnull(), ast_test_flag, ast_websocket_unref(), destroy_association(), EVENT_FLAG_SYSTEM, FALSE, manager_event, register_peer_exten(), set_socket_transport(), sip_unref_peer(), and unlink_peer_from_tables().

Referenced by parse_register_contact(), realtime_peer(), reg_source_db(), sip_show_sched(), and sip_unregister().

{
   struct sip_peer *peer = (struct sip_peer *)data;

   if (!peer) {      /* Hmmm. We have no peer. Weird. */
      return 0;
   }

   peer->expire = -1;
   peer->portinuri = 0;

   destroy_association(peer); /* remove registration data from storage */
   set_socket_transport(&peer->socket, peer->default_outbound_transport);

   if (peer->socket.tcptls_session) {
      ao2_ref(peer->socket.tcptls_session, -1);
      peer->socket.tcptls_session = NULL;
   } else if (peer->socket.ws_session) {
      ast_websocket_unref(peer->socket.ws_session);
      peer->socket.ws_session = NULL;
   }

   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unregistered\r\nCause: Expired\r\n", peer->name);
   register_peer_exten(peer, FALSE);   /* Remove regexten */
   ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);

   /* Do we need to release this peer from memory?
      Only for realtime peers and autocreated peers
   */
   if (peer->is_realtime) {
      ast_debug(3, "-REALTIME- peer expired registration. Name: %s. Realtime peer objects now %d\n", peer->name, rpeerobjs);
   }

   if (peer->selfdestruct ||
       ast_test_flag(&peer->flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
      unlink_peer_from_tables(peer);
   } else if (!ast_sockaddr_isnull(&peer->addr)) {
      /* If we aren't self-destructing a temp_peer, we still need to unlink the peer
       * from the peers_by_ip table, otherwise we end up with multiple copies hanging
       * around each time a registration expires and the peer re-registers. */
      ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
   }

   /* Only clear the addr after we check for destruction.  The addr must remain
    * in order to unlink from the peers_by_ip container correctly */
   memset(&peer->addr, 0, sizeof(peer->addr));

   sip_unref_peer(peer, "removing peer ref for expire_register");

   return 0;
}

static int extensionstate_update	(	const char *	context,
		const char *	exten,
		struct state_notify_data *	data,
		struct sip_pvt *	p,
		int	force
	)		[static]

Callback for the devicestate notification (SUBSCRIBE) support subsystem.

Note:

If you add an "hint" priority to the extension in the dial plan, you will get notifications on device state changes

Definition at line 16652 of file chan_sip.c.

References ao2_cleanup, ao2_ref, append_history, ast_channel_creationtime(), AST_EXTENSION_DEACTIVATED, AST_EXTENSION_REMOVED, AST_EXTENSION_RINGING, ast_extension_state2str(), ast_set_flag, ast_string_field_set, ast_test_flag, ast_tvcmp(), ast_verb, state_notify_data::device_state_info, FALSE, find_ringing_channel(), NONE, state_notify_data::presence_message, state_notify_data::presence_state, state_notify_data::presence_subtype, S_OR, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), state_notify_data::state, and transmit_state_notify().

Referenced by cb_extensionstate(), handle_request_subscribe(), and handle_response_notify().

{
   sip_pvt_lock(p);

   switch (data->state) {
   case AST_EXTENSION_DEACTIVATED:  /* Retry after a while */
   case AST_EXTENSION_REMOVED:   /* Extension is gone */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);  /* Delete subscription in 32 secs */
      ast_verb(2, "Extension state: Watcher for hint %s %s. Notify User %s\n", exten, data->state == AST_EXTENSION_DEACTIVATED ? "deactivated" : "removed", p->username);
      p->subscribed = NONE;
      append_history(p, "Subscribestatus", "%s", data->state == AST_EXTENSION_REMOVED ? "HintRemoved" : "Deactivated");
      break;
   default: /* Tell user */
      if (force) {
         /* we must skip the next two checks for a queued state change or resubscribe */
      } else if ((p->laststate == data->state && (~data->state & AST_EXTENSION_RINGING)) &&
            (p->last_presence_state == data->presence_state &&
               !strcmp(p->last_presence_subtype, data->presence_subtype) &&
               !strcmp(p->last_presence_message, data->presence_message))) {
         /* don't notify unchanged state or unchanged early-state causing parties again */
         sip_pvt_unlock(p);
         return 0;
      } else if (data->state & AST_EXTENSION_RINGING) {
         /* check if another channel than last time is ringing now to be notified */
         struct ast_channel *ringing = find_ringing_channel(data->device_state_info, p);
         if (ringing) {
            if (!ast_tvcmp(ast_channel_creationtime(ringing), p->last_ringing_channel_time)) {
               /* we assume here that no two channels have the exact same creation time */
               ao2_ref(ringing, -1);
               sip_pvt_unlock(p);
               return 0;
            } else {
               p->last_ringing_channel_time = ast_channel_creationtime(ringing);
               ao2_ref(ringing, -1);
            }
         }
         /* If no ringing channel was found, it doesn't necessarily indicate anything bad.
          * Likely, a device state change occurred for a custom device state, which does not
          * correspond to any channel. In such a case, just go ahead and pass the notification
          * along.
          */
      }
      /* ref before unref because the new could be the same as the old one. Don't risk destruction! */
      if (data->device_state_info) {
         ao2_ref(data->device_state_info, 1);
      }
      ao2_cleanup(p->last_device_state_info);
      p->last_device_state_info = data->device_state_info;
      p->laststate = data->state;
      p->last_presence_state = data->presence_state;
      ast_string_field_set(p, last_presence_subtype, S_OR(data->presence_subtype, ""));
      ast_string_field_set(p, last_presence_message, S_OR(data->presence_message, ""));
      break;
   }
   if (p->subscribed != NONE) {  /* Only send state NOTIFY if we know the format */
      if (!p->pendinginvite) {
         transmit_state_notify(p, data, 1, FALSE);
         if (p->last_device_state_info) {
            /* We don't need the saved ref anymore, don't keep channels ref'd. */
            ao2_ref(p->last_device_state_info, -1);
            p->last_device_state_info = NULL;
         }
      } else {
         /* We already have a NOTIFY sent that is not answered. Queue the state up.
            if many state changes happen meanwhile, we will only send a notification of the last one */
         ast_set_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
      }
   }

   if (!force) {
      ast_verb(2, "Extension Changed %s[%s] new state %s for Notify User %s %s\n", exten, context, ast_extension_state2str(data->state), p->username,
            ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE) ? "(queued)" : "");
   }

   sip_pvt_unlock(p);

   return 0;
}

static void extract_host_from_hostport

(

char **

hostport

)

[static]

Terminate a host:port at the ':'.

Parameters:

hostport

The address of the hostport string

Note:

In the case of a bracket-enclosed IPv6 address, the hostport variable will contain the non-bracketed host as a result of calling this function.

Definition at line 16873 of file chan_sip.c.

References ast_sockaddr_split_hostport(), and PARSE_PORT_IGNORE.

Referenced by check_user_full(), get_destination(), register_verify(), and sip_msg_send().

{
   char *dont_care;
   ast_sockaddr_split_hostport(*hostport, hostport, &dont_care, PARSE_PORT_IGNORE);
}

static void extract_uri	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Check Contact: URI of SIP message.

Definition at line 13777 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, ast_strlen_zero(), get_in_brackets(), remove_uri_parameters(), and sip_get_header().

Referenced by handle_incoming(), and handle_request_invite().

{
   char stripped[SIPBUFSIZE];
   char *c;

   ast_copy_string(stripped, sip_get_header(req, "Contact"), sizeof(stripped));
   c = get_in_brackets(stripped);
   /* Cut the URI at the at sign after the @, not in the username part */
   c = remove_uri_parameters(c);
   if (!ast_strlen_zero(c)) {
      ast_string_field_set(p, uri, c);
   }

}

static const char* faxec2str

(

int

faxec

)

[static]

Definition at line 19983 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), and sip_show_settings().

{
   return map_x_s(faxecmodes, faxec, "Unknown");
}

static int finalize_content

(

struct sip_request *

req

)

[static]

Add 'Content-Length' header and content to SIP message.

Definition at line 11431 of file chan_sip.c.

References add_header(), ast_log(), ast_str_append(), ast_str_buffer(), ast_str_strlen(), and LOG_WARNING.

Referenced by send_request(), and send_response().

{
   char clen[10];

   if (req->lines) {
      ast_log(LOG_WARNING, "finalize_content() called on a message that has already been finalized\n");
      return -1;
   }

   snprintf(clen, sizeof(clen), "%zd", ast_str_strlen(req->content));
   add_header(req, "Content-Length", clen);

   if (ast_str_strlen(req->content)) {
      ast_str_append(&req->data, 0, "\r\n%s", ast_str_buffer(req->content));
   }
   req->lines = ast_str_strlen(req->content) ? 1 : 0;
   return 0;
}

static const char * find_alias	(	const char *	name,
		const char *	_default
	)		[static]

Find compressed SIP alias.

Definition at line 8228 of file chan_sip.c.

References ARRAY_LEN.

Referenced by __get_header(), and add_header().

{
   int x;

   for (x = 0; x < ARRAY_LEN(aliases); x++) {
      if (!strcasecmp(aliases[x].fullname, name))
         return aliases[x].shortname;
   }

   return _default;
}

static int find_by_callid_helper	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1790 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and ast_cc_agent::private_data.

Referenced by find_sip_cc_agent_by_original_callid().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   struct sip_pvt *call_pvt = arg;

   return !strcmp(agent_pvt->original_callid, call_pvt->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static int find_by_name	(	void *	obj,
		void *	arg,
		void *	data,
		int	flags
	)		[static]

Definition at line 5628 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, and match().

Referenced by sip_find_peer_full().

{
   struct sip_peer *search = obj, *match = arg;
   int *which_objects = data;

   /* Usernames in SIP uri's are case sensitive. Domains are not */
   if (strcmp(search->name, match->name)) {
      return 0;
   }

   switch (*which_objects) {
   case FINDUSERS:
      if (!(search->type & SIP_TYPE_USER)) {
         return 0;
      }
      break;
   case FINDPEERS:
      if (!(search->type & SIP_TYPE_PEER)) {
         return 0;
      }
      break;
   case FINDALLDEVICES:
      break;
   }

   return CMP_MATCH | CMP_STOP;
}

static int find_by_notify_uri_helper	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1760 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, ast_cc_agent::private_data, and sip_uri_cmp().

Referenced by find_sip_cc_agent_by_notify_uri().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   const char *uri = arg;

   return !sip_uri_cmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static int find_by_subscribe_uri_helper	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1775 of file chan_sip.c.

References CMP_MATCH, CMP_STOP, ast_cc_agent::private_data, and sip_uri_cmp().

Referenced by find_sip_cc_agent_by_subscribe_uri().

{
   struct ast_cc_agent *agent = obj;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   const char *uri = arg;

   return !sip_uri_cmp(agent_pvt->subscribe_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static struct sip_pvt * find_call	(	struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const int	intended_method
	)		[static, read]

find or create a dialog structure for an incoming SIP message. Connect incoming SIP message to current dialog or create new dialog structure Returns a reference to the sip_pvt object, remember to give it back once done. Called by handle_request_do

Definition at line 9145 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_next, ao2_t_callback, ao2_t_find, args, ast_callid_unref, ast_create_callid(), ast_debug, ast_skip_blanks(), ast_strlen_zero(), match_req_args::authentication_present, match_req_args::callid, dialog_find_multiple(), forked_invite_init(), free_via(), match_req_args::fromtag, gettag(), match_req_to_dialog(), match_req_args::method, OBJ_MULTIPLE, OBJ_POINTER, parse_via(), match_req_args::respid, match_req_args::ruri, match_req_args::seqno, sip_alloc(), sip_cfg, sip_get_header(), sip_methods, sip_pvt_lock, sip_pvt_unlock, SIP_REQ_FORKED, SIP_REQ_LOOP_DETECTED, SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, cfsip_methods::text, match_req_args::totag, transmit_response_using_temp(), match_req_args::viabranch, and match_req_args::viasentby.

Referenced by handle_request_do().

{
   char totag[128];
   char fromtag[128];
   const char *callid = sip_get_header(req, "Call-ID");
   const char *from = sip_get_header(req, "From");
   const char *to = sip_get_header(req, "To");
   const char *cseq = sip_get_header(req, "Cseq");
   struct sip_pvt *sip_pvt_ptr;
   uint32_t seqno;
   /* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
   /* sip_get_header always returns non-NULL so we must use ast_strlen_zero() */
   if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
         ast_strlen_zero(from) || ast_strlen_zero(cseq) ||
         (sscanf(cseq, "%30u", &seqno) != 1)) {

      /* RFC 3261 section 24.4.1.   Send a 400 Bad Request if the request is malformed. */
      if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
         transmit_response_using_temp(callid, addr, 1, intended_method,
                       req, "400 Bad Request");
      }
      return NULL;   /* Invalid packet */
   }

   if (sip_cfg.pedanticsipchecking) {
      /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy
         we need more to identify a branch - so we have to check branch, from
         and to tags to identify a call leg.
         For Asterisk to behave correctly, you need to turn on pedanticsipchecking
         in sip.conf
         */
      if (gettag(req, "To", totag, sizeof(totag)))
         req->has_to_tag = 1; /* Used in handle_request/response */
      gettag(req, "From", fromtag, sizeof(fromtag));

      ast_debug(5, "= Looking for  Call ID: %s (Checking %s) --From tag %s --To-tag %s  \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);

      /* All messages must always have From: tag */
      if (ast_strlen_zero(fromtag)) {
         ast_debug(5, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
      /* reject requests that must always have a To: tag */
      if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
         if (req->method != SIP_ACK) {
            transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
         }
         ast_debug(5, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
         return NULL;
      }
   }

   if (!sip_cfg.pedanticsipchecking) {
      struct sip_pvt tmp_dialog = {
         .callid = callid,
      };
      sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find in dialogs");
      if (sip_pvt_ptr) {  /* well, if we don't find it-- what IS in there? */
         /* Found the call */
         return sip_pvt_ptr;
      }
   } else { /* in pedantic mode! -- do the fancy search */
      struct sip_pvt tmp_dialog = {
         .callid = callid,
      };
      /* if a Outbound forked Request is detected, this pvt will point
       * to the dialog the Request is forking off of. */
      struct sip_pvt *fork_pvt = NULL;
      struct match_req_args args = { 0, };
      int found;
      struct ao2_iterator *iterator = ao2_t_callback(dialogs,
         OBJ_POINTER | OBJ_MULTIPLE,
         dialog_find_multiple,
         &tmp_dialog,
         "pedantic ao2_find in dialogs");
      struct sip_via *via = NULL;

      args.method = req->method;
      args.callid = NULL; /* we already matched this. */
      args.totag = totag;
      args.fromtag = fromtag;
      args.seqno = seqno;
      /* get via header information. */
      args.ruri = REQ_OFFSET_TO_STR(req, rlpart2);
      via = parse_via(sip_get_header(req, "Via"));
      if (via) {
         args.viasentby = via->sent_by;
         args.viabranch = via->branch;
      }
      /* determine if this is a Request with authentication credentials. */
      if (!ast_strlen_zero(sip_get_header(req, "Authorization")) ||
         !ast_strlen_zero(sip_get_header(req, "Proxy-Authorization"))) {
         args.authentication_present = 1;
      }
      /* if it is a response, get the response code */
      if (req->method == SIP_RESPONSE) {
         const char* e = ast_skip_blanks(REQ_OFFSET_TO_STR(req, rlpart2));
         int respid;
         if (!ast_strlen_zero(e) && (sscanf(e, "%30d", &respid) == 1)) {
            args.respid = respid;
         }
      }

      /* Iterate a list of dialogs already matched by Call-id */
      while (iterator && (sip_pvt_ptr = ao2_iterator_next(iterator))) {
         sip_pvt_lock(sip_pvt_ptr);
         found = match_req_to_dialog(sip_pvt_ptr, &args);
         sip_pvt_unlock(sip_pvt_ptr);

         switch (found) {
         case SIP_REQ_MATCH:
            ao2_iterator_destroy(iterator);
            dialog_unref(fork_pvt, "unref fork_pvt");
            free_via(via);
            return sip_pvt_ptr; /* return pvt with ref */
         case SIP_REQ_LOOP_DETECTED:
            /* This is likely a forked Request that somehow resulted in us receiving multiple parts of the fork.
            * RFC 3261 section 8.2.2.2, Indicate that we want to merge requests by sending a 482 response. */
            transmit_response_using_temp(callid, addr, 1, intended_method, req, "482 (Loop Detected)");
            dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search.");
            ao2_iterator_destroy(iterator);
            dialog_unref(fork_pvt, "unref fork_pvt");
            free_via(via);
            return NULL;
         case SIP_REQ_FORKED:
            dialog_unref(fork_pvt, "throwing way pvt to fork off of.");
            fork_pvt = dialog_ref(sip_pvt_ptr, "this pvt has a forked request, save this off to copy information into new dialog\n");
            /* fall through */
         case SIP_REQ_NOT_MATCH:
         default:
            dialog_unref(sip_pvt_ptr, "pvt did not match incoming SIP msg, unref from search");
            break;
         }
      }
      if (iterator) {
         ao2_iterator_destroy(iterator);
      }

      /* Handle any possible forked requests. This must be done only after transaction matching is complete. */
      if (fork_pvt) {
         /* XXX right now we only support handling forked INVITE Requests. Any other
          * forked request type must be added here. */
         if (fork_pvt->method == SIP_INVITE) {
            forked_invite_init(req, args.totag, fork_pvt, addr);
            dialog_unref(fork_pvt, "throwing way old forked pvt");
            free_via(via);
            return NULL;
         }
         fork_pvt = dialog_unref(fork_pvt, "throwing way pvt to fork off of");
      }

      free_via(via);
   } /* end of pedantic mode Request/Reponse to Dialog matching */

   /* See if the method is capable of creating a dialog */
   if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
      struct sip_pvt *p = NULL;
      struct ast_callid *logger_callid = NULL;

      if (intended_method == SIP_INVITE) {
         logger_callid = ast_create_callid();
      }

      /* Ok, time to create a new SIP dialog object, a pvt */
      if (!(p = sip_alloc(callid, addr, 1, intended_method, req, logger_callid)))  {
         /* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
            getting a dialog from sip_alloc.

            Without a dialog we can't retransmit and handle ACKs and all that, but at least
            send an error message.

            Sorry, we apologize for the inconvienience
         */
         transmit_response_using_temp(callid, addr, 1, intended_method, req, "500 Server internal error");
         ast_debug(4, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
      }
      /* If we created an ast_callid for an invite, we need to free it now. */
      if (logger_callid) {
         ast_callid_unref(logger_callid);
      }
      return p; /* can be NULL */
   } else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
      /* A method we do not support, let's take it on the volley */
      transmit_response_using_temp(callid, addr, 1, intended_method, req, "501 Method Not Implemented");
      ast_debug(2, "Got a request with unsupported SIP method.\n");
   } else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
      /* This is a request outside of a dialog that we don't know about */
      transmit_response_using_temp(callid, addr, 1, intended_method, req, "481 Call leg/transaction does not exist");
      ast_debug(2, "That's odd...  Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>");
   }
   /* We do not respond to responses for dialogs that we don't know about, we just drop
      the session quickly */
   if (intended_method == SIP_RESPONSE)
      ast_debug(2, "That's odd...  Got a response on a call we don't know about. Callid %s\n", callid ? callid : "<unknown>");

   return NULL;
}

const char* find_closing_quote	(	const char *	start,
		const char *	lim
	)

Locate closing quote in a string, skipping escaped quotes. optionally with a limit on the search. start must be past the first quote.

Definition at line 4984 of file chan_sip.c.

Referenced by get_comma(), get_in_brackets_const(), get_in_brackets_full(), and parse_moved_contact().

{
   char last_char = '\0';
   const char *s;
   for (s = start; *s && s != lim; last_char = *s++) {
      if (*s == '"' && last_char != '\\')
         break;
   }
   return s;
}

static const char* find_full_alias	(	const char *	name,
		const char *	_default
	)		[static]

Find full SIP alias.

Definition at line 8241 of file chan_sip.c.

References ARRAY_LEN.

Referenced by set_message_vars_from_req().

{
   int x;

   if (strlen(name) == 1) {
      /* We have a short header name to convert. */
      for (x = 0; x < ARRAY_LEN(aliases); ++x) {
         if (!strcasecmp(aliases[x].shortname, name))
            return aliases[x].fullname;
      }
   }

   return _default;
}

static struct sip_auth * find_realm_authentication	(	struct sip_auth_container *	credentials,
		const char *	realm
	)		[static, read]

Definition at line 30435 of file chan_sip.c.

References AST_LIST_TRAVERSE.

Referenced by build_reply_digest().

{
   struct sip_auth *auth;

   if (credentials) {
      AST_LIST_TRAVERSE(&credentials->list, auth, node) {
         if (!strcasecmp(auth->realm, realm)) {
            break;
         }
      }
   } else {
      auth = NULL;
   }

   return auth;
}

static struct ast_channel* find_ringing_channel	(	struct ao2_container *	device_state_info,
		struct sip_pvt *	p
	)		[static, read]

Definition at line 14428 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_ref, ast_channel_creationtime(), ast_channel_lock, ast_channel_ref, ast_channel_unlock, AST_DEVICE_RINGING, AST_DEVICE_RINGINUSE, ast_tvcmp(), ast_tvzero(), ast_device_state_info::causing_channel, ast_device_state_info::device_state, and tv.

Referenced by extensionstate_update(), handle_request_subscribe(), and state_notify_build_xml().

{
   struct ao2_iterator citer;
   struct ast_device_state_info *device_state;
   struct ast_channel *c = NULL;
   struct timeval tv = {0,};

   /* iterate ringing devices and get the oldest of all causing channels */
   citer = ao2_iterator_init(device_state_info, 0);
   for (; (device_state = ao2_iterator_next(&citer)); ao2_ref(device_state, -1)) {
      if (!device_state->causing_channel || (device_state->device_state != AST_DEVICE_RINGING &&
          device_state->device_state != AST_DEVICE_RINGINUSE)) {
         continue;
      }
      ast_channel_lock(device_state->causing_channel);
      if (ast_tvzero(tv) || ast_tvcmp(ast_channel_creationtime(device_state->causing_channel), tv) < 0) {
         c = device_state->causing_channel;
         tv = ast_channel_creationtime(c);
      }
      ast_channel_unlock(device_state->causing_channel);
   }
   ao2_iterator_destroy(&citer);
   return c ? ast_channel_ref(c) : NULL;
}

static int find_sdp

(

struct sip_request *

req

)

[static]

Determine whether a SIP message contains an SDP in its body.

Parameters:

req	the SIP request to process

Returns:

1 if SDP found, 0 if not found

Also updates req->sdp_start and req->sdp_count to indicate where the SDP lives in the message body.

Definition at line 9698 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), FALSE, LOG_WARNING, sip_get_header(), and TRUE.

Referenced by handle_incoming(), handle_request_invite(), handle_response(), and handle_response_invite().

{
   const char *content_type;
   const char *content_length;
   const char *search;
   char *boundary;
   unsigned int x;
   int boundaryisquoted = FALSE;
   int found_application_sdp = FALSE;
   int found_end_of_headers = FALSE;

   content_length = sip_get_header(req, "Content-Length");

   if (!ast_strlen_zero(content_length)) {
      if (sscanf(content_length, "%30u", &x) != 1) {
         ast_log(LOG_WARNING, "Invalid Content-Length: %s\n", content_length);
         return 0;
      }

      /* Content-Length of zero means there can't possibly be an
         SDP here, even if the Content-Type says there is */
      if (x == 0)
         return 0;
   }

   content_type = sip_get_header(req, "Content-Type");

   /* if the body contains only SDP, this is easy */
   if (!strncasecmp(content_type, "application/sdp", 15)) {
      req->sdp_start = 0;
      req->sdp_count = req->lines;
      return req->lines ? 1 : 0;
   }

   /* if it's not multipart/mixed, there cannot be an SDP */
   if (strncasecmp(content_type, "multipart/mixed", 15))
      return 0;

   /* if there is no boundary marker, it's invalid */
   if ((search = strcasestr(content_type, ";boundary=")))
      search += 10;
   else if ((search = strcasestr(content_type, "; boundary=")))
      search += 11;
   else
      return 0;

   if (ast_strlen_zero(search))
      return 0;

   /* If the boundary is quoted with ", remove quote */
   if (*search == '\"')  {
      search++;
      boundaryisquoted = TRUE;
   }

   /* make a duplicate of the string, with two extra characters
      at the beginning */
   boundary = ast_strdupa(search - 2);
   boundary[0] = boundary[1] = '-';
   /* Remove final quote */
   if (boundaryisquoted)
      boundary[strlen(boundary) - 1] = '\0';

   /* search for the boundary marker, the empty line delimiting headers from
      sdp part and the end boundry if it exists */

   for (x = 0; x < (req->lines); x++) {
      const char *line = REQ_OFFSET_TO_STR(req, line[x]);
      if (!strncasecmp(line, boundary, strlen(boundary))){
         if (found_application_sdp && found_end_of_headers) {
            req->sdp_count = (x - 1) - req->sdp_start;
            return 1;
         }
         found_application_sdp = FALSE;
      }
      if (!strcasecmp(line, "Content-Type: application/sdp"))
         found_application_sdp = TRUE;
      
      if (ast_strlen_zero(line)) {
         if (found_application_sdp && !found_end_of_headers){
            req->sdp_start = x;
            found_end_of_headers = TRUE;
         }
      }
   }
   if (found_application_sdp && found_end_of_headers) {
      req->sdp_count = x - req->sdp_start;
      return TRUE;
   }
   return FALSE;
}

static struct ast_cc_agent* find_sip_cc_agent_by_notify_uri

(

const char *const

uri

)

[static, read]

Definition at line 1769 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_notify_uri_helper().

Referenced by cc_esc_publish_handler(), and get_destination().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_notify_uri_helper, (char *)uri, "SIP");
   return agent;
}

static struct ast_cc_agent* find_sip_cc_agent_by_original_callid

(

struct sip_pvt *

pvt

)

[static, read]

Definition at line 1799 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_callid_helper().

Referenced by add_cc_call_info_to_response().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_callid_helper, pvt, "SIP");
   return agent;
}

static struct ast_cc_agent* find_sip_cc_agent_by_subscribe_uri

(

const char *const

uri

)

[static, read]

Definition at line 1784 of file chan_sip.c.

References ast_cc_agent_callback(), and find_by_subscribe_uri_helper().

Referenced by cc_esc_publish_handler(), and handle_cc_subscribe().

{
   struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_subscribe_uri_helper, (char *)uri, "SIP");
   return agent;
}

static int find_sip_method

(

const char *

msg

)

[static]

find_sip_method: Find SIP method from header

Definition at line 3664 of file chan_sip.c.

References ARRAY_LEN, ast_strlen_zero(), cfsip_methods::id, method_match(), and sip_methods.

Referenced by __sip_pretend_ack(), handle_request_do(), handle_response(), mark_parsed_methods(), and sip_hangup().

{
   int i, res = 0;
   
   if (ast_strlen_zero(msg)) {
      return 0;
   }
   for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
      if (method_match(i, msg)) {
         res = sip_methods[i].id;
      }
   }
   return res;
}

static int find_sip_monitor_instance_by_subscription_pvt	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1997 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by handle_cc_notify(), and handle_response_subscribe().

{
   struct sip_monitor_instance *monitor_instance = obj;
   return monitor_instance->subscription_pvt == arg ? CMP_MATCH | CMP_STOP : 0;
}

static int find_sip_monitor_instance_by_suspension_entry	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 2003 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by cc_handle_publish_error().

{
   struct sip_monitor_instance *monitor_instance = obj;
   return monitor_instance->suspension_entry == arg ? CMP_MATCH | CMP_STOP : 0;
}

static struct epa_static_data* find_static_data

(

const char *const

event_package

)

[static, read]

Definition at line 964 of file chan_sip.c.

References AST_LIST_LOCK, AST_LIST_TRAVERSE, and AST_LIST_UNLOCK.

Referenced by create_epa_entry().

{
   const struct epa_backend *backend = NULL;

   AST_LIST_LOCK(&epa_static_data_list);
   AST_LIST_TRAVERSE(&epa_static_data_list, backend, next) {
      if (!strcmp(backend->static_data->name, event_package)) {
         break;
      }
   }
   AST_LIST_UNLOCK(&epa_static_data_list);
   return backend ? backend->static_data : NULL;
}

static struct cfsubscription_types * find_subscription_type

(

enum subscriptiontype

subtype

)

[static, read]

Find subscription type in array.

Definition at line 20923 of file chan_sip.c.

References ARRAY_LEN, subscription_types, and type.

Referenced by transmit_state_notify().

{
   int i;

   for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
      if (subscription_types[i].type == subtype) {
         return &subscription_types[i];
      }
   }
   return &subscription_types[0];
}

static void forked_invite_init	(	struct sip_request *	req,
		const char *	new_theirtag,
		struct sip_pvt *	original,
		struct ast_sockaddr *	addr
	)		[static]

This function creates a dialog to handle a forked request. This dialog exists only to properly terminiate the the forked request immediately.

Definition at line 9018 of file chan_sip.c.

References ast_callid_ref, ast_callid_unref, ast_string_field_set, build_route(), copy_request(), parse_ok_contact(), pvt_set_needdestroy(), sip_alloc(), sip_pvt_lock, sip_pvt_trylock, sip_pvt_unlock, transmit_request(), and TRUE.

Referenced by find_call().

{
   struct sip_pvt *p;
   const char *callid;
   struct ast_callid *logger_callid;

   sip_pvt_lock(original);
   callid = ast_strdupa(original->callid);
   logger_callid = original->logger_callid;
   if (logger_callid) {
      ast_callid_ref(logger_callid);
   }
   sip_pvt_unlock(original);

   p = sip_alloc(callid, addr, 1, SIP_INVITE, req, logger_callid);
   if (logger_callid) {
      ast_callid_unref(logger_callid);
   }
   if (!p)  {
      return; /* alloc error */
   }

   /* Lock p and original private structures. */
   sip_pvt_lock(p);
   while (sip_pvt_trylock(original)) {
      /* Can't use DEADLOCK_AVOIDANCE since p is an ao2 object */
      sip_pvt_unlock(p);
      sched_yield();
      sip_pvt_lock(p);
   }

   p->invitestate = INV_TERMINATED;
   p->ocseq = original->ocseq;
   p->branch = original->branch;

   memcpy(&p->flags, &original->flags, sizeof(p->flags));
   copy_request(&p->initreq, &original->initreq);
   ast_string_field_set(p, theirtag, new_theirtag);
   ast_string_field_set(p, tag, original->tag);
   ast_string_field_set(p, uri, original->uri);
   ast_string_field_set(p, our_contact, original->our_contact);
   ast_string_field_set(p, fullcontact, original->fullcontact);

   sip_pvt_unlock(original);

   parse_ok_contact(p, req);
   build_route(p, req, 1, 0);

   transmit_request(p, SIP_ACK, p->ocseq, XMIT_UNRELIABLE, TRUE);
   transmit_request(p, SIP_BYE, 0, XMIT_RELIABLE, TRUE);

   pvt_set_needdestroy(p, "forked request"); /* this dialog will terminate once the BYE is responed to or times out. */
   sip_pvt_unlock(p);
   dialog_unref(p, "setup forked invite termination");
}

static void free_old_route

(

struct sip_route *

route

)

[static]

Remove route from route list.

Definition at line 16212 of file chan_sip.c.

References ast_free.

Referenced by __sip_destroy(), and build_route().

{
   struct sip_route *next;

   while (route) {
      next = route->next;
      ast_free(route);
      route = next;
   }
}

static int func_check_sipdomain	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)		[static]

Dial plan function to check if domain is local.

Definition at line 22088 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strlen_zero(), check_sip_domain(), and LOG_WARNING.

{
   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "CHECKSIPDOMAIN requires an argument - A domain name\n");
      return -1;
   }
   if (check_sip_domain(data, NULL, 0))
      ast_copy_string(buf, data, len);
   else
      buf[0] = '\0';
   return 0;
}

static int func_header_read	(	struct ast_channel *	chan,
		const char *	function,
		char *	data,
		char *	buf,
		size_t	len
	)		[static]

Read SIP header (dialplan function)

Definition at line 22024 of file chan_sip.c.

References __get_header(), args, AST_APP_ARG, ast_channel_lock, ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, ast_log(), AST_STANDARD_APP_ARGS, ast_strlen_zero(), and LOG_WARNING.

{
   struct sip_pvt *p;
   const char *content = NULL;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(header);
      AST_APP_ARG(number);
   );
   int i, number, start = 0;

   if (!chan) {
      ast_log(LOG_WARNING, "No channel was provided to %s function.\n", function);
      return -1;
   }

   if (ast_strlen_zero(data)) {
      ast_log(LOG_WARNING, "This function requires a header name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(ast_channel_tech(chan))) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   AST_STANDARD_APP_ARGS(args, data);
   if (!args.number) {
      number = 1;
   } else {
      sscanf(args.number, "%30d", &number);
      if (number < 1)
         number = 1;
   }

   p = ast_channel_tech_pvt(chan);

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   for (i = 0; i < number; i++)
      content = __get_header(&p->initreq, args.header, &start);

   if (ast_strlen_zero(content)) {
      ast_channel_unlock(chan);
      return -1;
   }

   ast_copy_string(buf, content, len);
   ast_channel_unlock(chan);

   return 0;
}

static int function_sipchaninfo_read	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)		[static]

${SIPCHANINFO()} Dialplan function - reads sip channel data

Definition at line 22218 of file chan_sip.c.

References ast_channel_lock, ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_unlock, ast_copy_string(), ast_log(), ast_sockaddr_stringify_addr(), and LOG_WARNING.

{
   struct sip_pvt *p;
   static int deprecated = 0;

   *buf = 0;

   if (!chan) {
      ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
      return -1;
   }

   if (!data) {
      ast_log(LOG_WARNING, "This function requires a parameter name.\n");
      return -1;
   }

   ast_channel_lock(chan);
   if (!IS_SIP_TECH(ast_channel_tech(chan))) {
      ast_log(LOG_WARNING, "This function can only be used on SIP channels.\n");
      ast_channel_unlock(chan);
      return -1;
   }

   if (deprecated++ % 20 == 0) {
      /* Deprecated in 1.6.1 */
      ast_log(LOG_WARNING, "SIPCHANINFO() is deprecated.  Please transition to using CHANNEL().\n");
   }

   p = ast_channel_tech_pvt(chan);

   /* If there is no private structure, this channel is no longer alive */
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }

   if (!strcasecmp(data, "peerip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->sa), len);
   } else  if (!strcasecmp(data, "recvip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&p->recv), len);
   } else  if (!strcasecmp(data, "from")) {
      ast_copy_string(buf, p->from, len);
   } else  if (!strcasecmp(data, "uri")) {
      ast_copy_string(buf, p->uri, len);
   } else  if (!strcasecmp(data, "useragent")) {
      ast_copy_string(buf, p->useragent, len);
   } else  if (!strcasecmp(data, "peername")) {
      ast_copy_string(buf, p->peername, len);
   } else if (!strcasecmp(data, "t38passthrough")) {
      if ((p->t38.state == T38_DISABLED) || (p->t38.state == T38_REJECTED)) {
         ast_copy_string(buf, "0", len);
      } else { /* T38 is offered or enabled in this call */
         ast_copy_string(buf, "1", len);
      }
   } else {
      ast_channel_unlock(chan);
      return -1;
   }
   ast_channel_unlock(chan);

   return 0;
}

static int function_sippeer	(	struct ast_channel *	chan,
		const char *	cmd,
		char *	data,
		char *	buf,
		size_t	len
	)		[static]

${SIPPEER()} Dialplan function - reads peer data

Todo:

Will be deprecated after 1.4

Definition at line 22107 of file chan_sip.c.

References ast_codec_pref_index(), ast_copy_string(), ast_free, ast_getformatname(), ast_getformatname_multiple(), ast_log(), ast_print_group(), ast_print_namedgroups(), ast_sockaddr_port, ast_sockaddr_stringify_addr(), ast_str_alloca, ast_str_buffer(), ast_str_create(), ast_test_flag, chanvar, FALSE, LOG_WARNING, ast_variable::name, ast_variable::next, peer_mailboxes_to_str(), peer_status(), sip_find_peer(), sip_unref_peer(), TRUE, and ast_variable::value.

{
   struct sip_peer *peer;
   char *colname;

   if ((colname = strchr(data, ':'))) {   /*! \todo Will be deprecated after 1.4 */
      static int deprecation_warning = 0;
      *colname++ = '\0';
      if (deprecation_warning++ % 10 == 0)
         ast_log(LOG_WARNING, "SIPPEER(): usage of ':' to separate arguments is deprecated.  Please use ',' instead.\n");
   } else if ((colname = strchr(data, ',')))
      *colname++ = '\0';
   else
      colname = "ip";

   if (!(peer = sip_find_peer(data, NULL, TRUE, FINDPEERS, FALSE, 0)))
      return -1;

   if (!strcasecmp(colname, "ip")) {
      ast_copy_string(buf, ast_sockaddr_stringify_addr(&peer->addr), len);
   } else  if (!strcasecmp(colname, "port")) {
      snprintf(buf, len, "%d", ast_sockaddr_port(&peer->addr));
   } else  if (!strcasecmp(colname, "status")) {
      peer_status(peer, buf, len);
   } else  if (!strcasecmp(colname, "language")) {
      ast_copy_string(buf, peer->language, len);
   } else  if (!strcasecmp(colname, "regexten")) {
      ast_copy_string(buf, peer->regexten, len);
   } else  if (!strcasecmp(colname, "limit")) {
      snprintf(buf, len, "%d", peer->call_limit);
   } else  if (!strcasecmp(colname, "busylevel")) {
      snprintf(buf, len, "%d", peer->busy_level);
   } else  if (!strcasecmp(colname, "curcalls")) {
      snprintf(buf, len, "%d", peer->inuse);
   } else if (!strcasecmp(colname, "maxforwards")) {
      snprintf(buf, len, "%d", peer->maxforwards);
   } else  if (!strcasecmp(colname, "accountcode")) {
      ast_copy_string(buf, peer->accountcode, len);
   } else  if (!strcasecmp(colname, "callgroup")) {
      ast_print_group(buf, len, peer->callgroup);
   } else  if (!strcasecmp(colname, "pickupgroup")) {
      ast_print_group(buf, len, peer->pickupgroup);
   } else  if (!strcasecmp(colname, "namedcallgroup")) {
      struct ast_str *tmp_str = ast_str_create(1024);
      if (tmp_str) {
         ast_copy_string(buf, ast_print_namedgroups(&tmp_str, peer->named_callgroups), len);
         ast_free(tmp_str);
      }
   } else  if (!strcasecmp(colname, "namedpickupgroup")) {
      struct ast_str *tmp_str = ast_str_create(1024);
      if (tmp_str) {
         ast_copy_string(buf, ast_print_namedgroups(&tmp_str, peer->named_pickupgroups), len);
         ast_free(tmp_str);
      }
   } else  if (!strcasecmp(colname, "useragent")) {
      ast_copy_string(buf, peer->useragent, len);
   } else  if (!strcasecmp(colname, "mailbox")) {
      struct ast_str *mailbox_str = ast_str_alloca(512);
      peer_mailboxes_to_str(&mailbox_str, peer);
      ast_copy_string(buf, ast_str_buffer(mailbox_str), len);
   } else  if (!strcasecmp(colname, "context")) {
      ast_copy_string(buf, peer->context, len);
   } else  if (!strcasecmp(colname, "expire")) {
      snprintf(buf, len, "%d", peer->expire);
   } else  if (!strcasecmp(colname, "dynamic")) {
      ast_copy_string(buf, peer->host_dynamic ? "yes" : "no", len);
   } else  if (!strcasecmp(colname, "callerid_name")) {
      ast_copy_string(buf, peer->cid_name, len);
   } else  if (!strcasecmp(colname, "callerid_num")) {
      ast_copy_string(buf, peer->cid_num, len);
   } else  if (!strcasecmp(colname, "codecs")) {
      ast_getformatname_multiple(buf, len -1, peer->caps);
   } else if (!strcasecmp(colname, "encryption")) {
      snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
   } else  if (!strncasecmp(colname, "chanvar[", 8)) {
      char *chanvar=colname + 8;
      struct ast_variable *v;

      chanvar = strsep(&chanvar, "]");
      for (v = peer->chanvars ; v ; v = v->next) {
         if (!strcasecmp(v->name, chanvar)) {
            ast_copy_string(buf, v->value, len);
         }
      }
   } else  if (!strncasecmp(colname, "codec[", 6)) {
      char *codecnum;
      struct ast_format codec;

      codecnum = colname + 6; /* move past the '[' */
      codecnum = strsep(&codecnum, "]"); /* trim trailing ']' if any */
      if((ast_codec_pref_index(&peer->prefs, atoi(codecnum), &codec))) {
         ast_copy_string(buf, ast_getformatname(&codec), len);
      } else {
         buf[0] = '\0';
      }
   } else {
      buf[0] = '\0';
   }

   sip_unref_peer(peer, "sip_unref_peer from function_sippeer, just before return");

   return 0;
}

static char * generate_random_string	(	char *	buf,
		size_t	size
	)		[static]

Generate 32 byte random string for callid's etc.

Definition at line 8482 of file chan_sip.c.

References ast_random().

Referenced by ast_rtp_new(), build_callid_pvt(), build_callid_registry(), construct_pidf_body(), and generate_uri().

{
   long val[4];
   int x;

   for (x=0; x<4; x++)
      val[x] = ast_random();
   snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);

   return buf;
}

static char* generate_uri	(	struct sip_pvt *	pvt,
		char *	buf,
		size_t	size
	)		[static]

Definition at line 8494 of file chan_sip.c.

References ast_copy_string(), ast_sockaddr_stringify_remote(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_set(), and generate_random_string().

Referenced by add_cc_call_info_to_response(), and transmit_cc_notify().

{
   struct ast_str *uri = ast_str_alloca(size);
   ast_str_set(&uri, 0, "%s", pvt->socket.type == SIP_TRANSPORT_TLS ? "sips:" : "sip:");
   /* Here would be a great place to generate a UUID, but for now we'll
    * use the handy random string generation function we already have
    */
   ast_str_append(&uri, 0, "%s", generate_random_string(buf, size));
   ast_str_append(&uri, 0, "@%s", ast_sockaddr_stringify_remote(&pvt->ourip));
   ast_copy_string(buf, ast_str_buffer(uri), size);
   return buf;
}

int get_address_family_filter

(

unsigned int

transport

)

[static]

Helper for dns resolution to filter by address family.

Note:

return 0 if addr is [::] else it returns addr's family.

Definition at line 28663 of file chan_sip.c.

References ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), bindaddr, ast_tcptls_session_args::local_address, and ast_sockaddr::ss.

Referenced by __sip_subscribe_mwi_do(), ast_sockaddr_resolve_first(), ast_sockaddr_resolve_first_transport(), build_peer(), proxy_update(), realtime_peer_by_name(), and transmit_register().

{
   const struct ast_sockaddr *addr = NULL;

   if ((transport == SIP_TRANSPORT_UDP) || !transport) {
      addr = &bindaddr;
   } else if (transport == SIP_TRANSPORT_TCP || transport == SIP_TRANSPORT_WS) {
      addr = &sip_tcp_desc.local_address;
   } else if (transport == SIP_TRANSPORT_TLS || transport == SIP_TRANSPORT_WSS) {
      addr = &sip_tls_desc.local_address;
   }

   if (ast_sockaddr_is_ipv6(addr) && ast_sockaddr_is_any(addr)) {
      return 0;
   }

   return addr->ss.ss_family;
}

static int get_also_info	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)		[static]

Call transfer support (old way, deprecated by the IETF)

Note:

does not account for SIPS: uri requirements, nor check transport

Definition at line 17967 of file chan_sip.c.

References ast_canmatch_extension(), ast_channel_macrocontext(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_log(), ast_string_field_set, ast_strlen_zero(), ast_verbose(), context, get_in_brackets(), LOG_WARNING, parse_uri_legacy_check(), pbx_builtin_getvar_helper(), S_OR, sip_cfg, sip_debug_test_pvt(), sip_get_header(), SIP_PEDANTIC_DECODE, and sip_refer_alloc().

Referenced by handle_request_bye().

{
   char tmp[256] = "", *c, *a;
   struct sip_request *req = oreq ? oreq : &p->initreq;
   struct sip_refer *refer = NULL;
   const char *transfer_context = NULL;

   if (!p->refer && !sip_refer_alloc(p))
      return -1;

   refer = p->refer;

   ast_copy_string(tmp, sip_get_header(req, "Also"), sizeof(tmp));
   c = get_in_brackets(tmp);

   if (parse_uri_legacy_check(c, "sip:,sips:", &c, NULL, &a, NULL)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header in Also: transfer (%s)?\n", c);
      return -1;
   }

   SIP_PEDANTIC_DECODE(c);
   SIP_PEDANTIC_DECODE(a);

   if (!ast_strlen_zero(a)) {
      ast_string_field_set(refer, refer_to_domain, a);
   }

   if (sip_debug_test_pvt(p))
      ast_verbose("Looking for %s in %s\n", c, p->context);

   /* Determine transfer context */
   if (p->owner) {
      /* By default, use the context in the channel sending the REFER */
      transfer_context = pbx_builtin_getvar_helper(p->owner, "TRANSFER_CONTEXT");
      if (ast_strlen_zero(transfer_context)) {
         transfer_context = ast_channel_macrocontext(p->owner);
      }
   }
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(p->context, sip_cfg.default_context);
   }

   if (ast_exists_extension(NULL, transfer_context, c, 1, NULL)) {
      /* This is a blind transfer */
      ast_debug(1, "SIP Bye-also transfer to Extension %s@%s \n", c, transfer_context);
      ast_string_field_set(refer, refer_to, c);
      ast_string_field_set(refer, referred_by, "");
      ast_string_field_set(refer, refer_contact, "");
      /* Set new context */
      ast_string_field_set(p, context, transfer_context);
      return 0;
   } else if (ast_canmatch_extension(NULL, p->context, c, 1, NULL)) {
      return 1;
   }

   return -1;
}

static int get_cached_mwi	(	struct sip_peer *	peer,
		int *	new,
		int *	old
	)		[static]

Get cached MWI info.

Returns:

TRUE if found MWI in cache

Definition at line 28819 of file chan_sip.c.

References ast_event_destroy(), ast_event_get_cached(), ast_event_get_ie_uint(), AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_MWI, AST_LIST_TRAVERSE, mailbox, and S_OR.

Referenced by sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;
   int in_cache;

   in_cache = 0;
   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      struct ast_event *event;
      event = ast_event_get_cached(AST_EVENT_MWI,
         AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox->mailbox,
         AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, S_OR(mailbox->context, "default"),
         AST_EVENT_IE_END);
      if (!event)
         continue;
      *new += ast_event_get_ie_uint(event, AST_EVENT_IE_NEWMSGS);
      *old += ast_event_get_ie_uint(event, AST_EVENT_IE_OLDMSGS);
      ast_event_destroy(event);
      in_cache = 1;
   }

   return in_cache;
}

static char * get_content

(

struct sip_request *

req

)

[static]

Get message body content.

Definition at line 8306 of file chan_sip.c.

References ast_str_append(), ast_str_buffer(), ast_str_reset(), ast_str_thread_get(), sip_content_buf, and str.

Referenced by handle_request_info(), handle_request_notify(), receive_message(), and sip_pidf_validate().

{
   struct ast_str *str;
   int i;

   if (!(str = ast_str_thread_get(&sip_content_buf, 128))) {
      return NULL;
   }

   ast_str_reset(str);

   for (i = 0; i < req->lines; i++) {
      if (ast_str_append(&str, 0, "%s\n", REQ_OFFSET_TO_STR(req, line[i])) < 0) {
         return NULL;
      }
   }

   return ast_str_buffer(str);
}

static char* get_content_line	(	struct sip_request *	req,
		char *	name,
		char	delimiter
	)		[static]

Get a specific line from the message content.

Definition at line 8183 of file chan_sip.c.

References ast_skip_blanks(), and len().

Referenced by handle_cc_notify(), handle_request_info(), and handle_request_notify().

{
   int i;
   int len = strlen(name);
   const char *line;

   for (i = 0; i < req->lines; i++) {
      line = REQ_OFFSET_TO_STR(req, line[i]);
      if (!strncasecmp(line, name, len) && line[len] == delimiter) {
         return ast_skip_blanks(line + len + 1);
      }
   }

   return "";
}

static void get_crypto_attrib	(	struct sip_pvt *	p,
		struct sip_srtp *	srtp,
		const char **	a_crypto
	)		[static]

Definition at line 13017 of file chan_sip.c.

References ast_log(), ast_test_flag, LOG_WARNING, sdp_crypto_attrib(), sdp_crypto_offer(), and sdp_crypto_setup().

Referenced by add_sdp().

{
   int taglen = 80;

   /* Set encryption properties */
   if (srtp) {
      if (!srtp->crypto) {
         srtp->crypto = sdp_crypto_setup();
      }

      if (p->dtls_cfg.enabled) {
         /* If DTLS-SRTP is enabled the key details will be pulled from TLS */
         return;
      }

      /* set the key length based on INVITE or settings */
      if (ast_test_flag(srtp, SRTP_CRYPTO_TAG_80)) {
         taglen = 80;
      } else if (ast_test_flag(&p->flags[2], SIP_PAGE3_SRTP_TAG_32) ||
          ast_test_flag(srtp, SRTP_CRYPTO_TAG_32)) {
         taglen = 32;
      }

      if (srtp->crypto && (sdp_crypto_offer(srtp->crypto, taglen) >= 0)) {
         *a_crypto = sdp_crypto_attrib(srtp->crypto);
      }

      if (!*a_crypto) {
         ast_log(LOG_WARNING, "No SRTP key management enabled\n");
      }
   }
}

static enum sip_get_dest_result get_destination	(	struct sip_pvt *	p,
		struct sip_request *	oreq,
		int *	cc_recall_core_id
	)		[static]

Find out who the call is for.

We use the request uri as a destination. This code assumes authentication has been done, so that the device (peer/user) context is already set.

Returns:

0 on success (found a matching extension), non-zero on failure

Note:

If the incoming uri is a SIPS: uri, we are required to carry this across the dialplan, so that the outbound call also is a sips: call or encrypted IAX2 call. If that's not available, the call should FAIL.

Definition at line 17518 of file chan_sip.c.

References ao2_ref, ast_canmatch_extension(), ast_cc_agent_recalling(), ast_copy_string(), ast_debug, ast_exists_extension(), ast_get_hint(), AST_LIST_EMPTY, ast_log(), AST_MAX_EXTENSION, ast_pickup_ext(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_uri_sip_user, ast_verbose(), check_sip_domain(), context, ast_cc_agent::core_id, ast_cc_agent::device_name, exten, extract_host_from_hostport(), find_sip_cc_agent_by_notify_uri(), get_in_brackets(), LOG_WARNING, parse_uri_legacy_check(), ast_cc_agent::private_data, S_OR, sip_cfg, sip_debug_test_pvt(), sip_get_header(), sip_methods, SIP_PEDANTIC_DECODE, and cfsip_methods::text.

Referenced by handle_request_invite(), handle_request_options(), handle_request_subscribe(), and receive_message().

{
   char tmp[256] = "", *uri, *unused_password, *domain;
   char tmpf[256] = "", *from = NULL;
   struct sip_request *req;
   char *decoded_uri;

   req = oreq;
   if (!req) {
      req = &p->initreq;
   }

   /* Find the request URI */
   if (req->rlpart2) {
      ast_copy_string(tmp, REQ_OFFSET_TO_STR(req, rlpart2), sizeof(tmp));
   }

   uri = ast_strdupa(get_in_brackets(tmp));

   if (parse_uri_legacy_check(uri, "sip:,sips:", &uri, &unused_password, &domain, NULL)) {
      ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", uri);
      return SIP_GET_DEST_INVALID_URI;
   }

   SIP_PEDANTIC_DECODE(domain);
   SIP_PEDANTIC_DECODE(uri);

   extract_host_from_hostport(&domain);

   if (ast_strlen_zero(uri)) {
      /*
       * Either there really was no extension found or the request
       * URI had encoded nulls that made the string "empty".  Use "s"
       * as the extension.
       */
      uri = "s";
   }

   ast_string_field_set(p, domain, domain);

   /* Now find the From: caller ID and name */
   /* XXX Why is this done in get_destination? Isn't it already done?
      Needs to be checked
        */
   ast_copy_string(tmpf, sip_get_header(req, "From"), sizeof(tmpf));
   if (!ast_strlen_zero(tmpf)) {
      from = get_in_brackets(tmpf);
      if (parse_uri_legacy_check(from, "sip:,sips:", &from, NULL, &domain, NULL)) {
         ast_log(LOG_WARNING, "Not a SIP header (%s)?\n", from);
         return SIP_GET_DEST_INVALID_URI;
      }

      SIP_PEDANTIC_DECODE(from);
      SIP_PEDANTIC_DECODE(domain);

      extract_host_from_hostport(&domain);

      ast_string_field_set(p, fromdomain, domain);
   }

   if (!AST_LIST_EMPTY(&domain_list)) {
      char domain_context[AST_MAX_EXTENSION];

      domain_context[0] = '\0';
      if (!check_sip_domain(p->domain, domain_context, sizeof(domain_context))) {
         if (!sip_cfg.allow_external_domains && (req->method == SIP_INVITE || req->method == SIP_REFER)) {
            ast_debug(1, "Got SIP %s to non-local domain '%s'; refusing request.\n", sip_methods[req->method].text, p->domain);
            return SIP_GET_DEST_REFUSED;
         }
      }
      /* If we don't have a peer (i.e. we're a guest call),
       * overwrite the original context */
      if (!ast_test_flag(&p->flags[1], SIP_PAGE2_HAVEPEERCONTEXT) && !ast_strlen_zero(domain_context)) {
         ast_string_field_set(p, context, domain_context);
      }
   }

   /* If the request coming in is a subscription and subscribecontext has been specified use it */
   if (req->method == SIP_SUBSCRIBE && !ast_strlen_zero(p->subscribecontext)) {
      ast_string_field_set(p, context, p->subscribecontext);
   }

   if (sip_debug_test_pvt(p)) {
      ast_verbose("Looking for %s in %s (domain %s)\n", uri, p->context, p->domain);
   }

   /* Since extensions.conf can have unescaped characters, try matching a
    * decoded uri in addition to the non-decoded uri. */
   decoded_uri = ast_strdupa(uri);
   ast_uri_decode(decoded_uri, ast_uri_sip_user);

   /* If this is a subscription we actually just need to see if a hint exists for the extension */
   if (req->method == SIP_SUBSCRIBE) {
      char hint[AST_MAX_EXTENSION];
      int which = 0;
      if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, uri) ||
          (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, p->context, decoded_uri) && (which = 1))) {
         if (!oreq) {
            ast_string_field_set(p, exten, which ? decoded_uri : uri);
         }
         return SIP_GET_DEST_EXTEN_FOUND;
      } else {
         return SIP_GET_DEST_EXTEN_NOT_FOUND;
      }
   } else {
      struct ast_cc_agent *agent;
      /* Check the dialplan for the username part of the request URI,
         the domain will be stored in the SIPDOMAIN variable
         Return 0 if we have a matching extension */
      if (ast_exists_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))) {
         if (!oreq) {
            ast_string_field_set(p, exten, uri);
         }
         return SIP_GET_DEST_EXTEN_FOUND;
      }
      if (ast_exists_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
         || !strcmp(decoded_uri, ast_pickup_ext())) {
         if (!oreq) {
            ast_string_field_set(p, exten, decoded_uri);
         }
         return SIP_GET_DEST_EXTEN_FOUND;
      }
      if ((agent = find_sip_cc_agent_by_notify_uri(tmp))) {
         struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
         /* This is a CC recall. We can set p's extension to the exten from
          * the original INVITE
          */
         ast_string_field_set(p, exten, agent_pvt->original_exten);
         /* And we need to let the CC core know that the caller is attempting
          * his recall
          */
         ast_cc_agent_recalling(agent->core_id, "SIP caller %s is attempting recall",
               agent->device_name);
         if (cc_recall_core_id) {
            *cc_recall_core_id = agent->core_id;
         }
         ao2_ref(agent, -1);
         return SIP_GET_DEST_EXTEN_FOUND;
      }
   }

   if (ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)
      && (ast_canmatch_extension(NULL, p->context, uri, 1, S_OR(p->cid_num, from))
         || ast_canmatch_extension(NULL, p->context, decoded_uri, 1, S_OR(p->cid_num, from))
         || !strncmp(decoded_uri, ast_pickup_ext(), strlen(decoded_uri)))) {
      /* Overlap dialing is enabled and we need more digits to match an extension. */
      return SIP_GET_DEST_EXTEN_MATCHMORE;
   }

   return SIP_GET_DEST_EXTEN_NOT_FOUND;
}

static int get_domain	(	const char *	str,
		char *	domain,
		int	len
	)		[static]

Extract domain from SIP To/From header.

Returns:

-1 on error, 1 if domain string is empty, 0 if domain was properly extracted

Note:

TODO: Such code is all over SIP channel, there is a sense to organize this patern in one function

Definition at line 12296 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strlen_zero(), get_in_brackets(), and LOG_WARNING.

Referenced by get_realm().

{
   char tmpf[256];
   char *a, *from;

   *domain = '\0';
   ast_copy_string(tmpf, str, sizeof(tmpf));
   from = get_in_brackets(tmpf);
   if (!ast_strlen_zero(from)) {
      if (strncasecmp(from, "sip:", 4)) {
         ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", from);
         return -1;
      }
      from += 4;
   } else
      from = NULL;

   if (from) {
      int bracket = 0;

      /* Strip any params or options from user */
      if ((a = strchr(from, ';')))
         *a = '\0';
      /* Strip port from domain if present */
      for (a = from; *a != '\0'; ++a) {
         if (*a == ':' && bracket == 0) {
            *a = '\0';
            break;
         } else if (*a == '[') {
            ++bracket;
         } else if (*a == ']') {
            --bracket;
         }
      }
      if ((a = strchr(from, '@'))) {
         *a = '\0';
         ast_copy_string(domain, a + 1, len);
      } else
         ast_copy_string(domain, from, len);
   }

   return ast_strlen_zero(domain);
}

static struct event_state_compositor* get_esc

(

const char *const

event_package

)

[static, read]

Definition at line 1058 of file chan_sip.c.

References ARRAY_LEN, event_state_compositors, and name.

Referenced by create_new_sip_etag(), handle_request_publish(), and publish_expire().

                                                                                {
   int i;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      if (!strcasecmp(event_package, event_state_compositors[i].name)) {
         return &event_state_compositors[i];
      }
   }
   return NULL;
}

static struct sip_esc_entry* get_esc_entry	(	const char *	entity_tag,
		struct event_state_compositor *	esc
	)		[static, read]

Definition at line 1068 of file chan_sip.c.

References ao2_find, ast_copy_string(), event_state_compositor::compositor, and OBJ_POINTER.

Referenced by handle_sip_publish_modify(), handle_sip_publish_refresh(), and handle_sip_publish_remove().

                                                                                                        {
   struct sip_esc_entry *entry;
   struct sip_esc_entry finder;

   ast_copy_string(finder.entity_tag, entity_tag, sizeof(finder.entity_tag));

   entry = ao2_find(esc->compositor, &finder, OBJ_POINTER);

   return entry;
}

static struct ast_variable * get_insecure_variable_from_config

(

struct ast_config *

config

)

[static, read]

Definition at line 5263 of file chan_sip.c.

References ast_category_browse(), ast_category_root(), ast_test_flag, ast_variable_retrieve(), set_insecure_flags(), and var.

Referenced by get_insecure_variable_from_sippeers().

{
   struct ast_variable *var = NULL;
   struct ast_flags flags = {0};
   char *cat = NULL;
   const char *insecure;
   while ((cat = ast_category_browse(cfg, cat))) {
      insecure = ast_variable_retrieve(cfg, cat, "insecure");
      set_insecure_flags(&flags, insecure, -1);
      if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
         var = ast_category_root(cfg, cat);
         break;
      }
   }
   return var;
}

static struct ast_variable* get_insecure_variable_from_sippeers	(	const char *	column,
		const char *	value
	)		[static, read]

Definition at line 5280 of file chan_sip.c.

References ast_config_destroy(), ast_load_realtime_multientry(), ast_variables_dup(), get_insecure_variable_from_config(), SENTINEL, and var.

Referenced by realtime_peer_by_addr().

{
   struct ast_config *peerlist;
   struct ast_variable *var = NULL;
   if ((peerlist = ast_load_realtime_multientry("sippeers", column, value, "insecure LIKE", "%port%", SENTINEL))) {
      if ((var = get_insecure_variable_from_config(peerlist))) {
         /* Must clone, because var will get freed along with
          * peerlist. */
         var = ast_variables_dup(var);
      }
      ast_config_destroy(peerlist);
   }
   return var;
}

static struct ast_variable* get_insecure_variable_from_sipregs	(	const char *	column,
		const char *	value,
		struct ast_variable **	var
	)		[static, read]

Definition at line 5300 of file chan_sip.c.

References ast_category_browse(), ast_category_root(), ast_config_destroy(), ast_load_realtime_multientry(), ast_test_flag, ast_variable_retrieve(), ast_variables_destroy(), ast_variables_dup(), peers, SENTINEL, and set_insecure_flags().

Referenced by realtime_peer_by_addr().

{
   struct ast_variable *varregs = NULL;
   struct ast_config *regs, *peers;
   char *regscat;
   const char *regname;

   if (!(regs = ast_load_realtime_multientry("sipregs", column, value, SENTINEL))) {
      return NULL;
   }

   /* Load *all* peers that are probably insecure=port */
   if (!(peers = ast_load_realtime_multientry("sippeers", "insecure LIKE", "%port%", SENTINEL))) {
      ast_config_destroy(regs);
      return NULL;
   }

   /* Loop over the sipregs that match IP address and attempt to find an
    * insecure=port match to it in sippeers. */
   regscat = NULL;
   while ((regscat = ast_category_browse(regs, regscat)) && (regname = ast_variable_retrieve(regs, regscat, "name"))) {
      char *peerscat;
      const char *peername;

      peerscat = NULL;
      while ((peerscat = ast_category_browse(peers, peerscat)) && (peername = ast_variable_retrieve(peers, peerscat, "name"))) {
         if (!strcasecmp(regname, peername)) {
            /* Ensure that it really is insecure=port and
             * not something else. */
            const char *insecure = ast_variable_retrieve(peers, peerscat, "insecure");
            struct ast_flags flags = {0};
            set_insecure_flags(&flags, insecure, -1);
            if (ast_test_flag(&flags, SIP_INSECURE_PORT)) {
               /* ENOMEM checks till the bitter end. */
               if ((varregs = ast_variables_dup(ast_category_root(regs, regscat)))) {
                  if (!(*var = ast_variables_dup(ast_category_root(peers, peerscat)))) {
                     ast_variables_destroy(varregs);
                     varregs = NULL;
                  }
               }
               goto done;
            }
         }
      }
   }

done:
   ast_config_destroy(regs);
   ast_config_destroy(peers);
   return varregs;
}

static int get_ip_and_port_from_sdp	(	struct sip_request *	req,
		const enum media_type	media,
		struct ast_sockaddr *	addr
	)		[static]

Definition at line 9823 of file chan_sip.c.

References ast_log(), ast_sockaddr_resolve_first_af(), ast_strlen_zero(), get_sdp_iterate(), len(), and LOG_WARNING.

Referenced by handle_request_invite().

{
   const char *m;
   const char *c;
   int miterator = req->sdp_start;
   int citerator = req->sdp_start;
   int x = 0;
   int numberofports;
   int len;
   int af;
   char proto[4], host[258] = ""; /*Initialize to empty so we will know if we have any input */

   c = get_sdp_iterate(&citerator, req, "c");
   if (sscanf(c, "IN %3s %256s", proto, host) != 2) {
         ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
         /* Continue since there may be a valid host in a c= line specific to the audio stream */
   }
   /* We only want the m and c lines for audio */
   for (m = get_sdp_iterate(&miterator, req, "m"); !ast_strlen_zero(m); m = get_sdp_iterate(&miterator, req, "m")) {
      if ((media == SDP_AUDIO && ((sscanf(m, "audio %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "audio %30u RTP/AVP %n", &x, &len) == 1 && len > 0))) ||
         (media == SDP_VIDEO && ((sscanf(m, "video %30u/%30u RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
          (sscanf(m, "video %30u RTP/AVP %n", &x, &len) == 1 && len > 0)))) {
         /* See if there's a c= line for this media stream.
          * XXX There is no guarantee that we'll be grabbing the c= line for this
          * particular media stream here. However, this is the same logic used in process_sdp.
          */
         c = get_sdp_iterate(&citerator, req, "c");
         if (!ast_strlen_zero(c)) {
            sscanf(c, "IN %3s %256s", proto, host);
         }
         break;
      }
   }

   if (!strcmp("IP4", proto)) {
      af = AF_INET;
   } else if (!strcmp("IP6", proto)) {
      af = AF_INET6;
   } else {
      ast_log(LOG_WARNING, "Unknown protocol '%s'.\n", proto);
      return -1;
   }

   if (ast_strlen_zero(host) || x == 0) {
      ast_log(LOG_WARNING, "Failed to read an alternate host or port in SDP. Expect %s problems\n", media == SDP_AUDIO ? "audio" : "video");
      return -1;
   }

   if (ast_sockaddr_resolve_first_af(addr, host, 0, af)) {
      ast_log(LOG_WARNING, "Could not look up IP address of alternate hostname. Expect %s problems\n", media == SDP_AUDIO? "audio" : "video");
      return -1;
   }

   return 0;
}

static const char * get_name_from_variable

(

const struct ast_variable *

var

)

[static]

Definition at line 5352 of file chan_sip.c.

References ast_strlen_zero(), ast_variable::name, ast_variable::next, and ast_variable::value.

Referenced by realtime_peer_by_addr(), and realtime_peer_get_sippeer_helper().

{
   /* Don't expect this to return non-NULL. Both NULL and empty
    * values can cause the option to get removed from the variable
    * list. This is called on ast_variables gotten from both
    * ast_load_realtime and ast_load_realtime_multientry.
    * - ast_load_realtime removes options with empty values
    * - ast_load_realtime_multientry does not!
    * For consistent behaviour, we check for the empty name and
    * return NULL instead. */
   const struct ast_variable *tmp;
   for (tmp = var; tmp; tmp = tmp->next) {
      if (!strcasecmp(tmp->name, "name")) {
         if (!ast_strlen_zero(tmp->value)) {
            return tmp->value;
         }
         break;
      }
   }
   return NULL;
}

static void get_our_media_address	(	struct sip_pvt *	p,
		int	needvideo,
		int	needtext,
		struct ast_sockaddr *	addr,
		struct ast_sockaddr *	vaddr,
		struct ast_sockaddr *	taddr,
		struct ast_sockaddr *	dest,
		struct ast_sockaddr *	vdest,
		struct ast_sockaddr *	tdest
	)		[static]

Set all IP media addresses for this call.

Note:

called from add_sdp()

Definition at line 12924 of file chan_sip.c.

References ast_rtp_instance_get_local_address(), ast_sockaddr_cmp_addr(), ast_sockaddr_copy(), ast_sockaddr_is_any(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, and media_address.

Referenced by add_sdp().

{
   int use_externip = 0;

   /* First, get our address */
   ast_rtp_instance_get_local_address(p->rtp, addr);
   if (p->vrtp) {
      ast_rtp_instance_get_local_address(p->vrtp, vaddr);
   }
   if (p->trtp) {
      ast_rtp_instance_get_local_address(p->trtp, taddr);
   }

   /* If our real IP differs from the local address returned by the RTP engine, use it. */
   /* The premise is that if we are already using that IP to communicate with the client, */
   /* we should be using it for RTP too. */
        use_externip = ast_sockaddr_cmp_addr(&p->ourip, addr);

   /* Now, try to figure out where we want them to send data */
   /* Is this a re-invite to move the media out, then use the original offer from caller  */
   if (!ast_sockaddr_isnull(&p->redirip)) {  /* If we have a redirection IP, use it */
      ast_sockaddr_copy(dest, &p->redirip);
   } else {
      /*
       * Audio Destination IP:
       *
       * 1. Specifically configured media address.
       * 2. Local address as specified by the RTP engine.
       * 3. The local IP as defined by chan_sip.
       *
       * Audio Destination Port:
       *
       * 1. Provided by the RTP engine.
       */
      ast_sockaddr_copy(dest,
              !ast_sockaddr_isnull(&media_address) ? &media_address :
              !ast_sockaddr_is_any(addr) && !use_externip ? addr    :
              &p->ourip);
      ast_sockaddr_set_port(dest, ast_sockaddr_port(addr));
   }

   if (needvideo) {
      /* Determine video destination */
      if (!ast_sockaddr_isnull(&p->vredirip)) {
         ast_sockaddr_copy(vdest, &p->vredirip);
      } else {
         /*
          * Video Destination IP:
          *
          * 1. Specifically configured media address.
          * 2. Local address as specified by the RTP engine.
          * 3. The local IP as defined by chan_sip.
          *
          * Video Destination Port:
          *
          * 1. Provided by the RTP engine.
          */
         ast_sockaddr_copy(vdest,
                 !ast_sockaddr_isnull(&media_address) ? &media_address :
                 !ast_sockaddr_is_any(vaddr) && !use_externip ? vaddr  :
                 &p->ourip);
         ast_sockaddr_set_port(vdest, ast_sockaddr_port(vaddr));
      }
   }

   if (needtext) {
      /* Determine text destination */
      if (!ast_sockaddr_isnull(&p->tredirip)) {
         ast_sockaddr_copy(tdest, &p->tredirip);
      } else {
         /*
          * Text Destination IP:
          *
          * 1. Specifically configured media address.
          * 2. Local address as specified by the RTP engine.
          * 3. The local IP as defined by chan_sip.
          *
          * Text Destination Port:
          *
          * 1. Provided by the RTP engine.
          */
         ast_sockaddr_copy(tdest,
                 !ast_sockaddr_isnull(&media_address) ? &media_address  :
                 !ast_sockaddr_is_any(taddr) && !use_externip ? taddr   :
                 &p->ourip);
         ast_sockaddr_set_port(tdest, ast_sockaddr_port(taddr));
      }
   }
}

static int get_pai	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Parse the parts of the P-Asserted-Identity header on an incoming packet. Returns 1 if a valid header is found and it is different from the current caller id.

Definition at line 17225 of file chan_sip.c.

References ast_channel_caller(), ast_copy_string(), ast_free, ast_is_shrinkable_phonenumber(), AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, ast_set_callerid(), ast_shrink_phone_number(), ast_string_field_set, ast_strlen_zero(), cid_name, cid_num, get_in_brackets(), get_name_and_number(), ast_party_caller::id, ast_party_id::name, ast_party_id::number, ast_party_name::presentation, ast_party_number::presentation, and sip_get_header().

Referenced by get_rpid().

{
   char pai[256];
   char privacy[64];
   char *cid_num = NULL;
   char *cid_name = NULL;
   char emptyname[1] = "";
   int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
   char *uri = NULL;
   int is_anonymous = 0, do_update = 1, no_name = 0;

   ast_copy_string(pai, sip_get_header(req, "P-Asserted-Identity"), sizeof(pai));

   if (ast_strlen_zero(pai)) {
      return 0;
   }

   /* use the reqresp_parser function get_name_and_number*/
   if (get_name_and_number(pai, &cid_name, &cid_num)) {
      return 0;
   }

   if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num)) {
      ast_shrink_phone_number(cid_num);
   }

   uri = get_in_brackets(pai);
   if (!strncasecmp(uri, "sip:anonymous@anonymous.invalid", 31)) {
      callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
      /*XXX Assume no change in cid_num. Perhaps it should be
       * blanked?
       */
      ast_free(cid_num);
      is_anonymous = 1;
      cid_num = (char *)p->cid_num;
   }

   ast_copy_string(privacy, sip_get_header(req, "Privacy"), sizeof(privacy));
   if (!ast_strlen_zero(privacy) && !strncmp(privacy, "id", 2)) {
      callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
   }
   if (!cid_name) {
      no_name = 1;
      cid_name = (char *)emptyname;
   }  
   /* Only return true if the supplied caller id is different */
   if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres) {
      do_update = 0;
   } else {

      ast_string_field_set(p, cid_num, cid_num);
      ast_string_field_set(p, cid_name, cid_name);
      p->callingpres = callingpres;

      if (p->owner) {
         ast_set_callerid(p->owner, cid_num, cid_name, NULL);
         ast_channel_caller(p->owner)->id.name.presentation = callingpres;
         ast_channel_caller(p->owner)->id.number.presentation = callingpres;
      }
   }

   /* get_name_and_number allocates memory for cid_num and cid_name so we have to free it */
   if (!is_anonymous) {
      ast_free(cid_num);
   }
   if (!no_name) {
      ast_free(cid_name);
   }

   return do_update;
}

static int get_rdnis	(	struct sip_pvt *	p,
		struct sip_request *	oreq,
		char **	name,
		char **	number,
		int *	reason
	)		[static]

Get referring dnis.

Definition at line 17418 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_strdup, ast_strip_quoted(), ast_strlen_zero(), ast_verbose(), exten, get_in_brackets(), LOG_WARNING, pbx_builtin_setvar_helper(), S_OR, sip_debug_test_pvt(), sip_get_header(), sip_reason_str_to_code(), and sip_set_redirstr().

Referenced by change_redirecting_information().

{
   char tmp[256], *exten, *rexten, *rdomain, *rname = NULL;
   char *params, *reason_param = NULL;
   struct sip_request *req;

   req = oreq ? oreq : &p->initreq;

   ast_copy_string(tmp, sip_get_header(req, "Diversion"), sizeof(tmp));
   if (ast_strlen_zero(tmp))
      return -1;

   if ((params = strchr(tmp, '>'))) {
      params = strchr(params, ';');
   }

   exten = get_in_brackets(tmp);
   if (!strncasecmp(exten, "sip:", 4)) {
      exten += 4;
   } else if (!strncasecmp(exten, "sips:", 5)) {
      exten += 5;
   } else {
      ast_log(LOG_WARNING, "Huh?  Not an RDNIS SIP header (%s)?\n", exten);
      return -1;
   }

   /* Get diversion-reason param if present */
   if (params) {
      *params = '\0';   /* Cut off parameters  */
      params++;
      while (*params == ';' || *params == ' ')
         params++;
      /* Check if we have a reason parameter */
      if ((reason_param = strcasestr(params, "reason="))) {
         char *end;
         reason_param+=7;
         if ((end = strchr(reason_param, ';'))) {
            *end = '\0';
         }
         /* Remove enclosing double-quotes */
         if (*reason_param == '"')
            reason_param = ast_strip_quoted(reason_param, "\"", "\"");
         if (!ast_strlen_zero(reason_param)) {
            sip_set_redirstr(p, reason_param);
            if (p->owner) {
               pbx_builtin_setvar_helper(p->owner, "__PRIREDIRECTREASON", p->redircause);
               pbx_builtin_setvar_helper(p->owner, "__SIPREDIRECTREASON", reason_param);
            }
         }
      }
   }

   rdomain = exten;
   rexten = strsep(&rdomain, "@");  /* trim anything after @ */
   if (p->owner)
      pbx_builtin_setvar_helper(p->owner, "__SIPRDNISDOMAIN", rdomain);

   if (sip_debug_test_pvt(p))
      ast_verbose("RDNIS for this call is %s (reason %s)\n", exten, S_OR(reason_param, ""));

   /*ast_string_field_set(p, rdnis, rexten);*/

   if (*tmp == '\"') {
      char *end_quote;
      rname = tmp + 1;
      end_quote = strchr(rname, '\"');
      if (end_quote) {
         *end_quote = '\0';
      }
   }

   if (number) {
      *number = ast_strdup(rexten);
   }

   if (name && rname) {
      *name = ast_strdup(rname);
   }

   if (reason && !ast_strlen_zero(reason_param)) {
      *reason = sip_reason_str_to_code(reason_param);
   }

   return 0;
}

static void get_realm	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)		[static]

Choose realm based on From header and then To header or use globaly configured realm. Realm from From/To header should be listed among served domains in config file: domain=...

Definition at line 12344 of file chan_sip.c.

References AST_LIST_EMPTY, ast_string_field_set, ast_strlen_zero(), check_sip_domain(), get_domain(), sip_cfg, and sip_get_header().

Referenced by transmit_response_with_auth().

{
   char domain[MAXHOSTNAMELEN];

   if (!ast_strlen_zero(p->realm))
      return;

   if (sip_cfg.domainsasrealm &&
       !AST_LIST_EMPTY(&domain_list))
   {
      /* Check From header first */
      if (!get_domain(sip_get_header(req, "From"), domain, sizeof(domain))) {
         if (check_sip_domain(domain, NULL, 0)) {
            ast_string_field_set(p, realm, domain);
            return;
         }
      }
      /* Check To header */
      if (!get_domain(sip_get_header(req, "To"), domain, sizeof(domain))) {
         if (check_sip_domain(domain, NULL, 0)) {
            ast_string_field_set(p, realm, domain);
            return;
         }
      }
   }
   
   /* Use default realm from config file */
   ast_string_field_set(p, realm, sip_cfg.realm);
}

static int get_refer_info	(	struct sip_pvt *	transferer,
		struct sip_request *	outgoing_req
	)		[static]

Call transfer support (the REFER method) Extracts Refer headers into pvt dialog structure.

Note:

If we get a SIPS uri in the refer-to header, we're required to set up a secure signalling path to that extension. As a minimum, this needs to be added to a channel variable, if not a channel flag.

Definition at line 17765 of file chan_sip.c.

References ast_bridged_channel(), ast_channel_macrocontext(), ast_debug, ast_exists_extension(), ast_log(), ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_uri_decode(), ast_uri_sip_user, ast_verbose(), ast_channel::context, get_in_brackets(), LOG_WARNING, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), S_OR, sip_cfg, sip_debug_test_pvt(), sip_get_header(), and SIP_PEDANTIC_DECODE.

Referenced by handle_request_refer().

{
   const char *p_referred_by = NULL;
   char *h_refer_to = NULL;
   char *h_referred_by = NULL;
   char *refer_to;
   const char *p_refer_to;
   char *referred_by_uri = NULL;
   char *ptr;
   struct sip_request *req = NULL;
   const char *transfer_context = NULL;
   struct sip_refer *refer;

   req = outgoing_req;
   refer = transferer->refer;

   if (!req) {
      req = &transferer->initreq;
   }

   p_refer_to = sip_get_header(req, "Refer-To");
   if (ast_strlen_zero(p_refer_to)) {
      ast_log(LOG_WARNING, "Refer-To Header missing. Skipping transfer.\n");
      return -2;  /* Syntax error */
   }
   h_refer_to = ast_strdupa(p_refer_to);
   refer_to = get_in_brackets(h_refer_to);
   if (!strncasecmp(refer_to, "sip:", 4)) {
      refer_to += 4;       /* Skip sip: */
   } else if (!strncasecmp(refer_to, "sips:", 5)) {
      refer_to += 5;
   } else {
      ast_log(LOG_WARNING, "Can't transfer to non-sip: URI.  (Refer-to: %s)?\n", refer_to);
      return -3;
   }

   /* Get referred by header if it exists */
   p_referred_by = sip_get_header(req, "Referred-By");

   /* Give useful transfer information to the dialplan */
   if (transferer->owner) {
      struct ast_channel *peer = ast_bridged_channel(transferer->owner);
      if (peer) {
         pbx_builtin_setvar_helper(peer, "SIPREFERRINGCONTEXT", transferer->context);
         pbx_builtin_setvar_helper(peer, "SIPREFERREDBYHDR", p_referred_by);
      }
   }

   if (!ast_strlen_zero(p_referred_by)) {
      h_referred_by = ast_strdupa(p_referred_by);

      referred_by_uri = get_in_brackets(h_referred_by);

      if (!strncasecmp(referred_by_uri, "sip:", 4)) {
         referred_by_uri += 4;      /* Skip sip: */
      } else if (!strncasecmp(referred_by_uri, "sips:", 5)) {
         referred_by_uri += 5;      /* Skip sips: */
      } else {
         ast_log(LOG_WARNING, "Huh?  Not a sip: header (Referred-by: %s). Skipping.\n", referred_by_uri);
         referred_by_uri = NULL;
      }
   }

   /* Check for arguments in the refer_to header */
   if ((ptr = strcasestr(refer_to, "replaces="))) {
      char *to = NULL, *from = NULL, *callid;

      /* This is an attended transfer */
      refer->attendedtransfer = 1;

      callid = ast_strdupa(ptr + 9);
      ast_uri_decode(callid, ast_uri_sip_user);
      if ((ptr = strchr(callid, ';'))) { /* Find options */
         *ptr++ = '\0';
      }
      ast_string_field_set(refer, replaces_callid, callid);

      if (ptr) {
         /* Find the different tags before we destroy the string */
         to = strcasestr(ptr, "to-tag=");
         from = strcasestr(ptr, "from-tag=");
      }
      
      /* Grab the to header */
      if (to) {
         ptr = to + 7;
         if ((to = strchr(ptr, '&'))) {
            *to = '\0';
         }
         if ((to = strchr(ptr, ';'))) {
            *to = '\0';
         }
         ast_string_field_set(refer, replaces_callid_totag, ptr);
      }

      if (from) {
         ptr = from + 9;
         if ((from = strchr(ptr, '&'))) {
            *from = '\0';
         }
         if ((from = strchr(ptr, ';'))) {
            *from = '\0';
         }
         ast_string_field_set(refer, replaces_callid_fromtag, ptr);
      }

      if (!strcmp(refer->replaces_callid, transferer->callid) &&
         (!sip_cfg.pedanticsipchecking ||
         (!strcmp(refer->replaces_callid_fromtag, transferer->theirtag) &&
         !strcmp(refer->replaces_callid_totag, transferer->tag)))) {
            ast_log(LOG_WARNING, "Got an attempt to replace own Call-ID on %s\n", transferer->callid);
            return -4;
      }

      if (!sip_cfg.pedanticsipchecking) {
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s (No check of from/to tags)\n", refer->replaces_callid);
      } else {
         ast_debug(2, "Attended transfer: Will use Replace-Call-ID : %s F-tag: %s T-tag: %s\n", refer->replaces_callid, refer->replaces_callid_fromtag ? refer->replaces_callid_fromtag : "<none>", refer->replaces_callid_totag ? refer->replaces_callid_totag : "<none>");
      }
   }

   if ((ptr = strchr(refer_to, '@'))) {   /* Separate domain */
      char *urioption = NULL, *domain;
      int bracket = 0;
      *ptr++ = '\0';

      if ((urioption = strchr(ptr, ';'))) { /* Separate urioptions */
         *urioption++ = '\0';
      }

      domain = ptr;

      /* Remove :port */
      for (; *ptr != '\0'; ++ptr) {
         if (*ptr == ':' && bracket == 0) {
            *ptr = '\0';
            break;
         } else if (*ptr == '[') {
            ++bracket;
         } else if (*ptr == ']') {
            --bracket;
         }
      }

      SIP_PEDANTIC_DECODE(domain);
      SIP_PEDANTIC_DECODE(urioption);

      /* Save the domain for the dial plan */
      ast_string_field_set(refer, refer_to_domain, domain);
      if (urioption) {
         ast_string_field_set(refer, refer_to_urioption, urioption);
      }
   }

   if ((ptr = strchr(refer_to, ';'))) /* Remove options */
      *ptr = '\0';

   SIP_PEDANTIC_DECODE(refer_to);
   ast_string_field_set(refer, refer_to, refer_to);

   if (referred_by_uri) {
      if ((ptr = strchr(referred_by_uri, ';'))) /* Remove options */
         *ptr = '\0';
      SIP_PEDANTIC_DECODE(referred_by_uri);
      ast_string_field_build(refer, referred_by, "<sip:%s>", referred_by_uri);
   } else {
      ast_string_field_set(refer, referred_by, "");
   }

   /* Determine transfer context */
   if (transferer->owner) {
      /* By default, use the context in the channel sending the REFER */
      transfer_context = pbx_builtin_getvar_helper(transferer->owner, "TRANSFER_CONTEXT");
      if (ast_strlen_zero(transfer_context)) {
         transfer_context = ast_channel_macrocontext(transferer->owner);
      }
   }
   if (ast_strlen_zero(transfer_context)) {
      transfer_context = S_OR(transferer->context, sip_cfg.default_context);
   }

   ast_string_field_set(refer, refer_to_context, transfer_context);

   /* Either an existing extension or the parking extension */
   if (refer->attendedtransfer || ast_exists_extension(NULL, transfer_context, refer_to, 1, NULL)) {
      if (sip_debug_test_pvt(transferer)) {
         ast_verbose("SIP transfer to extension %s@%s by %s\n", refer_to, transfer_context, referred_by_uri);
      }
      /* We are ready to transfer to the extension */
      return 0;
   }
   if (sip_debug_test_pvt(transferer))
      ast_verbose("Failed SIP Transfer to non-existing extension %s in context %s\n n", refer_to, transfer_context);

   /* Failure, we can't find this extension */
   return -1;
}

static int get_rpid	(	struct sip_pvt *	p,
		struct sip_request *	oreq
	)		[static]

Get name, number and presentation from remote party id header, returns true if a valid header was found and it was different from the current caller id.

Definition at line 17301 of file chan_sip.c.

References ast_channel_caller(), ast_copy_string(), ast_is_shrinkable_phonenumber(), AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED, AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED, AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN, ast_set_callerid(), ast_shrink_phone_number(), ast_skip_blanks(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, cid_name, cid_num, get_pai(), ast_party_caller::id, ast_party_id::name, ast_party_id::number, ast_party_name::presentation, ast_party_number::presentation, and sip_get_header().

Referenced by check_peer_ok(), check_user_full(), handle_request_invite(), handle_request_update(), and handle_response_invite().

{
   char tmp[256];
   struct sip_request *req;
   char *cid_num = "";
   char *cid_name = "";
   int callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
   char *privacy = "";
   char *screen = "";
   char *start, *end;

   if (!ast_test_flag(&p->flags[0], SIP_TRUSTRPID))
      return 0;
   req = oreq;
   if (!req)
      req = &p->initreq;
   ast_copy_string(tmp, sip_get_header(req, "Remote-Party-ID"), sizeof(tmp));
   if (ast_strlen_zero(tmp)) {
      return get_pai(p, req);
   }

   /*
    * RPID is not:
    *   rpid = (name-addr / addr-spec) *(SEMI rpi-token)
    * But it is:
    *   rpid = [display-name] LAQUOT addr-spec RAQUOT *(SEMI rpi-token)
    * Ergo, calling parse_name_andor_addr() on it wouldn't be
    * correct because that would allow addr-spec style too.
    */
   start = tmp;
   /* Quoted (note that we're not dealing with escapes properly) */
   if (*start == '"') {
      *start++ = '\0';
      end = strchr(start, '"');
      if (!end)
         return 0;
      *end++ = '\0';
      cid_name = start;
      start = ast_skip_blanks(end);
   /* Unquoted */
   } else {
      cid_name = start;
      start = end = strchr(start, '<');
      if (!start) {
         return 0;
      }
      /* trim blanks if there are any. the mandatory NUL is done below */
      while (--end >= cid_name && *end < 33) {
         *end = '\0';
      }
   }

   if (*start != '<')
      return 0;
   *start++ = '\0';
   end = strchr(start, '@');
   if (!end)
      return 0;
   *end++ = '\0';
   if (strncasecmp(start, "sip:", 4))
      return 0;
   cid_num = start + 4;
   if (global_shrinkcallerid && ast_is_shrinkable_phonenumber(cid_num))
      ast_shrink_phone_number(cid_num);
   start = end;

   end = strchr(start, '>');
   if (!end)
      return 0;
   *end++ = '\0';
   if (*end) {
      start = end;
      if (*start != ';')
         return 0;
      *start++ = '\0';
      while (!ast_strlen_zero(start)) {
         end = strchr(start, ';');
         if (end)
            *end++ = '\0';
         if (!strncasecmp(start, "privacy=", 8))
            privacy = start + 8;
         else if (!strncasecmp(start, "screen=", 7))
            screen = start + 7;
         start = end;
      }

      if (!strcasecmp(privacy, "full")) {
         if (!strcasecmp(screen, "yes"))
            callingpres = AST_PRES_PROHIB_USER_NUMBER_PASSED_SCREEN;
         else if (!strcasecmp(screen, "no"))
            callingpres = AST_PRES_PROHIB_USER_NUMBER_NOT_SCREENED;
      } else {
         if (!strcasecmp(screen, "yes"))
            callingpres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
         else if (!strcasecmp(screen, "no"))
            callingpres = AST_PRES_ALLOWED_USER_NUMBER_NOT_SCREENED;
      }
   }

   /* Only return true if the supplied caller id is different */
   if (!strcasecmp(p->cid_num, cid_num) && !strcasecmp(p->cid_name, cid_name) && p->callingpres == callingpres)
      return 0;

   ast_string_field_set(p, cid_num, cid_num);
   ast_string_field_set(p, cid_name, cid_name);
   p->callingpres = callingpres;

   if (p->owner) {
      ast_set_callerid(p->owner, cid_num, cid_name, NULL);
      ast_channel_caller(p->owner)->id.name.presentation = callingpres;
      ast_channel_caller(p->owner)->id.number.presentation = callingpres;
   }

   return 1;
}

static const char * get_sdp_iterate	(	int *	start,
		struct sip_request *	req,
		const char *	name
	)		[static]

Lookup 'name' in the SDP starting at the 'start' line. Returns the matching line, and 'start' is updated with the next line number.

Definition at line 8138 of file chan_sip.c.

References ast_skip_blanks(), and len().

Referenced by get_ip_and_port_from_sdp(), and process_sdp().

{
   int len = strlen(name);
   const char *line;

   while (*start < (req->sdp_start + req->sdp_count)) {
      line = REQ_OFFSET_TO_STR(req, line[(*start)++]);
      if (!strncasecmp(line, name, len) && line[len] == '=') {
         return ast_skip_blanks(line + len + 1);
      }
   }

   /* if the line was not found, ensure that *start points past the SDP */
   (*start)++;

   return "";
}

static char get_sdp_line	(	int *	start,
		int	stop,
		struct sip_request *	req,
		const char **	value
	)		[static]

Fetches the next valid SDP line between the 'start' line (inclusive) and the 'stop' line (exclusive). Returns the type ('a', 'c', ...) and matching line in reference 'start' is updated with the next line number.

Definition at line 8161 of file chan_sip.c.

References ast_skip_blanks(), and type.

Referenced by process_sdp().

{
   char type = '\0';
   const char *line = NULL;

   if (stop > (req->sdp_start + req->sdp_count)) {
      stop = req->sdp_start + req->sdp_count;
   }

   while (*start < stop) {
      line = REQ_OFFSET_TO_STR(req, line[(*start)++]);
      if (line[1] == '=') {
         type = line[0];
         *value = ast_skip_blanks(line + 2);
         break;
      }
   }

   return type;
}

static char* get_sdp_rtp_profile	(	const struct sip_pvt *	p,
		unsigned int	secure,
		struct ast_rtp_instance *	instance
	)		[static]

Definition at line 13050 of file chan_sip.c.

References ast_rtp_engine_dtls::active, ast_rtp_instance_get_dtls(), and ast_test_flag.

Referenced by add_sdp().

{
   struct ast_rtp_engine_dtls *dtls;

   if ((dtls = ast_rtp_instance_get_dtls(instance)) && dtls->active(instance)) {
      return ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF) ? "UDP/TLS/RTP/SAVPF" : "UDP/TLS/RTP/SAVP";
   } else {
      if (ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
         return secure ? "RTP/SAVPF" : "RTP/AVPF";
      } else {
         return secure ? "RTP/SAVP" : "RTP/AVP";
      }
   }
}

static struct sip_pvt * get_sip_pvt_byid_locked	(	const char *	callid,
		const char *	totag,
		const char *	fromtag
	)		[static, read]

Lock dialog lock and find matching pvt lock.

Returns:

a reference, remember to release it when done

Definition at line 17673 of file chan_sip.c.

References ao2_t_find, ast_channel_trylock, ast_debug, ast_strlen_zero(), ast_test_flag, OBJ_POINTER, sip_cfg, sip_pvt_lock, sip_pvt_unlock, and TRUE.

Referenced by handle_request_invite(), and local_attended_transfer().

{
   struct sip_pvt *sip_pvt_ptr;
   struct sip_pvt tmp_dialog = {
      .callid = callid,
   };

   if (totag) {
      ast_debug(4, "Looking for callid %s (fromtag %s totag %s)\n", callid, fromtag ? fromtag : "<no fromtag>", totag ? totag : "<no totag>");
   }

   /* Search dialogs and find the match */

   sip_pvt_ptr = ao2_t_find(dialogs, &tmp_dialog, OBJ_POINTER, "ao2_find of dialog in dialogs table");
   if (sip_pvt_ptr) {
      /* Go ahead and lock it (and its owner) before returning */
      sip_pvt_lock(sip_pvt_ptr);
      if (sip_cfg.pedanticsipchecking) {
         unsigned char frommismatch = 0, tomismatch = 0;

         if (ast_strlen_zero(fromtag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no from tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }

         if (ast_strlen_zero(totag)) {
            sip_pvt_unlock(sip_pvt_ptr);
            ast_debug(4, "Matched %s call for callid=%s - no to tag specified, pedantic check fails\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid);
            return NULL;
         }
         /* RFC 3891
          * > 3.  User Agent Server Behavior: Receiving a Replaces Header
          * > The Replaces header contains information used to match an existing
          * > SIP dialog (call-id, to-tag, and from-tag).  Upon receiving an INVITE
          * > with a Replaces header, the User Agent (UA) attempts to match this
          * > information with a confirmed or early dialog.  The User Agent Server
          * > (UAS) matches the to-tag and from-tag parameters as if they were tags
          * > present in an incoming request.  In other words, the to-tag parameter
          * > is compared to the local tag, and the from-tag parameter is compared
          * > to the remote tag.
          *
          * Thus, the totag is always compared to the local tag, regardless if
          * this our call is an incoming or outgoing call.
          */
         frommismatch = !!strcmp(fromtag, sip_pvt_ptr->theirtag);
         tomismatch = !!strcmp(totag, sip_pvt_ptr->tag);

                        /* Don't check from if the dialog is not established, due to multi forking the from
                         * can change when the call is not answered yet.
                         */
         if ((frommismatch && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) || tomismatch) {
            sip_pvt_unlock(sip_pvt_ptr);
            if (frommismatch) {
               ast_debug(4, "Matched %s call for callid=%s - pedantic from tag check fails; their tag is %s our tag is %s\n",
                    sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
                    fromtag, sip_pvt_ptr->theirtag);
            }
            if (tomismatch) {
               ast_debug(4, "Matched %s call for callid=%s - pedantic to tag check fails; their tag is %s our tag is %s\n",
                    sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING", sip_pvt_ptr->callid,
                    totag, sip_pvt_ptr->tag);
            }
            return NULL;
         }
      }

      if (totag)
         ast_debug(4, "Matched %s call - their tag is %s Our tag is %s\n",
                 sip_pvt_ptr->outgoing_call == TRUE ? "OUTGOING": "INCOMING",
                 sip_pvt_ptr->theirtag, sip_pvt_ptr->tag);

      /* deadlock avoidance... */
      while (sip_pvt_ptr->owner && ast_channel_trylock(sip_pvt_ptr->owner)) {
         sip_pvt_unlock(sip_pvt_ptr);
         usleep(1);
         sip_pvt_lock(sip_pvt_ptr);
      }
   }
   
   return sip_pvt_ptr;
}

static const char* get_srv_protocol

(

enum sip_transport

t

)

[inline, static]

Return protocol string for srv dns query.

Definition at line 3813 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do(), build_peer(), create_addr(), and transmit_register().

{
   switch (t) {
   case SIP_TRANSPORT_UDP:
      return "udp";
   case SIP_TRANSPORT_WS:
      return "ws";
   case SIP_TRANSPORT_TLS:
   case SIP_TRANSPORT_TCP:
      return "tcp";
   case SIP_TRANSPORT_WSS:
      return "wss";
   }

   return "udp";
}

static const char* get_srv_service

(

enum sip_transport

t

)

[inline, static]

Return service string for srv dns query.

Definition at line 3831 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do(), build_peer(), create_addr(), and transmit_register().

{
   switch (t) {
   case SIP_TRANSPORT_TCP:
   case SIP_TRANSPORT_UDP:
   case SIP_TRANSPORT_WS:
      return "sip";
   case SIP_TRANSPORT_TLS:
   case SIP_TRANSPORT_WSS:
      return "sips";
   }
   return "sip";
}

static const char* get_transport_list

(

unsigned int

transports

)

[inline, static]

Return configuration of transports for a device.

Definition at line 3756 of file chan_sip.c.

References ast_threadstorage_get(), sip_transport_str_buf, and SIP_TRANSPORT_STR_BUFSIZE.

Referenced by _sip_show_peer(), peers_data_provider_get(), and sip_show_settings().

{
   char *buf;

   if (!transports) {
      return "UNKNOWN";
   }

   if (!(buf = ast_threadstorage_get(&sip_transport_str_buf, SIP_TRANSPORT_STR_BUFSIZE))) {
      return "";
   }

   memset(buf, 0, SIP_TRANSPORT_STR_BUFSIZE);

   if (transports & SIP_TRANSPORT_UDP) {
      strncat(buf, "UDP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
   }
   if (transports & SIP_TRANSPORT_TCP) {
      strncat(buf, "TCP,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
   }
   if (transports & SIP_TRANSPORT_TLS) {
      strncat(buf, "TLS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
   }
   if (transports & SIP_TRANSPORT_WS) {
      strncat(buf, "WS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
   }
   if (transports & SIP_TRANSPORT_WSS) {
      strncat(buf, "WSS,", SIP_TRANSPORT_STR_BUFSIZE - strlen(buf));
   }

   /* Remove the trailing ',' if present */
   if (strlen(buf)) {
      buf[strlen(buf) - 1] = 0;
   }

   return buf;
}

static const char* get_transport_pvt

(

struct sip_pvt *

p

)

[inline, static]

Return transport of dialog.

Note:

this is based on a false assumption. We don't always use the outbound proxy for all requests in a dialog. It depends on the "force" parameter. The FIRST request is always sent to the ob proxy.

Todo:

Fix this function to work correctly

Definition at line 3851 of file chan_sip.c.

References set_socket_transport(), and sip_get_transport().

Referenced by __sip_xmit(), and build_via().

{
   if (p->outboundproxy && p->outboundproxy->transport) {
      set_socket_transport(&p->socket, p->outboundproxy->transport);
   }

   return sip_get_transport(p->socket.type);
}

static int get_transport_str2enum

(

const char *

transport

)

[static]

Return int representing a bit field of transport types found in const char *transport.

Definition at line 3728 of file chan_sip.c.

References ast_strlen_zero().

Referenced by __set_address_from_contact(), and parse_register_contact().

{
   int res = 0;

   if (ast_strlen_zero(transport)) {
      return res;
   }

   if (!strcasecmp(transport, "udp")) {
      res |= SIP_TRANSPORT_UDP;
   }
   if (!strcasecmp(transport, "tcp")) {
      res |= SIP_TRANSPORT_TCP;
   }
   if (!strcasecmp(transport, "tls")) {
      res |= SIP_TRANSPORT_TLS;
   }
   if (!strcasecmp(transport, "ws")) {
      res |= SIP_TRANSPORT_WS;
   }
   if (!strcasecmp(transport, "wss")) {
      res |= SIP_TRANSPORT_WSS;
   }

   return res;
}

static const char * gettag	(	const struct sip_request *	req,
		const char *	header,
		char *	tagbuf,
		int	tagbufsize
	)		[static]

Get tag from packet.

Returns:

Returns the pointer to the provided tag buffer, or NULL if the tag was not found.

Definition at line 24545 of file chan_sip.c.

References ast_copy_string(), and sip_get_header().

Referenced by find_call(), handle_incoming(), handle_request_subscribe(), and handle_response().

{
   const char *thetag;

   if (!tagbuf)
      return NULL;
   tagbuf[0] = '\0';    /* reset the buffer */
   thetag = sip_get_header(req, header);
   thetag = strcasestr(thetag, ";tag=");
   if (thetag) {
      thetag += 5;
      ast_copy_string(tagbuf, thetag, tagbufsize);
      return strsep(&tagbuf, ";");
   }
   return NULL;
}

static int handle_cc_notify	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)		[static]

Definition at line 24562 of file chan_sip.c.

References ao2_callback, ao2_ref, ast_cc_monitor_callee_available(), ast_cc_monitor_request_acked(), ast_string_field_set, ast_strlen_zero(), construct_pidf_body(), find_sip_monitor_instance_by_subscription_pvt(), get_content_line(), get_in_brackets(), sip_get_header(), status, transmit_publish(), and transmit_response().

Referenced by handle_request_notify().

{
   struct sip_monitor_instance *monitor_instance = ao2_callback(sip_monitor_instances, 0,
         find_sip_monitor_instance_by_subscription_pvt, pvt);
   const char *status = get_content_line(req, "cc-state", ':');
   struct cc_epa_entry *cc_entry;
   char *uri;

   if (!monitor_instance) {
      transmit_response(pvt, "400 Bad Request", req);
      return -1;
   }

   if (ast_strlen_zero(status)) {
      ao2_ref(monitor_instance, -1);
      transmit_response(pvt, "400 Bad Request", req);
      return -1;
   }

   if (!strcmp(status, "queued")) {
      /* We've been told that we're queued. This is the endpoint's way of telling
       * us that it has accepted our CC request. We need to alert the core of this
       * development
       */
      ast_cc_monitor_request_acked(monitor_instance->core_id, "SIP endpoint %s accepted request", monitor_instance->device_name);
      transmit_response(pvt, "200 OK", req);
      ao2_ref(monitor_instance, -1);
      return 0;
   }

   /* It's open! Yay! */
   uri = get_content_line(req, "cc-URI", ':');
   if (ast_strlen_zero(uri)) {
      uri = get_in_brackets((char *)sip_get_header(req, "From"));
   }

   ast_string_field_set(monitor_instance, notify_uri, uri);
   if (monitor_instance->suspension_entry) {
      cc_entry = monitor_instance->suspension_entry->instance_data;
      if (cc_entry->current_state == CC_CLOSED) {
         /* If we've created a suspension entry and the current state is closed, then that means
          * we got a notice from the CC core earlier to suspend monitoring, but because this particular
          * call leg had not yet notified us that it was ready for recall, it meant that we
          * could not yet send a PUBLISH. Now, however, we can.
          */
         construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body,
               sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
         transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_INITIAL, monitor_instance->notify_uri);
      } else {
         ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
      }
   } else {
      ast_cc_monitor_callee_available(monitor_instance->core_id, "SIP monitored callee has become available");
   }
   ao2_ref(monitor_instance, -1);
   transmit_response(pvt, "200 OK", req);

   return 0;
}

static int handle_cc_subscribe	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Definition at line 27628 of file chan_sip.c.

References ao2_ref, ast_cc_agent_accept_request(), ast_cc_failed(), ast_log(), ast_strlen_zero(), ast_cc_agent::core_id, ast_cc_agent::device_name, find_sip_cc_agent_by_subscribe_uri(), LOG_WARNING, ast_cc_agent::private_data, sip_get_header(), and transmit_response().

Referenced by handle_request_subscribe().

{
   const char *uri = REQ_OFFSET_TO_STR(req, rlpart2);
   char *param_separator;
   struct ast_cc_agent *agent;
   struct sip_cc_agent_pvt *agent_pvt;
   const char *expires_str = sip_get_header(req, "Expires");
   int expires = -1; /* Just need it to be non-zero */

   if (!ast_strlen_zero(expires_str)) {
      sscanf(expires_str, "%30d", &expires);
   }

   if ((param_separator = strchr(uri, ';'))) {
      *param_separator = '\0';
   }

   p->subscribed = CALL_COMPLETION;

   if (!(agent = find_sip_cc_agent_by_subscribe_uri(uri))) {
      if (!expires) {
         /* Typically, if a 0 Expires reaches us and we can't find
          * the corresponding agent, it means that the CC transaction
          * has completed and so the calling side is just trying to
          * clean up its subscription. We'll just respond with a
          * 200 OK and be done with it
          */
         transmit_response(p, "200 OK", req);
         return 0;
      }
      ast_log(LOG_WARNING, "Invalid URI '%s' in CC subscribe\n", uri);
      transmit_response(p, "404 Not Found", req);
      return -1;
   }

   agent_pvt = agent->private_data;

   if (!expires) {
      /* We got sent a SUBSCRIBE and found an agent. This means that CC
       * is being canceled.
       */
      ast_cc_failed(agent->core_id, "CC is being canceled by %s", agent->device_name);
      transmit_response(p, "200 OK", req);
      ao2_ref(agent, -1);
      return 0;
   }

   agent_pvt->subscribe_pvt = dialog_ref(p, "SIP CC agent gains reference to subscription dialog");
   ast_cc_agent_accept_request(agent->core_id, "SIP caller %s has requested CC via SUBSCRIBE",
         agent->device_name);

   /* We don't send a response here. That is done in the agent's ack callback or in the
    * agent destructor, should a failure occur before we have responded
    */
   ao2_ref(agent, -1);
   return 0;
}

static int handle_common_options	(	struct ast_flags *	flags,
		struct ast_flags *	mask,
		struct ast_variable *	v
	)		[static]

Handle flag-type options common to configuration of devices - peers.

Parameters:

flags	array of three struct ast_flags
mask	array of three struct ast_flags
v	linked list of config variables to process

Returns:

non-zero if any config options were handled, zero otherwise

Definition at line 30117 of file chan_sip.c.

References ast_clear_flag, ast_copy_string(), ast_false(), ast_log(), ast_set2_flag, ast_set_flag, ast_true(), ast_variable::lineno, LOG_WARNING, ast_variable::name, set_insecure_flags(), sip_parse_nat_option(), ast_variable::value, and word.

Referenced by build_peer(), and reload_config().

{
   int res = 1;

   if (!strcasecmp(v->name, "trustrpid")) {
      ast_set_flag(&mask[0], SIP_TRUSTRPID);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_TRUSTRPID);
   } else if (!strcasecmp(v->name, "sendrpid")) {
      ast_set_flag(&mask[0], SIP_SENDRPID);
      if (!strcasecmp(v->value, "pai")) {
         ast_set_flag(&flags[0], SIP_SENDRPID_PAI);
      } else if (!strcasecmp(v->value, "rpid")) {
         ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
      } else if (ast_true(v->value)) {
         ast_set_flag(&flags[0], SIP_SENDRPID_RPID);
      }
   } else if (!strcasecmp(v->name, "rpid_update")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RPID_UPDATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_UPDATE);
   } else if (!strcasecmp(v->name, "rpid_immediate")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RPID_IMMEDIATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RPID_IMMEDIATE);
   } else if (!strcasecmp(v->name, "trust_id_outbound")) {
      ast_set_flag(&mask[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
      ast_clear_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND);
      if (!strcasecmp(v->value, "legacy")) {
         ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
      } else if (ast_true(v->value)) {
         ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_YES);
      } else if (ast_false(v->value)) {
         ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_NO);
      } else {
         ast_log(LOG_WARNING, "Unknown trust_id_outbound mode '%s' on line %d, using legacy\n", v->value, v->lineno);
         ast_set_flag(&flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND_LEGACY);
      }
   } else if (!strcasecmp(v->name, "g726nonstandard")) {
      ast_set_flag(&mask[0], SIP_G726_NONSTANDARD);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_G726_NONSTANDARD);
   } else if (!strcasecmp(v->name, "useclientcode")) {
      ast_set_flag(&mask[0], SIP_USECLIENTCODE);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_USECLIENTCODE);
   } else if (!strcasecmp(v->name, "dtmfmode")) {
      ast_set_flag(&mask[0], SIP_DTMF);
      ast_clear_flag(&flags[0], SIP_DTMF);
      if (!strcasecmp(v->value, "inband"))
         ast_set_flag(&flags[0], SIP_DTMF_INBAND);
      else if (!strcasecmp(v->value, "rfc2833"))
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      else if (!strcasecmp(v->value, "info"))
         ast_set_flag(&flags[0], SIP_DTMF_INFO);
      else if (!strcasecmp(v->value, "shortinfo"))
         ast_set_flag(&flags[0], SIP_DTMF_SHORTINFO);
      else if (!strcasecmp(v->value, "auto"))
         ast_set_flag(&flags[0], SIP_DTMF_AUTO);
      else {
         ast_log(LOG_WARNING, "Unknown dtmf mode '%s' on line %d, using rfc2833\n", v->value, v->lineno);
         ast_set_flag(&flags[0], SIP_DTMF_RFC2833);
      }
   } else if (!strcasecmp(v->name, "nat")) {
      sip_parse_nat_option(v->value, mask, flags);
   } else if (!strcasecmp(v->name, "directmedia") || !strcasecmp(v->name, "canreinvite")) {
      ast_set_flag(&mask[0], SIP_REINVITE);
      ast_clear_flag(&flags[0], SIP_REINVITE);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[0], SIP_DIRECT_MEDIA | SIP_DIRECT_MEDIA_NAT);
      } else if (!ast_false(v->value)) {
         char buf[64];
         char *word, *next = buf;

         ast_copy_string(buf, v->value, sizeof(buf));
         while ((word = strsep(&next, ","))) {
            if (!strcasecmp(word, "update")) {
               ast_set_flag(&flags[0], SIP_REINVITE_UPDATE | SIP_DIRECT_MEDIA);
            } else if (!strcasecmp(word, "nonat")) {
               ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
               ast_clear_flag(&flags[0], SIP_DIRECT_MEDIA_NAT);
            } else if (!strcasecmp(word, "outgoing")) {
               ast_set_flag(&flags[0], SIP_DIRECT_MEDIA);
               ast_set_flag(&mask[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
               ast_set_flag(&flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
            } else {
               ast_log(LOG_WARNING, "Unknown directmedia mode '%s' on line %d\n", v->value, v->lineno);
            }
         }
      }
   } else if (!strcasecmp(v->name, "insecure")) {
      ast_set_flag(&mask[0], SIP_INSECURE);
      ast_clear_flag(&flags[0], SIP_INSECURE);
      set_insecure_flags(&flags[0], v->value, v->lineno);   
   } else if (!strcasecmp(v->name, "progressinband")) {
      ast_set_flag(&mask[0], SIP_PROG_INBAND);
      ast_clear_flag(&flags[0], SIP_PROG_INBAND);
      if (ast_true(v->value))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_YES);
      else if (strcasecmp(v->value, "never"))
         ast_set_flag(&flags[0], SIP_PROG_INBAND_NO);
   } else if (!strcasecmp(v->name, "promiscredir")) {
      ast_set_flag(&mask[0], SIP_PROMISCREDIR);
      ast_set2_flag(&flags[0], ast_true(v->value), SIP_PROMISCREDIR);
   } else if (!strcasecmp(v->name, "videosupport")) {
      if (!strcasecmp(v->value, "always")) {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
         ast_set_flag(&flags[1], SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
      } else {
         ast_set_flag(&mask[1], SIP_PAGE2_VIDEOSUPPORT);
         ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_VIDEOSUPPORT);
      }
   } else if (!strcasecmp(v->name, "textsupport")) {
      ast_set_flag(&mask[1], SIP_PAGE2_TEXTSUPPORT);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_TEXTSUPPORT);
      res = 1;
   } else if (!strcasecmp(v->name, "allowoverlap")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWOVERLAP);
      ast_clear_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_YES);
      } else if (!strcasecmp(v->value, "dtmf")){
         ast_set_flag(&flags[1], SIP_PAGE2_ALLOWOVERLAP_DTMF);
      }
   } else if (!strcasecmp(v->name, "allowsubscribe")) {
      ast_set_flag(&mask[1], SIP_PAGE2_ALLOWSUBSCRIBE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_ALLOWSUBSCRIBE);
   } else if (!strcasecmp(v->name, "ignoresdpversion")) {
      ast_set_flag(&mask[1], SIP_PAGE2_IGNORESDPVERSION);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_IGNORESDPVERSION);
   } else if (!strcasecmp(v->name, "faxdetect")) {
      ast_set_flag(&mask[1], SIP_PAGE2_FAX_DETECT);
      if (ast_true(v->value)) {
         ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
      } else if (ast_false(v->value)) {
         ast_clear_flag(&flags[1], SIP_PAGE2_FAX_DETECT_BOTH);
      } else {
         char *buf = ast_strdupa(v->value);
         char *word, *next = buf;

         while ((word = strsep(&next, ","))) {
            if (!strcasecmp(word, "cng")) {
               ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_CNG);
            } else if (!strcasecmp(word, "t38")) {
               ast_set_flag(&flags[1], SIP_PAGE2_FAX_DETECT_T38);
            } else {
               ast_log(LOG_WARNING, "Unknown faxdetect mode '%s' on line %d.\n", word, v->lineno);
            }
         }
      }
   } else if (!strcasecmp(v->name, "rfc2833compensate")) {
      ast_set_flag(&mask[1], SIP_PAGE2_RFC2833_COMPENSATE);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_RFC2833_COMPENSATE);
   } else if (!strcasecmp(v->name, "buggymwi")) {
      ast_set_flag(&mask[1], SIP_PAGE2_BUGGY_MWI);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_BUGGY_MWI);
   } else
      res = 0;

   return res;
}

static int handle_incoming	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		int *	recount,
		int *	nounlock
	)		[static]

Handle incoming SIP requests (methods)

Note:

This is where all incoming requests go first.

called with p and p->owner locked

Definition at line 28169 of file chan_sip.c.

References __get_header(), __sip_ack(), ast_alloca, ast_control_pvt_cause_code::ast_cause, ast_channel_hangupcause_hash_set(), AST_CHANNEL_NAME, ast_channel_name(), AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_SRCCHANGE, ast_copy_string(), ast_debug, ast_log(), ast_queue_control(), ast_queue_control_data(), ast_random(), ast_skip_blanks(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose(), ast_control_pvt_cause_code::chan_name, check_pendings(), ast_control_pvt_cause_code::code, ast_control_pvt_cause_code::emulate_sip_cause, extract_uri(), find_sdp(), gettag(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_message(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_refer(), handle_request_register(), handle_request_subscribe(), handle_request_update(), handle_response(), hangup_sip2cause(), cfsip_methods::id, len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, process_sdp(), pvt_set_needdestroy(), sip_cfg, sip_get_header(), sip_methods, sip_report_security_event(), sip_scheddestroy(), cfsip_methods::text, transmit_response(), transmit_response_reliable(), transmit_response_with_allow(), and transmit_response_with_retry_after().

Referenced by handle_request_do().

{
   /* Called with p->lock held, as well as p->owner->lock if appropriate, keeping things
      relatively static */
   const char *cmd;
   const char *cseq;
   const char *useragent;
   const char *via;
   const char *callid;
   int via_pos = 0;
   uint32_t seqno;
   int len;
   int respid;
   int res = 0;
   const char *e;
   int error = 0;
   int oldmethod = p->method;
   int acked = 0;

   /* RFC 3261 - 8.1.1 A valid SIP request must contain To, From, CSeq, Call-ID and Via.
    * 8.2.6.2 Response must have To, From, Call-ID CSeq, and Via related to the request,
    * so we can check to make sure these fields exist for all requests and responses */
   cseq = sip_get_header(req, "Cseq");
   cmd = REQ_OFFSET_TO_STR(req, header[0]);
   /* Save the via_pos so we can check later that responses only have 1 Via header */
   via = __get_header(req, "Via", &via_pos);
   /* This must exist already because we've called find_call by now */
   callid = sip_get_header(req, "Call-ID");

   /* Must have Cseq */
   if (ast_strlen_zero(cmd) || ast_strlen_zero(cseq) || ast_strlen_zero(via)) {
      ast_log(LOG_ERROR, "Dropping this SIP message with Call-ID '%s', it's incomplete.\n", callid);
      error = 1;
   }
   if (!error && sscanf(cseq, "%30u%n", &seqno, &len) != 1) {
      ast_log(LOG_ERROR, "No seqno in '%s'. Dropping incomplete message.\n", cmd);
      error = 1;
   }
   if (error) {
      if (!p->initreq.headers) { /* New call */
         pvt_set_needdestroy(p, "no headers");
      }
      return -1;
   }
   /* Get the command XXX */

   cmd = REQ_OFFSET_TO_STR(req, rlpart1);
   e = ast_skip_blanks(REQ_OFFSET_TO_STR(req, rlpart2));

   /* Save useragent of the client */
   useragent = sip_get_header(req, "User-Agent");
   if (!ast_strlen_zero(useragent))
      ast_string_field_set(p, useragent, useragent);

   /* Find out SIP method for incoming request */
   if (req->method == SIP_RESPONSE) {  /* Response to our request */
      /* ignore means "don't do anything with it" but still have to
       * respond appropriately.
       * But in this case this is a response already, so we really
       * have nothing to do with this message, and even setting the
       * ignore flag is pointless.
       */
      if (ast_strlen_zero(e)) {
         return 0;
      }
      if (sscanf(e, "%30d %n", &respid, &len) != 1) {
         ast_log(LOG_WARNING, "Invalid response: '%s'\n", e);
         return 0;
      }
      if (respid <= 0) {
         ast_log(LOG_WARNING, "Invalid SIP response code: '%d'\n", respid);
         return 0;
      }
      /* RFC 3261 - 8.1.3.3 If more than one Via header field value is present in a reponse
       * the UAC SHOULD discard the message. This is not perfect, as it will not catch multiple
       * headers joined with a comma. Fixing that would pretty much involve writing a new parser */
      if (!ast_strlen_zero(__get_header(req, "via", &via_pos))) {
         ast_log(LOG_WARNING, "Misrouted SIP response '%s' with Call-ID '%s', too many vias\n", e, callid);
         return 0;
      }
      if (p->ocseq && (p->ocseq < seqno)) {
         ast_debug(1, "Ignoring out of order response %u (expecting %u)\n", seqno, p->ocseq);
         return -1;
      } else {
         if ((respid == 200) || ((respid >= 300) && (respid <= 399))) {
            extract_uri(p, req);
         }

         if (p->owner) {
            struct ast_control_pvt_cause_code *cause_code;
            int data_size = sizeof(*cause_code);
            /* size of the string making up the cause code is "SIP " + cause length */
            data_size += 4 + strlen(REQ_OFFSET_TO_STR(req, rlpart2));
            cause_code = ast_alloca(data_size);
            memset(cause_code, 0, data_size);

            ast_copy_string(cause_code->chan_name, ast_channel_name(p->owner), AST_CHANNEL_NAME);

            snprintf(cause_code->code, data_size - sizeof(*cause_code) + 1, "SIP %s", REQ_OFFSET_TO_STR(req, rlpart2));

            cause_code->ast_cause = hangup_sip2cause(respid);
            if (global_store_sip_cause) {
               cause_code->emulate_sip_cause = 1;
            }

            ast_queue_control_data(p->owner, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
            ast_channel_hangupcause_hash_set(p->owner, cause_code, data_size);
         }

         handle_response(p, respid, e + len, req, seqno);
      }
      return 0;
   }

   /* New SIP request coming in
      (could be new request in existing SIP dialog as well...)
    */
   p->method = req->method;   /* Find out which SIP method they are using */
   ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);

   if (p->icseq && (p->icseq > seqno) ) {
      if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
         ast_debug(2, "Got CANCEL or ACK on INVITE with transactions in between.\n");
      } else {
         ast_debug(1, "Ignoring too old SIP packet packet %u (expecting >= %u)\n", seqno, p->icseq);
         if (req->method == SIP_INVITE) {
            unsigned int ran = (ast_random() % 10) + 1;
            char seconds[4];
            snprintf(seconds, sizeof(seconds), "%u", ran);
            transmit_response_with_retry_after(p, "500 Server error", req, seconds);   /* respond according to RFC 3261 14.2 with Retry-After betwewn 0 and 10 */
         } else if (req->method != SIP_ACK) {
            transmit_response(p, "500 Server error", req);  /* We must respond according to RFC 3261 sec 12.2 */
         }
         return -1;
      }
   } else if (p->icseq &&
         p->icseq == seqno &&
         req->method != SIP_ACK &&
         (p->method != SIP_CANCEL || p->alreadygone)) {
      /* ignore means "don't do anything with it" but still have to
         respond appropriately.  We do this if we receive a repeat of
         the last sequence number  */
      req->ignore = 1;
      ast_debug(3, "Ignoring SIP message because of retransmit (%s Seqno %u, ours %u)\n", sip_methods[p->method].text, p->icseq, seqno);
   }

   /* RFC 3261 section 9. "CANCEL has no effect on a request to which a UAS has
    * already given a final response." */
   if (!p->pendinginvite && (req->method == SIP_CANCEL)) {
      transmit_response(p, "481 Call/Transaction Does Not Exist", req);
      return res;
   }

   if (seqno >= p->icseq)
      /* Next should follow monotonically (but not necessarily
         incrementally -- thanks again to the genius authors of SIP --
         increasing */
      p->icseq = seqno;

   /* Find their tag if we haven't got it */
   if (ast_strlen_zero(p->theirtag)) {
      char tag[128];

      gettag(req, "From", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   }
   snprintf(p->lastmsg, sizeof(p->lastmsg), "Rx: %s", cmd);

   if (sip_cfg.pedanticsipchecking) {
      /* If this is a request packet without a from tag, it's not
         correct according to RFC 3261  */
      /* Check if this a new request in a new dialog with a totag already attached to it,
         RFC 3261 - section 12.2 - and we don't want to mess with recovery  */
      if (!p->initreq.headers && req->has_to_tag) {
         /* If this is a first request and it got a to-tag, it is not for us */
         if (!req->ignore && req->method == SIP_INVITE) {
            /* Just because we think this is a dialog-starting INVITE with a to-tag
             * doesn't mean it actually is. It could be a reinvite for an established, but
             * unknown dialog. In such a case, we need to change our tag to the
             * incoming INVITE's to-tag so that they will recognize the 481 we send and
             * so that we will properly match their incoming ACK.
             */
            char totag[128];
            gettag(req, "To", totag, sizeof(totag));
            ast_string_field_set(p, tag, totag);
            p->pendinginvite = p->icseq;
            transmit_response_reliable(p, "481 Call/Transaction Does Not Exist", req);
            /* Will cease to exist after ACK */
            return res;
         } else if (req->method != SIP_ACK) {
            transmit_response(p, "481 Call/Transaction Does Not Exist", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            return res;
         }
         /* Otherwise, this is an ACK. It will always have a to-tag */
      }
   }

   if (!e && (p->method == SIP_INVITE || p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_NOTIFY || p->method == SIP_PUBLISH)) {
      transmit_response(p, "400 Bad request", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return -1;
   }

   /* Handle various incoming SIP methods in requests */
   switch (p->method) {
   case SIP_OPTIONS:
      res = handle_request_options(p, req, addr, e);
      break;
   case SIP_INVITE:
      res = handle_request_invite(p, req, addr, seqno, recount, e, nounlock);

      if (res < 9) {
         sip_report_security_event(p, req, res);
      }

      switch (res) {
      case INV_REQ_SUCCESS:
         res = 1;
         break;
      case INV_REQ_FAILED:
         res = 0;
         break;
      case INV_REQ_ERROR:
         res = -1;
         break;
      default:
         res = 0;
         break;
      }

      break;
   case SIP_REFER:
      res = handle_request_refer(p, req, seqno, nounlock);
      break;
   case SIP_CANCEL:
      res = handle_request_cancel(p, req);
      break;
   case SIP_BYE:
      res = handle_request_bye(p, req);
      break;
   case SIP_MESSAGE:
      res = handle_request_message(p, req, addr, e);
      break;
   case SIP_PUBLISH:
      res = handle_request_publish(p, req, addr, seqno, e);
      break;
   case SIP_SUBSCRIBE:
      res = handle_request_subscribe(p, req, addr, seqno, e);
      break;
   case SIP_REGISTER:
      res = handle_request_register(p, req, addr, e);
      sip_report_security_event(p, req, res);
      break;
   case SIP_INFO:
      if (req->debug)
         ast_verbose("Receiving INFO!\n");
      if (!req->ignore)
         handle_request_info(p, req);
      else  /* if ignoring, transmit response */
         transmit_response(p, "200 OK", req);
      break;
   case SIP_NOTIFY:
      res = handle_request_notify(p, req, addr, seqno, e);
      break;
   case SIP_UPDATE:
      res = handle_request_update(p, req);
      break;
   case SIP_ACK:
      /* Make sure we don't ignore this */
      if (seqno == p->pendinginvite) {
         p->invitestate = INV_TERMINATED;
         p->pendinginvite = 0;
         acked = __sip_ack(p, seqno, 1 /* response */, 0);
         if (p->owner && find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_NONE)) {
               return -1;
            }
            if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
               ast_queue_control(p->owner, AST_CONTROL_SRCCHANGE);
            }
         }
         check_pendings(p);
      } else if (p->glareinvite == seqno) {
         /* handle ack for the 491 pending sent for glareinvite */
         p->glareinvite = 0;
         acked = __sip_ack(p, seqno, 1, 0);
      }
      if (!acked) {
         /* Got an ACK that did not match anything. Ignore
          * silently and restore previous method */
         p->method = oldmethod;
      }
      if (!p->lastinvite && ast_strlen_zero(p->nonce)) {
         pvt_set_needdestroy(p, "unmatched ACK");
      }
      break;
   default:
      transmit_response_with_allow(p, "501 Method Not Implemented", req, 0);
      ast_log(LOG_NOTICE, "Unknown SIP command '%s' from '%s'\n",
         cmd, ast_sockaddr_stringify(&p->sa));
      /* If this is some new method, and we don't have a call, destroy it now */
      if (!p->initreq.headers) {
         pvt_set_needdestroy(p, "unimplemented method");
      }
      break;
   }
   return res;
}

static int handle_invite_replaces	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		uint32_t	seqno,
		int *	nounlock
	)		[static]

Handle the transfer part of INVITE with a replaces: header, meaning a target pickup or an attended transfer. Used only once. XXX 'ignore' is unused.

Note:

this function is called by handle_request_invite(). Four locks held at the beginning of this function, p, p->owner, p->refer->refer_call and p->refere->refer_call->owner. only p's lock should remain at the end of this function. p's lock as well as the channel p->owner's lock are held by handle_request_do(), we unlock p->owner before the masq. By setting nounlock we are indicating to handle_request_do() that we have already unlocked the owner.

Definition at line 24873 of file chan_sip.c.

References append_history, ast_bridged_channel(), AST_CAUSE_SWITCH_CONGESTION, ast_channel_hangupcause_set(), ast_channel_lock, ast_channel_masquerade(), ast_channel_name(), ast_channel_ref, ast_channel_unlock, ast_channel_unref, ast_debug, ast_do_masquerade(), ast_hangup(), ast_log(), ast_quiet_chan(), ast_set_flag, ast_setstate(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, FALSE, LOG_ERROR, LOG_NOTICE, LOG_WARNING, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), transmit_response(), transmit_response_reliable(), and transmit_response_with_sdp().

Referenced by handle_request_invite().

{
   int earlyreplace = 0;
   int oneleggedreplace = 0;     /* Call with no bridge, propably IVR or voice message */
   struct ast_channel *c = p->owner;   /* Our incoming call */
   struct ast_channel *replacecall = p->refer->refer_call->owner; /* The channel we're about to take over */
   struct ast_channel *targetcall;     /* The bridge to the take-over target */

   /* Check if we're in ring state */
   if (ast_channel_state(replacecall) == AST_STATE_RING)
      earlyreplace = 1;

   /* Check if we have a bridge */
   if (!(targetcall = ast_bridged_channel(replacecall))) {
      /* We have no bridge */
      if (!earlyreplace) {
         ast_debug(2, " Attended transfer attempted to replace call with no bridge (maybe ringing). Channel %s!\n", ast_channel_name(replacecall));
         oneleggedreplace = 1;
      }
   }
   if (targetcall && ast_channel_state(targetcall) == AST_STATE_RINGING)
      ast_debug(4, "SIP transfer: Target channel is in ringing state\n");

   if (targetcall)
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should bridge to channel %s while hanging up channel %s\n", ast_channel_name(targetcall), ast_channel_name(replacecall));
   else
      ast_debug(4, "SIP transfer: Invite Replace incoming channel should replace and hang up channel %s (one call leg)\n", ast_channel_name(replacecall));

   if (req->ignore) {
      ast_log(LOG_NOTICE, "Ignoring this INVITE with replaces in a stupid way.\n");
      /* We should answer something here. If we are here, the
         call we are replacing exists, so an accepted
         can't harm */
      transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);
      /* Do something more clever here */
      if (c) {
         *nounlock = 1;
         ast_channel_unlock(c);
      }
      ast_channel_unlock(replacecall);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   }
   if (!c) {
      /* What to do if no channel ??? */
      ast_log(LOG_ERROR, "Unable to create new channel.  Invite/replace failed.\n");
      transmit_response_reliable(p, "503 Service Unavailable", req);
      append_history(p, "Xfer", "INVITE/Replace Failed. No new channel.");
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_channel_unlock(replacecall);
      sip_pvt_unlock(p->refer->refer_call);
      return 1;
   }
   append_history(p, "Xfer", "INVITE/Replace received");
   /* We have three channels to play with
      channel c: New incoming call
      targetcall: Call from PBX to target
      p->refer->refer_call: SIP pvt dialog from transferer to pbx.
      replacecall: The owner of the previous
      We need to masq C into refer_call to connect to
      targetcall;
      If we are talking to internal audio stream, target call is null.
   */

   /* Fake call progress */
   transmit_response(p, "100 Trying", req);
   ast_setstate(c, AST_STATE_RING);

   /* Masquerade the new call into the referred call to connect to target call
      Targetcall is not touched by the masq */

   /* Answer the incoming call and set channel to UP state */
   transmit_response_with_sdp(p, "200 OK", req, XMIT_RELIABLE, FALSE, FALSE);

   ast_setstate(c, AST_STATE_UP);

   /* Stop music on hold and other generators */
   ast_quiet_chan(replacecall);
   ast_quiet_chan(targetcall);
   ast_debug(4, "Invite/Replaces: preparing to masquerade %s into %s\n", ast_channel_name(c), ast_channel_name(replacecall));

   /* Make sure that the masq does not free our PVT for the old call */
   if (! earlyreplace && ! oneleggedreplace )
      ast_set_flag(&p->refer->refer_call->flags[0], SIP_DEFER_BYE_ON_TRANSFER);  /* Delay hangup */

   /* Prepare the masquerade - if this does not happen, we will be gone */
   if(ast_channel_masquerade(replacecall, c))
      ast_log(LOG_ERROR, "Failed to masquerade C into Replacecall\n");
   else
      ast_debug(4, "Invite/Replaces: Going to masquerade %s into %s\n", ast_channel_name(c), ast_channel_name(replacecall));

   /* C should now be in place of replacecall. all channel locks and pvt locks should be removed
    * before issuing the masq.  Since we are unlocking both the pvt (p) and its owner channel (c)
    * it is possible for channel c to be destroyed on us.  To prevent this, we must give c a reference
    * before any unlocking takes place and remove it only once we are completely done with it */
   ast_channel_ref(c);
   ast_channel_unlock(replacecall);
   ast_channel_unlock(c);
   sip_pvt_unlock(p->refer->refer_call);
   sip_pvt_unlock(p);
   if (ast_do_masquerade(replacecall)) {
      ast_log(LOG_WARNING, "Failed to perform masquerade with INVITE replaces\n");
   }
   if (earlyreplace || oneleggedreplace ) {
      ast_channel_lock(c);
      ast_channel_hangupcause_set(c, AST_CAUSE_SWITCH_CONGESTION);
      ast_channel_unlock(c);
   }

   /* c and c's tech pvt must be unlocked at this point for ast_hangup */
   ast_hangup(c);
   /* this indicates to handle_request_do that the owner channel has already been unlocked */
   *nounlock = 1;
   /* lock PVT structure again after hangup */
   sip_pvt_lock(p);
   ast_channel_unref(c);
   return 0;
}

static int handle_request_bye	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Handle incoming BYE request.

Definition at line 26787 of file chan_sip.c.

References __sip_pretend_ack(), append_history, ARRAY_LEN, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_PROTOCOL_ERROR, ast_channel_lock, ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_trylock, ast_channel_unlock, ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, ast_log(), AST_MAX_USER_FIELD, ast_queue_control(), ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_rtp_instance_get_quality(), ast_rtp_instance_set_stats_vars(), AST_RTP_INSTANCE_STAT_FIELD_QUALITY, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, check_via(), context, ast_channel::context, copy_request(), get_also_info(), LOG_NOTICE, LOG_WARNING, parse_sip_options(), pbx_builtin_setvar_helper(), quality, sip_alreadygone(), sip_cfg, sip_get_header(), sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_queue_hangup_cause(), sip_scheddestroy_final(), stop_media_flows(), stop_session_timer(), cfsip_methods::text, transmit_response(), transmit_response_reliable(), and transmit_response_with_unsupported().

Referenced by handle_incoming().

{
   struct ast_channel *c=NULL;
   int res;
   struct ast_channel *bridged_to;
   const char *required;

   /* If we have an INCOMING invite that we haven't answered, terminate that transaction */
   if (p->pendinginvite && !ast_test_flag(&p->flags[0], SIP_OUTGOING) && !req->ignore) {
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
   }

   __sip_pretend_ack(p);

   p->invitestate = INV_TERMINATED;

   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);
   sip_alreadygone(p);

   /* Get RTCP quality before end of call */
   if (p->do_history || p->owner) {
      char quality_buf[AST_MAX_USER_FIELD], *quality;
      struct ast_channel *bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;

      /* We need to get the lock on bridge because ast_rtp_instance_set_stats_vars will attempt
       * to lock the bridge. This may get hairy...
       */
      while (bridge && ast_channel_trylock(bridge)) {
         ast_channel_unlock(p->owner);
         do {
            /* Can't use DEADLOCK_AVOIDANCE since p is an ao2 object */
            sip_pvt_unlock(p);
            usleep(1);
            sip_pvt_lock(p);
         } while (p->owner && ast_channel_trylock(p->owner));
         bridge = p->owner ? ast_bridged_channel(p->owner) : NULL;
      }


      if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPaudio", "Quality:%s", quality);

            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioJitter", "Quality:%s", quality);
            }
            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioLoss", "Quality:%s", quality);
            }
            if ((quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
               append_history(p, "RTCPaudioRTT", "Quality:%s", quality);
            }
         }

         if (p->owner) {
            ast_rtp_instance_set_stats_vars(p->owner, p->rtp);
         }

      }

      if (bridge) {
         struct sip_pvt *q = ast_channel_tech_pvt(bridge);

         if (IS_SIP_TECH(ast_channel_tech(bridge)) && q && q->rtp) {
            ast_rtp_instance_set_stats_vars(bridge, q->rtp);
         }
         ast_channel_unlock(bridge);
      }

      if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPvideo", "Quality:%s", quality);
         }
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", quality);
         }
      }
      if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
         if (p->do_history) {
            append_history(p, "RTCPtext", "Quality:%s", quality);
         }
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "RTPTEXTQOS", quality);
         }
      }
   }

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   stop_session_timer(p); /* Stop Session-Timer */

   if (!ast_strlen_zero(sip_get_header(req, "Also"))) {
      ast_log(LOG_NOTICE, "Client '%s' using deprecated BYE/Also transfer method.  Ask vendor to support REFER instead\n",
         ast_sockaddr_stringify(&p->recv));
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, sip_cfg.default_context);
      res = get_also_info(p, req);
      if (!res) {
         c = p->owner;
         if (c) {
            bridged_to = ast_bridged_channel(c);
            if (bridged_to) {
               /* Don't actually hangup here... */
               ast_queue_control(c, AST_CONTROL_UNHOLD);
               ast_channel_unlock(c);  /* async_goto can do a masquerade, no locks can be held during a masq */
               ast_async_goto(bridged_to, p->context, p->refer->refer_to, 1);
               ast_channel_lock(c);
            } else
               ast_queue_hangup(p->owner);
         }
      } else {
         ast_log(LOG_WARNING, "Invalid transfer information from '%s'\n", ast_sockaddr_stringify(&p->recv));
         if (p->owner)
            ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
      }
   } else if (p->owner) {
      sip_queue_hangup_cause(p, 0);
      sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
      ast_debug(3, "Received bye, issuing owner hangup\n");
   } else {
      sip_scheddestroy_final(p, DEFAULT_TRANS_TIMEOUT);
      ast_debug(3, "Received bye, no owner, selfdestruct soon.\n");
   }
   ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);

   /* Find out what they require */
   required = sip_get_header(req, "Require");
   if (!ast_strlen_zero(required)) {
      char unsupported[256] = { 0, };
      parse_sip_options(required, unsupported, ARRAY_LEN(unsupported));
      /* If there are any options required that we do not support,
       * then send a 420 with only those unsupported options listed */
      if (!ast_strlen_zero(unsupported)) {
         transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, unsupported);
         ast_log(LOG_WARNING, "Received SIP BYE with unsupported required extension: required:%s unsupported:%s\n", required, unsupported);
      } else {
         transmit_response(p, "200 OK", req);
      }
   } else {
      transmit_response(p, "200 OK", req);
   }

   return 1;
}

static int handle_request_cancel	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Handle incoming CANCEL request.

Definition at line 26719 of file chan_sip.c.

References __sip_pretend_ack(), ast_debug, ast_free, AST_SCHED_DEL, AST_STATE_UP, ast_str_strlen(), ast_test_flag, check_via(), sip_alreadygone(), sip_queue_hangup_cause(), sip_scheddestroy(), stop_media_flows(), transmit_response(), transmit_response_reliable(), UNLINK, and update_call_counter().

Referenced by handle_incoming().

{

   check_via(p, req);
   sip_alreadygone(p);

   if (p->owner && ast_channel_state(p->owner) == AST_STATE_UP) {
      /* This call is up, cancel is ignored, we need a bye */
      transmit_response(p, "200 OK", req);
      ast_debug(1, "Got CANCEL on an answered call. Ignoring... \n");
      return 0;
   }

   /* At this point, we could have cancelled the invite at the same time
      as the other side sends a CANCEL. Our final reply with error code
      might not have been received by the other side before the CANCEL
      was sent, so let's just give up retransmissions and waiting for
      ACK on our error code. The call is hanging up any way. */
   if (p->invitestate == INV_TERMINATED || p->invitestate == INV_COMPLETED) {
      __sip_pretend_ack(p);
   }
   if (p->invitestate != INV_TERMINATED)
      p->invitestate = INV_CANCELLED;

   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD))
      update_call_counter(p, DEC_CALL_LIMIT);

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */
   if (p->owner) {
      sip_queue_hangup_cause(p, 0);
   } else {
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }
   if (ast_str_strlen(p->initreq.data) > 0) {
      struct sip_pkt *pkt, *prev_pkt;
      /* If the CANCEL we are receiving is a retransmission, and we already have scheduled
       * a reliable 487, then we don't want to schedule another one on top of the previous
       * one.
       *
       * As odd as this may sound, we can't rely on the previously-transmitted "reliable"
       * response in this situation. What if we've sent all of our reliable responses
       * already and now all of a sudden, we get this second CANCEL?
       *
       * The only way to do this correctly is to cancel our previously-scheduled reliably-
       * transmitted response and send a new one in its place.
       */
      for (pkt = p->packets, prev_pkt = NULL; pkt; prev_pkt = pkt, pkt = pkt->next) {
         if (pkt->seqno == p->lastinvite && pkt->response_code == 487) {
            AST_SCHED_DEL(sched, pkt->retransid);
            UNLINK(pkt, p->packets, prev_pkt);
            dialog_unref(pkt->owner, "unref packet->owner from dialog");
            if (pkt->data) {
               ast_free(pkt->data);
            }
            ast_free(pkt);
            break;
         }
      }
      transmit_response_reliable(p, "487 Request Terminated", &p->initreq);
      transmit_response(p, "200 OK", req);
      return 1;
   } else {
      transmit_response(p, "481 Call Leg Does Not Exist", req);
      return 0;
   }
}

static int handle_request_do	(	struct sip_request *	req,
		struct ast_sockaddr *	addr
	)		[static]

Handle incoming SIP message - request or response.

This is used for all transports (udp, tcp and tcp/tls)

Definition at line 28529 of file chan_sip.c.

References ao2_t_ref, append_history, ast_callid_threadassoc_add(), ast_callid_threadassoc_remove(), ast_channel_unlock, ast_channel_unref, ast_debug, ast_mutex_lock, ast_mutex_unlock, ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_str_buffer(), ast_str_reset(), ast_str_strlen(), ast_update_use_count(), ast_verbose(), copy_socket_data(), find_call(), find_sip_method(), handle_incoming(), lws2sws(), netlock, parse_request(), sip_cfg, sip_debug_test_addr(), sip_get_header(), sip_get_transport(), sip_pvt_lock_full(), and sip_pvt_unlock.

Referenced by _sip_tcp_helper_thread(), sip_websocket_callback(), and sipsock_read().

{
   struct sip_pvt *p;
   struct ast_channel *owner_chan_ref = NULL;
   int recount = 0;
   int nounlock = 0;

   if (sip_debug_test_addr(addr))   /* Set the debug flag early on packet level */
      req->debug = 1;
   if (sip_cfg.pedanticsipchecking)
      lws2sws(req->data);  /* Fix multiline headers */
   if (req->debug) {
      ast_verbose("\n<--- SIP read from %s:%s --->\n%s\n<------------->\n",
         sip_get_transport(req->socket.type), ast_sockaddr_stringify(addr), ast_str_buffer(req->data));
   }

   if (parse_request(req) == -1) { /* Bad packet, can't parse */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }
   req->method = find_sip_method(REQ_OFFSET_TO_STR(req, rlpart1));

   if (req->debug)
      ast_verbose("--- (%d headers %d lines)%s ---\n", req->headers, req->lines, (req->headers + req->lines == 0) ? " Nat keepalive" : "");

   if (req->headers < 2) { /* Must have at least two headers */
      ast_str_reset(req->data); /* nulling this out is NOT a good idea here. */
      return 1;
   }
   ast_mutex_lock(&netlock);

   /* Find the active SIP dialog or create a new one */
   p = find_call(req, addr, req->method); /* returns p with a reference only. _NOT_ locked*/
   if (p == NULL) {
      ast_debug(1, "Invalid SIP message - rejected , no callid, len %zu\n", ast_str_strlen(req->data));
      ast_mutex_unlock(&netlock);
      return 1;
   }

   if (p->logger_callid) {
      ast_callid_threadassoc_add(p->logger_callid);
   }

   /* Lock both the pvt and the owner if owner is present.  This will
    * not fail. */
   owner_chan_ref = sip_pvt_lock_full(p);

   copy_socket_data(&p->socket, &req->socket);

   ast_sockaddr_copy(&p->recv, addr);

   /* if we have an owner, then this request has been authenticated */
   if (p->owner) {
      req->authenticated = 1;
   }

   if (p->do_history) /* This is a request or response, note what it was for */
      append_history(p, "Rx", "%s / %s / %s", ast_str_buffer(req->data), sip_get_header(req, "CSeq"), REQ_OFFSET_TO_STR(req, rlpart2));

   if (handle_incoming(p, req, addr, &recount, &nounlock) == -1) {
      /* Request failed */
      ast_debug(1, "SIP message could not be handled, bad request: %-70.70s\n", p->callid[0] ? p->callid : "<no callid>");
   }

   if (recount) {
      ast_update_use_count();
   }

   if (p->owner && !nounlock) {
      ast_channel_unlock(p->owner);
   }
   if (owner_chan_ref) {
      ast_channel_unref(owner_chan_ref);
   }
   sip_pvt_unlock(p);
   ao2_t_ref(p, -1, "throw away dialog ptr from find_call at end of routine"); /* p is gone after the return */
   ast_mutex_unlock(&netlock);

   if (p->logger_callid) {
      ast_callid_threadassoc_remove();
   }

   return 1;
}

static void handle_request_info	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Receive SIP INFO Message.

Definition at line 21415 of file chan_sip.c.

References ast_bridged_channel(), ast_cdr_setuserfield(), ast_channel_cdr(), AST_CONTROL_FLASH, AST_CONTROL_VIDUPDATE, ast_debug, ast_find_call_feature(), AST_FRAME_CONTROL, AST_FRAME_DTMF, ast_log(), AST_LOG_WARNING, ast_queue_control(), ast_queue_frame(), ast_rdlock_call_features(), ast_strlen_zero(), ast_test_flag, ast_unlock_call_features(), ast_verbose(), ast_call_feature::exten, f, get_content(), get_content_line(), ast_frame_subclass::integer, ast_frame::len, LOG_ERROR, LOG_WARNING, sip_get_header(), sip_scheddestroy(), ast_frame::subclass, and transmit_response().

Referenced by handle_incoming().

{
   const char *buf = "";
   unsigned int event;
   const char *c = sip_get_header(req, "Content-Type");

   /* Need to check the media/type */
   if (!strcasecmp(c, "application/dtmf-relay") ||
       !strcasecmp(c, "application/vnd.nortelnetworks.digits") ||
       !strcasecmp(c, "application/dtmf")) {
      unsigned int duration = 0;

      if (!p->owner) {  /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      /* If dtmf-relay or vnd.nortelnetworks.digits, parse the signal and duration;
       * otherwise use the body as the signal */
      if (strcasecmp(c, "application/dtmf")) {
         const char *tmp;

         if (ast_strlen_zero(buf = get_content_line(req, "Signal", '='))
            && ast_strlen_zero(buf = get_content_line(req, "d", '='))) {
            ast_log(LOG_WARNING, "Unable to retrieve DTMF signal for INFO message on "
                  "call %s\n", p->callid);
            transmit_response(p, "200 OK", req);
            return;
         }
         if (!ast_strlen_zero((tmp = get_content_line(req, "Duration", '=')))) {
            sscanf(tmp, "%30u", &duration);
         }
      } else {
         /* Type is application/dtmf, simply use what's in the message body */
         buf = get_content(req);
      }

      /* An empty message body requires us to send a 200 OK */
      if (ast_strlen_zero(buf)) {
         transmit_response(p, "200 OK", req);
         return;
      }

      if (!duration) {
         duration = 100; /* 100 ms */
      }

      if (buf[0] == '*') {
         event = 10;
      } else if (buf[0] == '#') {
         event = 11;
      } else if (buf[0] == '!') {
         event = 16;
      } else if ('A' <= buf[0] && buf[0] <= 'D') {
         event = 12 + buf[0] - 'A';
      } else if ('a' <= buf[0] && buf[0] <= 'd') {
         event = 12 + buf[0] - 'a';
      } else if ((sscanf(buf, "%30u", &event) != 1) || event > 16) {
         ast_log(AST_LOG_WARNING, "Unable to convert DTMF event signal code to a valid "
               "value for INFO message on call %s\n", p->callid);
         transmit_response(p, "200 OK", req);
         return;
      }

      if (event == 16) {
         /* send a FLASH event */
         struct ast_frame f = { AST_FRAME_CONTROL, { AST_CONTROL_FLASH, } };
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: FLASH\n");
         }
      } else {
         /* send a DTMF event */
         struct ast_frame f = { AST_FRAME_DTMF, };
         if (event < 10) {
            f.subclass.integer = '0' + event;
         } else if (event == 10) {
            f.subclass.integer = '*';
         } else if (event == 11) {
            f.subclass.integer = '#';
         } else {
            f.subclass.integer = 'A' + (event - 12);
         }
         f.len = duration;
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event received: %c\n", (int) f.subclass.integer);
         }
      }
      transmit_response(p, "200 OK", req);
      return;
   } else if (!strcasecmp(c, "application/media_control+xml")) {
      /* Eh, we'll just assume it's a fast picture update for now */
      if (p->owner) {
         ast_queue_control(p->owner, AST_CONTROL_VIDUPDATE);
      }
      transmit_response(p, "200 OK", req);
      return;
   } else if (!ast_strlen_zero(c = sip_get_header(req, "X-ClientCode"))) {
      /* Client code (from SNOM phone) */
      if (ast_test_flag(&p->flags[0], SIP_USECLIENTCODE)) {
         if (p->owner && ast_channel_cdr(p->owner)) {
            ast_cdr_setuserfield(p->owner, c);
         }
         if (p->owner && ast_bridged_channel(p->owner) && ast_channel_cdr(ast_bridged_channel(p->owner))) {
            ast_cdr_setuserfield(ast_bridged_channel(p->owner), c);
         }
         transmit_response(p, "200 OK", req);
      } else {
         transmit_response(p, "403 Forbidden", req);
      }
      return;
   } else if (!ast_strlen_zero(c = sip_get_header(req, "Record"))) {
      /* INFO messages generated by some phones to start/stop recording
       * on phone calls.
       */

      struct ast_call_feature *feat = NULL;
      int j;
      struct ast_frame f = { AST_FRAME_DTMF, };
      int suppress_warning = 0; /* Supress warning if the feature is blank */

      if (!p->owner) {        /* not a PBX call */
         transmit_response(p, "481 Call leg/transaction does not exist", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }

      /* first, get the feature string, if it exists */
      ast_rdlock_call_features();
      if (p->relatedpeer) {
         if (!strcasecmp(c, "on")) {
            if (ast_strlen_zero(p->relatedpeer->record_on_feature)) {
               suppress_warning = 1;
            } else {
               feat = ast_find_call_feature(p->relatedpeer->record_on_feature);
            }
         } else if (!strcasecmp(c, "off")) {
            if (ast_strlen_zero(p->relatedpeer->record_off_feature)) {
               suppress_warning = 1;
            } else {
               feat = ast_find_call_feature(p->relatedpeer->record_off_feature);
            }
         } else {
            ast_log(LOG_ERROR, "Received INFO requesting to record with invalid value: %s\n", c);
         }
      }
      if (!feat || ast_strlen_zero(feat->exten)) {
         if (!suppress_warning) {
            ast_log(LOG_WARNING, "Recording requested, but no One Touch Monitor registered. (See features.conf)\n");
         }
         /* 403 means that we don't support this feature, so don't request it again */
         transmit_response(p, "403 Forbidden", req);
         ast_unlock_call_features();
         return;
      }
      /* Send the feature code to the PBX as DTMF, just like the handset had sent it */
      f.len = 100;
      for (j=0; j < strlen(feat->exten); j++) {
         f.subclass.integer = feat->exten[j];
         ast_queue_frame(p->owner, &f);
         if (sipdebug) {
            ast_verbose("* DTMF-relay event faked: %c\n", f.subclass.integer);
         }
      }
      ast_unlock_call_features();

      ast_debug(1, "Got a Request to Record the channel, state %s\n", c);
      transmit_response(p, "200 OK", req);
      return;
   } else if (ast_strlen_zero(c = sip_get_header(req, "Content-Length")) || !strcasecmp(c, "0")) {
      /* This is probably just a packet making sure the signalling is still up, just send back a 200 OK */
      transmit_response(p, "200 OK", req);
      return;
   }

   /* Other type of INFO message, not really understood by Asterisk */

   ast_log(LOG_WARNING, "Unable to parse INFO message from %s. Content %s\n", p->callid, buf);
   transmit_response(p, "415 Unsupported media type", req);
   return;
}

static int handle_request_invite	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		uint32_t	seqno,
		int *	recount,
		const char *	e,
		int *	nounlock
	)		[static]

Handle incoming INVITE request.

Note:

If the INVITE has a Replaces header, it is part of an attended transfer. If so, we do not go through the dial plan but try to find the active call and masquerade into it

This is a spiral. What we need to do is to just change the outgoing INVITE so that it now routes to the new Request URI. Since we created the INVITE ourselves that should be all we need to do.

Todo:

XXX This needs to be reviewed. YOu don't change the request URI really, you route the packet correctly instead...

Definition at line 25231 of file chan_sip.c.

References __sip_ack(), append_history, ARRAY_LEN, AST_CAUSE_FAILURE, ast_cc_agent_set_interfaces_chanvar(), ast_channel_hangupcause_set(), ast_channel_name(), ast_channel_queue_connected_line_update(), ast_channel_set_redirecting(), ast_channel_unlock, ast_clear_flag, AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, AST_CONTROL_BUSY, AST_CONTROL_SRCUPDATE, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, ast_copy_string(), ast_debug, ast_format_cap_copy(), ast_format_cap_is_empty(), ast_hangup(), ast_log(), AST_MAX_CONTEXT, AST_MAX_EXTENSION, ast_null_frame, ast_party_connected_line_init(), ast_party_redirecting_free(), ast_party_redirecting_init(), AST_PBX_CALL_LIMIT, AST_PBX_FAILED, ast_pbx_start(), AST_PBX_SUCCESS, ast_pickup_ext(), ast_queue_control(), ast_queue_frame(), ast_rtp_instance_set_alt_remote_address(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_sched_add(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_set_party_id_all(), ast_setstate(), ast_setup_cc_recall_datastore(), ast_skip_blanks(), ast_sockaddr_stringify(), AST_STATE_DOWN, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_uri_sip_user, ast_verbose(), build_contact(), build_route(), change_hold_state(), change_redirecting_information(), check_user_full(), check_via(), connected, context, copy_request(), do_magic_pickup(), exten, ast_channel::exten, extract_uri(), FALSE, find_sdp(), get_destination(), get_ip_and_port_from_sdp(), get_rpid(), get_sip_pvt_byid_locked(), handle_invite_replaces(), handle_request_invite_st(), ast_party_connected_line::id, ast_set_party_connected_line::id, LOG_NOTICE, LOG_WARNING, make_our_tag(), ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_ok_contact(), parse_oli(), parse_sip_options(), ast_party_name::presentation, ast_party_number::presentation, ast_set_party_connected_line::priv, process_sdp(), restart_session_timer(), S_OR, set_pvt_allowed_methods(), sip_alreadygone(), sip_cancel_destroy(), sip_cfg, sip_get_header(), sip_methods, sip_new(), sip_pickup(), sip_pvt_lock, sip_pvt_unlock, sip_ref_peer(), sip_refer_alloc(), sip_scheddestroy(), sip_t38_abort(), sip_unref_peer(), sip_uri_cmp(), ast_party_connected_line::source, start_session_timer(), ast_party_name::str, ast_party_number::str, ast_party_id::tag, cfsip_methods::text, transmit_provisional_response(), transmit_response(), transmit_response_reliable(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), transmit_response_with_unsupported(), TRUE, update_call_counter(), update_redirecting(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_incoming().

{
   int res = INV_REQ_SUCCESS;
   int gotdest;
   const char *p_replaces;
   char *replace_id = NULL;
   int refer_locked = 0;
   const char *required;
   unsigned int required_profile = 0;
   struct ast_channel *c = NULL;    /* New channel */
   struct sip_peer *authpeer = NULL;   /* Matching Peer */
   int reinvite = 0;
   struct ast_party_redirecting redirecting;
   struct ast_set_party_redirecting update_redirecting;

   struct {
      char exten[AST_MAX_EXTENSION];
      char context[AST_MAX_CONTEXT];
   } pickup = {
         .exten = "",
   };

   /* Find out what they support */
   if (!p->sipoptions) {
      const char *supported = sip_get_header(req, "Supported");
      if (!ast_strlen_zero(supported)) {
         p->sipoptions = parse_sip_options(supported, NULL, 0);
      }
   }

   /* Find out what they require */
   required = sip_get_header(req, "Require");
   if (!ast_strlen_zero(required)) {
      char unsupported[256] = { 0, };
      required_profile = parse_sip_options(required, unsupported, ARRAY_LEN(unsupported));

      /* If there are any options required that we do not support,
       * then send a 420 with only those unsupported options listed */
      if (!ast_strlen_zero(unsupported)) {
         transmit_response_with_unsupported(p, "420 Bad extension (unsupported)", req, unsupported);
         ast_log(LOG_WARNING, "Received SIP INVITE with unsupported required extension: required:%s unsupported:%s\n", required, unsupported);
         p->invitestate = INV_COMPLETED;
         if (!p->lastinvite)
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = INV_REQ_ERROR;
         goto request_invite_cleanup;
      }
   }

   /* The option tags may be present in Supported: or Require: headers.
   Include the Require: option tags for further processing as well */
   p->sipoptions |= required_profile;
   p->reqsipoptions = required_profile;

   /* Check if this is a loop */
   if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && p->owner && (p->invitestate != INV_TERMINATED && p->invitestate != INV_CONFIRMED) && ast_channel_state(p->owner) != AST_STATE_UP) {
      /* This is a call to ourself.  Send ourselves an error code and stop
         processing immediately, as SIP really has no good mechanism for
         being able to call yourself */
      /* If pedantic is on, we need to check the tags. If they're different, this is
         in fact a forked call through a SIP proxy somewhere. */
      int different;
      const char *initial_rlpart2 = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
      const char *this_rlpart2 = REQ_OFFSET_TO_STR(req, rlpart2);
      if (sip_cfg.pedanticsipchecking)
         different = sip_uri_cmp(initial_rlpart2, this_rlpart2);
      else
         different = strcmp(initial_rlpart2, this_rlpart2);
      if (!different) {
         transmit_response(p, "482 Loop Detected", req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = INV_REQ_FAILED;
         goto request_invite_cleanup;
      } else {
         /*! This is a spiral. What we need to do is to just change the outgoing INVITE
          * so that it now routes to the new Request URI. Since we created the INVITE ourselves
          * that should be all we need to do.
          *
          * \todo XXX This needs to be reviewed.  YOu don't change the request URI really, you route the packet
          * correctly instead...
          */
         char *uri = ast_strdupa(this_rlpart2);
         char *at = strchr(uri, '@');
         char *peerorhost;
         ast_debug(2, "Potential spiral detected. Original RURI was %s, new RURI is %s\n", initial_rlpart2, this_rlpart2);
         transmit_response(p, "100 Trying", req);
         if (at) {
            *at = '\0';
         }
         /* Parse out "sip:" */
         if ((peerorhost = strchr(uri, ':'))) {
            *peerorhost++ = '\0';
         }
         ast_string_field_set(p, theirtag, NULL);
         /* Treat this as if there were a call forward instead...
          */
         ast_channel_call_forward_set(p->owner, peerorhost);
         ast_queue_control(p->owner, AST_CONTROL_BUSY);
         res = INV_REQ_FAILED;
         goto request_invite_cleanup;
      }
   }

   if (!req->ignore && p->pendinginvite) {
      if (!ast_test_flag(&p->flags[0], SIP_OUTGOING) && (p->invitestate == INV_COMPLETED || p->invitestate == INV_TERMINATED)) {
         /* What do these circumstances mean? We have received an INVITE for an "incoming" dialog for which we
          * have sent a final response. We have not yet received an ACK, though (which is why p->pendinginvite is non-zero).
          * We also know that the INVITE is not a retransmission, because otherwise the "ignore" flag would be set.
          * This means that either we are receiving a reinvite for a terminated dialog, or we are receiving an INVITE with
          * credentials based on one we challenged earlier.
          *
          * The action to take in either case is to treat the INVITE as though it contains an implicit ACK for the previous
          * transaction. Calling __sip_ack will take care of this by clearing the p->pendinginvite and removing the response
          * from the previous transaction from the list of outstanding packets.
          */
         __sip_ack(p, p->pendinginvite, 1, 0);
      } else {
         /* We already have a pending invite. Sorry. You are on hold. */
         p->glareinvite = seqno;
         if (p->rtp && find_sdp(req)) {
            struct ast_sockaddr addr;
            if (get_ip_and_port_from_sdp(req, SDP_AUDIO, &addr)) {
               ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.\n");
            } else {
               ast_rtp_instance_set_alt_remote_address(p->rtp, &addr);
            }
            if (p->vrtp) {
               if (get_ip_and_port_from_sdp(req, SDP_VIDEO, &addr)) {
                  ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Video may not work properly on this call.\n");
               } else {
                  ast_rtp_instance_set_alt_remote_address(p->vrtp, &addr);
               }
            }
         }
         transmit_response_reliable(p, "491 Request Pending", req);
         check_via(p, req);
         ast_debug(1, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);
         /* Don't destroy dialog here */
         res = INV_REQ_FAILED;
         goto request_invite_cleanup;
      }
   }

   p_replaces = sip_get_header(req, "Replaces");
   if (!ast_strlen_zero(p_replaces)) {
      /* We have a replaces header */
      char *ptr;
      char *fromtag = NULL;
      char *totag = NULL;
      char *start, *to;
      int error = 0;

      if (p->owner) {
         ast_debug(3, "INVITE w Replaces on existing call? Refusing action. [%s]\n", p->callid);
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         check_via(p, req);
         copy_request(&p->initreq, req);
         /* Do not destroy existing call */
         res = INV_REQ_ERROR;
         goto request_invite_cleanup;
      }

      if (sipdebug)
         ast_debug(3, "INVITE part of call transfer. Replaces [%s]\n", p_replaces);
      /* Create a buffer we can manipulate */
      replace_id = ast_strdupa(p_replaces);
      ast_uri_decode(replace_id, ast_uri_sip_user);

      if (!p->refer && !sip_refer_alloc(p)) {
         transmit_response_reliable(p, "500 Server Internal Error", req);
         append_history(p, "Xfer", "INVITE/Replace Failed. Out of memory.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         p->invitestate = INV_COMPLETED;
         res = INV_REQ_ERROR;
         check_via(p, req);
         copy_request(&p->initreq, req);
         goto request_invite_cleanup;
      }

      /*  Todo: (When we find phones that support this)
         if the replaces header contains ";early-only"
         we can only replace the call in early
         stage, not after it's up.

         If it's not in early mode, 486 Busy.
      */

      /* Skip leading whitespace */
      replace_id = ast_skip_blanks(replace_id);

      start = replace_id;
      while ( (ptr = strsep(&start, ";")) ) {
         ptr = ast_skip_blanks(ptr); /* XXX maybe unnecessary ? */
         if ( (to = strcasestr(ptr, "to-tag=") ) )
            totag = to + 7;   /* skip the keyword */
         else if ( (to = strcasestr(ptr, "from-tag=") ) ) {
            fromtag = to + 9; /* skip the keyword */
            fromtag = strsep(&fromtag, "&"); /* trim what ? */
         }
      }

      if (sipdebug)
         ast_debug(4, "Invite/replaces: Will use Replace-Call-ID : %s Fromtag: %s Totag: %s\n",
                 replace_id,
                 fromtag ? fromtag : "<no from tag>",
                 totag ? totag : "<no to tag>");

      /* Try to find call that we are replacing.
         If we have a Replaces header, we need to cancel that call if we succeed with this call.
         First we cheat a little and look for a magic call-id from phones that support
         dialog-info+xml so we can do technology independent pickup... */
      if (strncmp(replace_id, "pickup-", 7) == 0) {
         struct sip_pvt *subscription = NULL;
         replace_id += 7; /* Worst case we are looking at \0 */

         if ((subscription = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
            ast_log(LOG_NOTICE, "Unable to find subscription with call-id: %s\n", replace_id);
            transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
            error = 1;
         } else {
            ast_log(LOG_NOTICE, "Trying to pick up %s@%s\n", subscription->exten, subscription->context);
            ast_copy_string(pickup.exten, subscription->exten, sizeof(pickup.exten));
            ast_copy_string(pickup.context, subscription->context, sizeof(pickup.context));
            sip_pvt_unlock(subscription);
            if (subscription->owner) {
               ast_channel_unlock(subscription->owner);
            }
            subscription = dialog_unref(subscription, "unref dialog subscription");
         }
      }

      /* This locks both refer_call pvt and refer_call pvt's owner!!!*/
      if (!error && ast_strlen_zero(pickup.exten) && (p->refer->refer_call = get_sip_pvt_byid_locked(replace_id, totag, fromtag)) == NULL) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existent call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replaces)", req);
         error = 1;
      } else {
         refer_locked = 1;
      }

      /* The matched call is the call from the transferer to Asterisk .
         We want to bridge the bridged part of the call to the
         incoming invite, thus taking over the refered call */

      if (p->refer->refer_call == p) {
         ast_log(LOG_NOTICE, "INVITE with replaces into it's own call id (%s == %s)!\n", replace_id, p->callid);
         transmit_response_reliable(p, "400 Bad request", req);   /* The best way to not not accept the transfer */
         error = 1;
      }

      if (!error && ast_strlen_zero(pickup.exten) && !p->refer->refer_call->owner) {
         /* Oops, someting wrong anyway, no owner, no call */
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-existing call id (%s)!\n", replace_id);
         /* Check for better return code */
         transmit_response_reliable(p, "481 Call Leg Does Not Exist (Replace)", req);
         error = 1;
      }

      if (!error && ast_strlen_zero(pickup.exten) && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_RINGING && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_RING && ast_channel_state(p->refer->refer_call->owner) != AST_STATE_UP) {
         ast_log(LOG_NOTICE, "Supervised transfer attempted to replace non-ringing or active call id (%s)!\n", replace_id);
         transmit_response_reliable(p, "603 Declined (Replaces)", req);
         error = 1;
      }

      if (error) {   /* Give up this dialog */
         append_history(p, "Xfer", "INVITE/Replace Failed.");
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         sip_pvt_unlock(p);
         if (p->refer->refer_call) {
            sip_pvt_unlock(p->refer->refer_call);
            if (p->refer->refer_call->owner) {
               ast_channel_unlock(p->refer->refer_call->owner);
            }
            p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
         }
         refer_locked = 0;
         p->invitestate = INV_COMPLETED;
         res = INV_REQ_ERROR;
         check_via(p, req);
         copy_request(&p->initreq, req);
         goto request_invite_cleanup;
      }
   }

   /* Check if this is an INVITE that sets up a new dialog or
      a re-invite in an existing dialog */

   if (!req->ignore) {
      int newcall = (p->initreq.headers ? TRUE : FALSE);

      if (sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      /* This also counts as a pending invite */
      p->pendinginvite = seqno;
      check_via(p, req);

      copy_request(&p->initreq, req);     /* Save this INVITE as the transaction basis */
      if (sipdebug)
         ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      if (!p->owner) {  /* Not a re-invite */
         if (req->debug)
            ast_verbose("Using INVITE request as basis request - %s\n", p->callid);
         if (newcall)
            append_history(p, "Invite", "New call: %s", p->callid);
         parse_ok_contact(p, req);
      } else { /* Re-invite on existing call */
         ast_clear_flag(&p->flags[0], SIP_OUTGOING);  /* This is now an inbound dialog */
         if (get_rpid(p, req)) {
            struct ast_party_connected_line connected;
            struct ast_set_party_connected_line update_connected;

            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));

            update_connected.id.number = 1;
            connected.id.number.valid = 1;
            connected.id.number.str = (char *) p->cid_num;
            connected.id.number.presentation = p->callingpres;

            update_connected.id.name = 1;
            connected.id.name.valid = 1;
            connected.id.name.str = (char *) p->cid_name;
            connected.id.name.presentation = p->callingpres;

            /* Invalidate any earlier private connected id representation */
            ast_set_party_id_all(&update_connected.priv);

            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         /* Handle SDP here if we already have an owner */
         if (find_sdp(req)) {
            if (process_sdp(p, req, SDP_T38_INITIATE)) {
               if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {
                  /* Asterisk does not yet support any Content-Encoding methods.  Always
                   * attempt to process the sdp, but return a 415 if a Content-Encoding header
                   * was present after processing failed.  */
                  transmit_response_reliable(p, "415 Unsupported Media type", req);
               } else {
                  transmit_response_reliable(p, "488 Not acceptable here", req);
               }
               if (!p->lastinvite)
                  sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
               res = INV_REQ_ERROR;
               goto request_invite_cleanup;
            }
            ast_queue_control(p->owner, AST_CONTROL_SRCUPDATE);
         } else {
            ast_format_cap_copy(p->jointcaps, p->caps);
            ast_debug(1, "Hm....  No sdp for the moment\n");
            /* Some devices signal they want to be put off hold by sending a re-invite
               *without* an SDP, which is supposed to mean "Go back to your state"
               and since they put os on remote hold, we go back to off hold */
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
               ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
               /* Activate a re-invite */
               ast_queue_frame(p->owner, &ast_null_frame);
               change_hold_state(p, req, FALSE, 0);
            }
         }
         if (p->do_history) /* This is a response, note what it was for */
            append_history(p, "ReInv", "Re-invite received");
      }
   } else if (req->debug)
      ast_verbose("Ignoring this INVITE request\n");

   if (!p->lastinvite && !req->ignore && !p->owner) {
      /* This is a new invite */
      /* Handle authentication if this is our first invite */
      int cc_recall_core_id = -1;
      set_pvt_allowed_methods(p, req);
      res = check_user_full(p, req, SIP_INVITE, e, XMIT_RELIABLE, addr, &authpeer);
      if (res == AUTH_CHALLENGE_SENT) {
         p->invitestate = INV_COMPLETED;     /* Needs to restart in another INVITE transaction */
         goto request_invite_cleanup;
      }
      if (res < 0) { /* Something failed in authentication */
         ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", sip_get_header(req, "From"));
         transmit_response_reliable(p, "403 Forbidden", req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         goto request_invite_cleanup;
      }

      /* Successful authentication and peer matching so record the peer related to this pvt (for easy access to peer settings) */
      if (p->relatedpeer) {
         p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
      }
      if (authpeer) {
         p->relatedpeer = sip_ref_peer(authpeer, "setting dialog's relatedpeer pointer");
      }

      req->authenticated = 1;

      /* We have a successful authentication, process the SDP portion if there is one */
      if (find_sdp(req)) {
         if (process_sdp(p, req, SDP_T38_INITIATE)) {
            /* Asterisk does not yet support any Content-Encoding methods.  Always
             * attempt to process the sdp, but return a 415 if a Content-Encoding header
             * was present after processing fails. */
            if (!ast_strlen_zero(sip_get_header(req, "Content-Encoding"))) {
               transmit_response_reliable(p, "415 Unsupported Media type", req);
            } else {
               /* Unacceptable codecs */
               transmit_response_reliable(p, "488 Not acceptable here", req);
            }
            p->invitestate = INV_COMPLETED;
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            ast_debug(1, "No compatible codecs for this SIP call.\n");
            res = INV_REQ_ERROR;
            goto request_invite_cleanup;
         }
      } else { /* No SDP in invite, call control session */
         ast_format_cap_copy(p->jointcaps, p->caps);
         ast_debug(2, "No SDP in Invite, third party call control\n");
      }

      /* Initialize the context if it hasn't been already */
      if (ast_strlen_zero(p->context))
         ast_string_field_set(p, context, sip_cfg.default_context);


      /* Check number of concurrent calls -vs- incoming limit HERE */
      ast_debug(1, "Checking SIP call limits for device %s\n", p->username);
      if ((res = update_call_counter(p, INC_CALL_LIMIT))) {
         if (res < 0) {
            ast_log(LOG_NOTICE, "Failed to place call for device %s, too many calls\n", p->username);
            transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req);
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            p->invitestate = INV_COMPLETED;

            res = AUTH_SESSION_LIMIT;
         }

         goto request_invite_cleanup;
      }
      gotdest = get_destination(p, NULL, &cc_recall_core_id);  /* Get destination right away */
      extract_uri(p, req);       /* Get the Contact URI */
      build_contact(p);       /* Build our contact header */

      if (p->rtp) {
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
      }

      if (!replace_id && (gotdest != SIP_GET_DEST_EXTEN_FOUND)) { /* No matching extension found */
         switch(gotdest) {
         case SIP_GET_DEST_INVALID_URI:
            transmit_response_reliable(p, "416 Unsupported URI scheme", req);
            break;
         case SIP_GET_DEST_EXTEN_MATCHMORE:
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)
               == SIP_PAGE2_ALLOWOVERLAP_YES) {
               transmit_response_reliable(p, "484 Address Incomplete", req);
               break;
            }
            /*
             * XXX We would have to implement collecting more digits in
             * chan_sip for any other schemes of overlap dialing.
             *
             * For SIP_PAGE2_ALLOWOVERLAP_DTMF it is better to do this in
             * the dialplan using the Incomplete application rather than
             * having the channel driver do it.
             */
            /* Fall through */
         case SIP_GET_DEST_EXTEN_NOT_FOUND:
            {
               char *decoded_exten = ast_strdupa(p->exten);
               transmit_response_reliable(p, "404 Not Found", req);
               ast_uri_decode(decoded_exten, ast_uri_sip_user);
               ast_log(LOG_NOTICE, "Call from '%s' (%s) to extension"
                  " '%s' rejected because extension not found in context '%s'.\n",
                  S_OR(p->username, p->peername), ast_sockaddr_stringify(&p->recv), decoded_exten, p->context);
            }
            break;
         case SIP_GET_DEST_REFUSED:
         default:
            transmit_response_reliable(p, "403 Forbidden", req);
         } /* end switch */

         p->invitestate = INV_COMPLETED;
         update_call_counter(p, DEC_CALL_LIMIT);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         res = INV_REQ_FAILED;
         goto request_invite_cleanup;
      } else {
         /* If no extension was specified, use the s one */
         /* Basically for calling to IP/Host name only */
         if (ast_strlen_zero(p->exten))
            ast_string_field_set(p, exten, "s");
         /* Initialize our tag */

         make_our_tag(p);

         if (handle_request_invite_st(p, req, required, reinvite)) {
            p->invitestate = INV_COMPLETED;
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            res = INV_REQ_ERROR;
            goto request_invite_cleanup;
         }

         /* First invitation - create the channel.  Allocation
          * failures are handled below. */

         c = sip_new(p, AST_STATE_DOWN, S_OR(p->peername, NULL), NULL, p->logger_callid);

         if (cc_recall_core_id != -1) {
            ast_setup_cc_recall_datastore(c, cc_recall_core_id);
            ast_cc_agent_set_interfaces_chanvar(c);
         }
         *recount = 1;

         /* Save Record-Route for any later requests we make on this dialogue */
         build_route(p, req, 0, 0);

         if (c) {
            ast_party_redirecting_init(&redirecting);
            memset(&update_redirecting, 0, sizeof(update_redirecting));
            change_redirecting_information(p, req, &redirecting, &update_redirecting,
               FALSE); /*Will return immediately if no Diversion header is present */
            ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
            ast_party_redirecting_free(&redirecting);
         }
      }
   } else {
      ast_party_redirecting_init(&redirecting);
      memset(&update_redirecting, 0, sizeof(update_redirecting));
      if (sipdebug) {
         if (!req->ignore)
            ast_debug(2, "Got a SIP re-invite for call %s\n", p->callid);
         else
            ast_debug(2, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
      }
      if (!req->ignore)
         reinvite = 1;

      if (handle_request_invite_st(p, req, required, reinvite)) {
         p->invitestate = INV_COMPLETED;
         if (!p->lastinvite) {
            sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         }
         res = INV_REQ_ERROR;
         goto request_invite_cleanup;
      }

      c = p->owner;
      change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE); /*Will return immediately if no Diversion header is present */
      if (c) {
         ast_channel_set_redirecting(c, &redirecting, &update_redirecting);
      }
      ast_party_redirecting_free(&redirecting);
   }

   /* Check if OLI/ANI-II is present in From: */
   parse_oli(req, p->owner);

   if (p->stimer->st_active == TRUE) {
      if (reinvite == 0) {
         start_session_timer(p);
      } else {
         restart_session_timer(p);
      }
   }

   if (!req->ignore && p)
      p->lastinvite = seqno;

   if (c && replace_id) {  /* Attended transfer or call pickup - we're the target */
      if (!ast_strlen_zero(pickup.exten)) {
         append_history(p, "Xfer", "INVITE/Replace received");

         /* Let the caller know we're giving it a shot */
         transmit_response(p, "100 Trying", req);
         p->invitestate = INV_PROCEEDING;
         ast_setstate(c, AST_STATE_RING);

         /* Do the pickup itself */
         ast_channel_unlock(c);
         *nounlock = 1;

         /* since p->owner (c) is unlocked, we need to go ahead and unlock pvt for both
          * magic pickup and ast_hangup.  Both of these functions will attempt to lock
          * p->owner again, which can cause a deadlock if we already hold a lock on p.
          * Locking order is, channel then pvt.  Dead lock avoidance must be used if
          * called the other way around. */
         sip_pvt_unlock(p);
         do_magic_pickup(c, pickup.exten, pickup.context);
         /* Now we're either masqueraded or we failed to pickup, in either case we... */
         ast_hangup(c);
         sip_pvt_lock(p); /* pvt is expected to remain locked on return, so re-lock it */

         res = INV_REQ_FAILED;
         goto request_invite_cleanup;
      } else {
         /* Go and take over the target call */
         if (sipdebug)
            ast_debug(4, "Sending this call to the invite/replcaes handler %s\n", p->callid);
         res = handle_invite_replaces(p, req, addr, seqno, nounlock);
         refer_locked = 0;
         goto request_invite_cleanup;
      }
   }


   if (c) { /* We have a call  -either a new call or an old one (RE-INVITE) */
      enum ast_channel_state c_state = ast_channel_state(c);

      if (c_state != AST_STATE_UP && reinvite &&
         (p->invitestate == INV_TERMINATED || p->invitestate == INV_CONFIRMED)) {
         /* If these conditions are true, and the channel is still in the 'ringing'
          * state, then this likely means that we have a situation where the initial
          * INVITE transaction has completed *but* the channel's state has not yet been
          * changed to UP. The reason this could happen is if the reinvite is received
          * on the SIP socket prior to an application calling ast_read on this channel
          * to read the answer frame we earlier queued on it. In this case, the reinvite
          * is completely legitimate so we need to handle this the same as if the channel
          * were already UP. Thus we are purposely falling through to the AST_STATE_UP case.
          */
         c_state = AST_STATE_UP;
      }

      switch(c_state) {
      case AST_STATE_DOWN:
         ast_debug(2, "%s: New call is still down.... Trying... \n", ast_channel_name(c));
         transmit_provisional_response(p, "100 Trying", req, 0);
         p->invitestate = INV_PROCEEDING;
         ast_setstate(c, AST_STATE_RING);
         if (strcmp(p->exten, ast_pickup_ext())) { /* Call to extension -start pbx on this call */
            enum ast_pbx_result result;

            result = ast_pbx_start(c);

            switch(result) {
            case AST_PBX_FAILED:
               ast_log(LOG_WARNING, "Failed to start PBX :(\n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "503 Unavailable", req);
               break;
            case AST_PBX_CALL_LIMIT:
               ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
               p->invitestate = INV_COMPLETED;
               transmit_response_reliable(p, "480 Temporarily Unavailable", req);
               res = AUTH_SESSION_LIMIT;
               break;
            case AST_PBX_SUCCESS:
               /* nothing to do */
               break;
            }

            if (result) {

               /* Unlock locks so ast_hangup can do its magic */
               ast_channel_unlock(c);
               *nounlock = 1;
               sip_pvt_unlock(p);
               ast_hangup(c);
               sip_pvt_lock(p);
               c = NULL;
            }
         } else { /* Pickup call in call group */
            if (sip_pickup(c)) {
               ast_log(LOG_WARNING, "Failed to start Group pickup by %s\n", ast_channel_name(c));
               transmit_response_reliable(p, "480 Temporarily Unavailable", req);
               sip_alreadygone(p);
               ast_channel_hangupcause_set(c, AST_CAUSE_FAILURE);

               /* Unlock locks so ast_hangup can do its magic */
               ast_channel_unlock(c);
               *nounlock = 1;

               p->invitestate = INV_COMPLETED;
               sip_pvt_unlock(p);
               ast_hangup(c);
               sip_pvt_lock(p);
               c = NULL;
            }
         }
         break;
      case AST_STATE_RING:
         transmit_provisional_response(p, "100 Trying", req, 0);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_RINGING:
         transmit_provisional_response(p, "180 Ringing", req, 0);
         p->invitestate = INV_PROCEEDING;
         break;
      case AST_STATE_UP:
         ast_debug(2, "%s: This call is UP.... \n", ast_channel_name(c));

         transmit_response(p, "100 Trying", req);

         if (p->t38.state == T38_PEER_REINVITE) {
            if (p->t38id > -1) {
               /* reset t38 abort timer */
               AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "remove ref for t38id"));
            }
            p->t38id = ast_sched_add(sched, 5000, sip_t38_abort, dialog_ref(p, "passing dialog ptr into sched structure based on t38id for sip_t38_abort."));
         } else if (p->t38.state == T38_ENABLED) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            transmit_response_with_t38_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)));
         } else if ((p->t38.state == T38_DISABLED) || (p->t38.state == T38_REJECTED)) {
            /* If this is not a re-invite or something to ignore - it's critical */
            if (p->srtp && !ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)) {
               ast_log(LOG_WARNING, "Target does not support required crypto\n");
               transmit_response_reliable(p, "488 Not Acceptable Here (crypto)", req);
            } else {
               ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
               transmit_response_with_sdp(p, "200 OK", req, (reinvite ? XMIT_RELIABLE : (req->ignore ?  XMIT_UNRELIABLE : XMIT_CRITICAL)), p->session_modify == TRUE ? FALSE : TRUE, FALSE);
               ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
            }
         }

         p->invitestate = INV_TERMINATED;
         break;
      default:
         ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", ast_channel_state(c));
         transmit_response(p, "100 Trying", req);
         break;
      }
   } else {
      if (p && (p->autokillid == -1)) {
         const char *msg;

         if ((ast_format_cap_is_empty(p->jointcaps)))
            msg = "488 Not Acceptable Here (codec error)";
         else {
            ast_log(LOG_NOTICE, "Unable to create/find SIP channel for this INVITE\n");
            msg = "503 Unavailable";
         }
         transmit_response_reliable(p, msg, req);
         p->invitestate = INV_COMPLETED;
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
   }

request_invite_cleanup:

   if (refer_locked && p->refer && p->refer->refer_call) {
      sip_pvt_unlock(p->refer->refer_call);
      if (p->refer->refer_call->owner) {
         ast_channel_unlock(p->refer->refer_call->owner);
      }
      p->refer->refer_call = dialog_unref(p->refer->refer_call, "unref dialog p->refer->refer_call");
   }
   if (authpeer) {
      authpeer = sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_invite authpeer");
   }

   return res;
}

static int handle_request_invite_st	(	struct sip_pvt *	p,
		struct sip_request *	req,
		const char *	required,
		int	reinvite
	)		[static]

Definition at line 25091 of file chan_sip.c.

References ast_debug, ast_log(), ast_strlen_zero(), FALSE, global_max_se, LOG_ERROR, LOG_WARNING, MAX, parse_minse(), parse_session_expires(), sip_get_header(), sip_st_alloc(), st_get_mode(), st_get_refresher(), st_get_se(), transmit_response_reliable(), transmit_response_with_minse(), transmit_response_with_unsupported(), and TRUE.

Referenced by handle_request_invite().

{
   const char *p_uac_se_hdr;       /* UAC's Session-Expires header string                      */
   const char *p_uac_min_se;       /* UAC's requested Min-SE interval (char string)            */
   int uac_max_se = -1;            /* UAC's Session-Expires in integer format                  */
   int uac_min_se = -1;            /* UAC's Min-SE in integer format                           */
   int st_active = FALSE;          /* Session-Timer on/off boolean                             */
   int st_interval = 0;            /* Session-Timer negotiated refresh interval                */
   enum st_refresher tmp_st_ref = SESSION_TIMER_REFRESHER_AUTO; /* Session-Timer refresher     */
   int dlg_min_se = -1;
   int dlg_max_se = global_max_se;
   int rtn;

   /* Session-Timers */
   if ((p->sipoptions & SIP_OPT_TIMER)) {
      enum st_refresher_param st_ref_param = SESSION_TIMER_REFRESHER_PARAM_UNKNOWN;

      /* The UAC has requested session-timers for this session. Negotiate
      the session refresh interval and who will be the refresher */
      ast_debug(2, "Incoming INVITE with 'timer' option supported\n");

      /* Allocate Session-Timers struct w/in the dialog */
      if (!p->stimer) {
         sip_st_alloc(p);
      }

      /* Parse the Session-Expires header */
      p_uac_se_hdr = sip_get_header(req, "Session-Expires");
      if (!ast_strlen_zero(p_uac_se_hdr)) {
         ast_debug(2, "INVITE also has \"Session-Expires\" header.\n");
         rtn = parse_session_expires(p_uac_se_hdr, &uac_max_se, &st_ref_param);
         tmp_st_ref = (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
         if (rtn != 0) {
            transmit_response_reliable(p, "400 Session-Expires Invalid Syntax", req);
            return -1;
         }
      }

      /* Parse the Min-SE header */
      p_uac_min_se = sip_get_header(req, "Min-SE");
      if (!ast_strlen_zero(p_uac_min_se)) {
         ast_debug(2, "INVITE also has \"Min-SE\" header.\n");
         rtn = parse_minse(p_uac_min_se, &uac_min_se);
         if (rtn != 0) {
            transmit_response_reliable(p, "400 Min-SE Invalid Syntax", req);
            return -1;
         }
      }

      dlg_min_se = st_get_se(p, FALSE);
      switch (st_get_mode(p, 1)) {
      case SESSION_TIMER_MODE_ACCEPT:
      case SESSION_TIMER_MODE_ORIGINATE:
         if (uac_max_se > 0 && uac_max_se < dlg_min_se) {
            transmit_response_with_minse(p, "422 Session Interval Too Small", req, dlg_min_se);
            return -1;
         }

         p->stimer->st_active_peer_ua = TRUE;
         st_active = TRUE;
         if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UNKNOWN) {
            tmp_st_ref = st_get_refresher(p);
         }

         dlg_max_se = st_get_se(p, TRUE);
         if (uac_max_se > 0) {
            if (dlg_max_se >= uac_min_se) {
               st_interval = (uac_max_se < dlg_max_se) ? uac_max_se : dlg_max_se;
            } else {
               st_interval = uac_max_se;
            }
         } else if (uac_min_se > 0) {
            st_interval = MAX(dlg_max_se, uac_min_se);
         } else {
            st_interval = dlg_max_se;
         }
         break;

      case SESSION_TIMER_MODE_REFUSE:
         if (p->reqsipoptions & SIP_OPT_TIMER) {
            transmit_response_with_unsupported(p, "420 Option Disabled", req, required);
            ast_log(LOG_WARNING, "Received SIP INVITE with supported but disabled option: %s\n", required);
            return -1;
         }
         break;

      default:
         ast_log(LOG_ERROR, "Internal Error %d at %s:%d\n", st_get_mode(p, 1), __FILE__, __LINE__);
         break;
      }
   } else {
      /* The UAC did not request session-timers.  Asterisk (UAS), will now decide
      (based on session-timer-mode in sip.conf) whether to run session-timers for
      this session or not. */
      switch (st_get_mode(p, 1)) {
      case SESSION_TIMER_MODE_ORIGINATE:
         st_active = TRUE;
         st_interval = st_get_se(p, TRUE);
         tmp_st_ref = SESSION_TIMER_REFRESHER_US;
         p->stimer->st_active_peer_ua = (p->sipoptions & SIP_OPT_TIMER) ? TRUE : FALSE;
         break;

      default:
         break;
      }
   }

   if (reinvite == 0) {
      /* Session-Timers: Start session refresh timer based on negotiation/config */
      if (st_active == TRUE) {
         p->stimer->st_active = TRUE;
         p->stimer->st_interval = st_interval;
         p->stimer->st_ref = tmp_st_ref;
      }
   } else {
      if (p->stimer->st_active == TRUE) {
         /* Session-Timers:  A re-invite request sent within a dialog will serve as
         a refresh request, no matter whether the re-invite was sent for refreshing
         the session or modifying it.*/
         ast_debug (2, "Restarting session-timers on a refresh - %s\n", p->callid);

         /* The UAC may be adjusting the session-timers mid-session */
         if (st_interval > 0) {
            p->stimer->st_interval = st_interval;
            p->stimer->st_ref      = tmp_st_ref;
         }
      }
   }

   return 0;
}

static int handle_request_message	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const char *	e
	)		[static]

Handle incoming MESSAGE request.

Definition at line 26935 of file chan_sip.c.

References ast_verbose(), receive_message(), and transmit_response().

Referenced by handle_incoming().

{
   if (!req->ignore) {
      if (req->debug)
         ast_verbose("Receiving message!\n");
      receive_message(p, req, addr, e);
   } else
      transmit_response(p, "202 Accepted", req);
   return 1;
}

static int handle_request_notify	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		uint32_t	seqno,
		const char *	e
	)		[static]

Handle incoming notifications.

Definition at line 24623 of file chan_sip.c.

References AST_CONTROL_TRANSFER, ast_debug, AST_EVENT_IE_CONTEXT, AST_EVENT_IE_END, AST_EVENT_IE_MAILBOX, AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_STR, AST_EVENT_IE_PLTYPE_UINT, AST_EVENT_MWI, ast_event_new(), ast_event_queue_and_cache(), ast_log(), ast_queue_control_data(), ast_skip_blanks(), ast_strlen_zero(), AST_TRANSFER_FAILED, AST_TRANSFER_SUCCESS, FALSE, get_content(), get_content_line(), handle_cc_notify(), LOG_NOTICE, LOG_WARNING, mailbox, sip_find_peer(), sip_get_header(), sip_scheddestroy(), sip_unref_peer(), transmit_response(), and TRUE.

Referenced by handle_incoming().

{
   /* This is mostly a skeleton for future improvements */
   /* Mostly created to return proper answers on notifications on outbound REFER's */
   int res = 0;
   const char *event = sip_get_header(req, "Event");
   char *sep;

   if( (sep = strchr(event, ';')) ) {  /* XXX bug here - overwriting string ? */
      *sep++ = '\0';
   }

   if (sipdebug)
      ast_debug(2, "Got NOTIFY Event: %s\n", event);

   if (!strcmp(event, "refer")) {
      /* Save nesting depth for now, since there might be other events we will
         support in the future */

      /* Handle REFER notifications */
      char *buf, *cmd, *code;
      int respcode;
      int success = TRUE;

      /* EventID for each transfer... EventID is basically the REFER cseq

       We are getting notifications on a call that we transferred
       We should hangup when we are getting a 200 OK in a sipfrag
       Check if we have an owner of this event */

      /* Check the content type */
      if (strncasecmp(sip_get_header(req, "Content-Type"), "message/sipfrag", strlen("message/sipfrag"))) {
         /* We need a sipfrag */
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /* Get the text of the attachment */
      if (ast_strlen_zero(buf = get_content(req))) {
         ast_log(LOG_WARNING, "Unable to retrieve attachment from NOTIFY %s\n", p->callid);
         transmit_response(p, "400 Bad request", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return -1;
      }

      /*
      From the RFC...
      A minimal, but complete, implementation can respond with a single
      NOTIFY containing either the body:
         SIP/2.0 100 Trying
      
      if the subscription is pending, the body:
         SIP/2.0 200 OK
      if the reference was successful, the body:
         SIP/2.0 503 Service Unavailable
      if the reference failed, or the body:
         SIP/2.0 603 Declined

      if the REFER request was accepted before approval to follow the
      reference could be obtained and that approval was subsequently denied
      (see Section 2.4.7).
      
      If there are several REFERs in the same dialog, we need to
      match the ID of the event header...
      */
      ast_debug(3, "* SIP Transfer NOTIFY Attachment: \n---%s\n---\n", buf);
      cmd = ast_skip_blanks(buf);
      code = cmd;
      /* We are at SIP/2.0 */
      while(*code && (*code > 32)) {   /* Search white space */
         code++;
      }
      *code++ = '\0';
      code = ast_skip_blanks(code);
      sep = code;
      sep++;
      while(*sep && (*sep > 32)) {  /* Search white space */
         sep++;
      }
      *sep++ = '\0';       /* Response string */
      respcode = atoi(code);
      switch (respcode) {
      case 200:   /* OK: The new call is up, hangup this call */
         /* Hangup the call that we are replacing */
         break;
      case 301: /* Moved permenantly */
      case 302: /* Moved temporarily */
         /* Do we get the header in the packet in this case? */
         success = FALSE;
         break;
      case 503:   /* Service Unavailable: The new call failed */
      case 603:   /* Declined: Not accepted */
            /* Cancel transfer, continue the current call */
         success = FALSE;
         break;
      case 0:     /* Parse error */
            /* Cancel transfer, continue the current call */
         ast_log(LOG_NOTICE, "Error parsing sipfrag in NOTIFY in response to REFER.\n");
         success = FALSE;
         break;
      default:
         if (respcode < 200) {
            /* ignore provisional responses */
            success = -1;
         } else {
            ast_log(LOG_NOTICE, "Got unknown code '%d' in NOTIFY in response to REFER.\n", respcode);
            success = FALSE;
         }
         break;
      }
      if (success == FALSE) {
         ast_log(LOG_NOTICE, "Transfer failed. Sorry. Nothing further to do with this call\n");
      }

      if (p->owner && success != -1) {
         enum ast_control_transfer message = success ? AST_TRANSFER_SUCCESS : AST_TRANSFER_FAILED;
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      /* Confirm that we received this packet */
      transmit_response(p, "200 OK", req);
   } else if (!strcmp(event, "message-summary")) {
      const char *mailbox = NULL;
      char *c = ast_strdupa(get_content_line(req, "Voice-Message", ':'));

      if (!p->mwi) {
         struct sip_peer *peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, FALSE, p->socket.type);

         if (peer) {
            mailbox = ast_strdupa(peer->unsolicited_mailbox);
            sip_unref_peer(peer, "removing unsolicited mwi ref");
         }
      } else {
         mailbox = p->mwi->mailbox;
      }

      if (!ast_strlen_zero(mailbox) && !ast_strlen_zero(c)) {
         char *old = strsep(&c, " ");
         char *new = strsep(&old, "/");
         struct ast_event *event;

         if ((event = ast_event_new(AST_EVENT_MWI,
                     AST_EVENT_IE_MAILBOX, AST_EVENT_IE_PLTYPE_STR, mailbox,
                     AST_EVENT_IE_CONTEXT, AST_EVENT_IE_PLTYPE_STR, "SIP_Remote",
                     AST_EVENT_IE_NEWMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(new),
                     AST_EVENT_IE_OLDMSGS, AST_EVENT_IE_PLTYPE_UINT, atoi(old),
                     AST_EVENT_IE_END))) {
            ast_event_queue_and_cache(event);
         }
         transmit_response(p, "200 OK", req);
      } else {
         transmit_response(p, "489 Bad event", req);
         res = -1;
      }
   } else if (!strcmp(event, "keep-alive")) {
       /* Used by Sipura/Linksys for NAT pinhole,
        * just confirm that we received the packet. */
      transmit_response(p, "200 OK", req);
   } else if (!strcmp(event, "call-completion")) {
      res = handle_cc_notify(p, req);
   } else {
      /* We don't understand this event. */
      transmit_response(p, "489 Bad event", req);
      res = -1;
   }

   if (!p->lastinvite)
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return res;
}

static int handle_request_options	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const char *	e
	)		[static]

Handle incoming OPTIONS request An OPTIONS request should be answered like an INVITE from the same UA, including SDP.

Definition at line 24798 of file chan_sip.c.

References ast_log(), ast_shutting_down(), ast_string_field_set, ast_strlen_zero(), build_contact(), check_user(), context, copy_request(), get_destination(), LOG_NOTICE, set_pvt_allowed_methods(), sip_cfg, sip_get_header(), sip_scheddestroy(), transmit_response(), and transmit_response_with_allow().

Referenced by handle_incoming().

{
   const char *msg;
   enum sip_get_dest_result gotdest;
   int res;

   if (p->lastinvite) {
      /* if this is a request in an active dialog, just confirm that the dialog exists. */
      transmit_response_with_allow(p, "200 OK", req, 0);
      return 0;
   }

   if (sip_cfg.auth_options_requests) {
      /* Do authentication if this OPTIONS request began the dialog */
      copy_request(&p->initreq, req);
      set_pvt_allowed_methods(p, req);
      res = check_user(p, req, SIP_OPTIONS, e, XMIT_UNRELIABLE, addr);
      if (res == AUTH_CHALLENGE_SENT) {
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      }
      if (res < 0) { /* Something failed in authentication */
         ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", sip_get_header(req, "From"));
         transmit_response(p, "403 Forbidden", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return 0;
      }
   }

   /* must go through authentication before getting here */
   gotdest = get_destination(p, req, NULL);
   build_contact(p);

   if (ast_strlen_zero(p->context))
      ast_string_field_set(p, context, sip_cfg.default_context);

   if (ast_shutting_down()) {
      msg = "503 Unavailable";
   } else {
      msg = "404 Not Found";
      switch (gotdest) {
      case SIP_GET_DEST_INVALID_URI:
         msg = "416 Unsupported URI scheme";
         break;
      case SIP_GET_DEST_EXTEN_MATCHMORE:
      case SIP_GET_DEST_REFUSED:
      case SIP_GET_DEST_EXTEN_NOT_FOUND:
         //msg = "404 Not Found";
         break;
      case SIP_GET_DEST_EXTEN_FOUND:
         msg = "200 OK";
         break;
      }
   }
   transmit_response_with_allow(p, msg, req, 0);

   /* Destroy if this OPTIONS was the opening request, but not if
      it's in the middle of a normal call flow. */
   sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   return 0;
}

static int handle_request_publish	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const uint32_t	seqno,
		const char *	uri
	)		[static]

Definition at line 27525 of file chan_sip.c.

References __sip_ack(), ast_log(), ast_string_field_set, ast_strlen_zero(), check_user(), determine_sip_publish_type(), get_esc(), handle_sip_publish_initial(), handle_sip_publish_modify(), handle_sip_publish_refresh(), handle_sip_publish_remove(), LOG_NOTICE, max_expiry, pvt_set_needdestroy(), sip_get_header(), sip_scheddestroy(), transmit_response(), and transmit_response_with_minexpires().

Referenced by handle_incoming().

{
   const char *etag = sip_get_header(req, "SIP-If-Match");
   const char *event = sip_get_header(req, "Event");
   struct event_state_compositor *esc;
   enum sip_publish_type publish_type;
   const char *expires_str = sip_get_header(req, "Expires");
   int expires_int;
   int auth_result;
   int handler_result = -1;

   if (ast_strlen_zero(event)) {
      transmit_response(p, "489 Bad Event", req);
      pvt_set_needdestroy(p, "missing Event: header");
      return -1;
   }

   if (!(esc = get_esc(event))) {
      transmit_response(p, "489 Bad Event", req);
      pvt_set_needdestroy(p, "unknown event package in publish");
      return -1;
   }

   auth_result = check_user(p, req, SIP_PUBLISH, uri, XMIT_UNRELIABLE, addr);
   if (auth_result == AUTH_CHALLENGE_SENT) {
      p->lastinvite = seqno;
      return 0;
   } else if (auth_result < 0) {
      ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", sip_get_header(req, "From"));
      transmit_response(p, "403 Forbidden", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_string_field_set(p, theirtag, NULL);
      return 0;
   } else if (auth_result == AUTH_SUCCESSFUL && p->lastinvite) {
      /* We need to stop retransmitting the 401 */
      __sip_ack(p, p->lastinvite, 1, 0);
   }

   publish_type = determine_sip_publish_type(req, event, etag, expires_str, &expires_int);

   if (expires_int > max_expiry) {
      expires_int = max_expiry;
   } else if (expires_int < min_expiry && expires_int > 0) {
      transmit_response_with_minexpires(p, "423 Interval too small", req, min_expiry);
      pvt_set_needdestroy(p, "Expires is less that the min expires allowed.");
      return 0;
   }
   p->expiry = expires_int;

   /* It is the responsibility of these handlers to formulate any response
    * sent for a PUBLISH
    */
   switch (publish_type) {
   case SIP_PUBLISH_UNKNOWN:
      transmit_response(p, "400 Bad Request", req);
      break;
   case SIP_PUBLISH_INITIAL:
      handler_result = handle_sip_publish_initial(p, req, esc, expires_int);
      break;
   case SIP_PUBLISH_REFRESH:
      handler_result = handle_sip_publish_refresh(p, req, esc, etag, expires_int);
      break;
   case SIP_PUBLISH_MODIFY:
      handler_result = handle_sip_publish_modify(p, req, esc, etag, expires_int);
      break;
   case SIP_PUBLISH_REMOVE:
      handler_result = handle_sip_publish_remove(p, req, esc, etag);
      break;
   default:
      transmit_response(p, "400 Impossible Condition", req);
      break;
   }
   if (!handler_result && p->expiry > 0) {
      sip_scheddestroy(p, (p->expiry + 10) * 1000);
   } else {
      pvt_set_needdestroy(p, "forcing expiration");
   }

   return handler_result;
}

static int handle_request_refer	(	struct sip_pvt *	p,
		struct sip_request *	req,
		uint32_t	seqno,
		int *	nounlock
	)		[static]

Chan1: Call between asterisk and transferer Chan2: Call between asterisk and transferee

Definition at line 26306 of file chan_sip.c.

References append_history, ast_async_goto(), ast_bridged_channel(), AST_CAUSE_NORMAL_CLEARING, AST_CEL_ATTENDEDTRANSFER, AST_CEL_BLINDTRANSFER, ast_cel_report_event(), ast_channel_hangupcause_set(), ast_channel_lock, ast_channel_name(), ast_channel_ref, ast_channel_uniqueid(), ast_channel_unlock, ast_channel_unref, ast_channel_update_redirecting(), ast_clear_flag, AST_CONTROL_UNHOLD, ast_debug, ast_indicate(), AST_LIST_EMPTY, ast_manager_event_multichan, ast_parking_ext_valid(), ast_party_redirecting_free(), ast_party_redirecting_init(), ast_queue_control(), ast_set_flag, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verbose(), change_redirecting_information(), check_sip_domain(), context, EVENT_FLAG_CALL, FALSE, get_refer_info(), local_attended_transfer(), pbx_builtin_setvar_helper(), pvt_set_needdestroy(), sip_alreadygone(), sip_cfg, sip_park(), sip_pvt_lock, sip_pvt_unlock, sip_refer_alloc(), transmit_notify_with_sipfrag(), transmit_response(), TRUE, and update_redirecting().

Referenced by handle_incoming().

{
   /*!
    * Chan1: Call between asterisk and transferer
    * Chan2: Call between asterisk and transferee
    */
   struct sip_dual current = { 0, };
   struct ast_channel *chans[2] = { 0, };
   char *refer_to = NULL;
   char *refer_to_domain = NULL;
   char *refer_to_context = NULL;
   char *referred_by = NULL;
   char *callid = NULL;
   int localtransfer = 0;
   int attendedtransfer = 0;
   int res = 0;
   struct ast_party_redirecting redirecting;
   struct ast_set_party_redirecting update_redirecting;

   if (req->debug) {
      ast_verbose("Call %s got a SIP call transfer from %s: (REFER)!\n",
         p->callid,
         ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "callee" : "caller");
   }

   if (!p->owner) {
      /* This is a REFER outside of an existing SIP dialog */
      /* We can't handle that, so decline it */
      ast_debug(3, "Call %s: Declined REFER, outside of dialog...\n", p->callid);
      transmit_response(p, "603 Declined (No dialog)", req);
      if (!req->ignore) {
         append_history(p, "Xfer", "Refer failed. Outside of dialog.");
         sip_alreadygone(p);
         pvt_set_needdestroy(p, "outside of dialog");
      }
      res = 0;
      goto handle_refer_cleanup;
   }

   /* Check if transfer is allowed from this device */
   if (p->allowtransfer == TRANSFER_CLOSED ) {
      /* Transfer not allowed, decline */
      transmit_response(p, "603 Declined (policy)", req);
      append_history(p, "Xfer", "Refer failed. Allowtransfer == closed.");
      /* Do not destroy SIP session */
      res = 0;
      goto handle_refer_cleanup;
   }

   if (!req->ignore && ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
      /* Already have a pending REFER */
      transmit_response(p, "491 Request pending", req);
      append_history(p, "Xfer", "Refer failed. Request pending.");
      res = 0;
      goto handle_refer_cleanup;
   }

   /* Allocate memory for call transfer data */
   if (!p->refer && !sip_refer_alloc(p)) {
      transmit_response(p, "500 Internal Server Error", req);
      append_history(p, "Xfer", "Refer failed. Memory allocation error.");
      res = -3;
      goto handle_refer_cleanup;
   }

   res = get_refer_info(p, req); /* Extract headers */

   p->refer->status = REFER_SENT;

   if (res != 0) {
      switch (res) {
      case -2: /* Syntax error */
         transmit_response(p, "400 Bad Request (Refer-to missing)", req);
         append_history(p, "Xfer", "Refer failed. Refer-to missing.");
         if (req->debug) {
            ast_debug(1, "SIP transfer to black hole can't be handled (no refer-to: )\n");
         }
         break;
      case -3:
         transmit_response(p, "603 Declined (Non sip: uri)", req);
         append_history(p, "Xfer", "Refer failed. Non SIP uri");
         if (req->debug) {
            ast_debug(1, "SIP transfer to non-SIP uri denied\n");
         }
         break;
      default:
         /* Refer-to extension not found, fake a failed transfer */
         transmit_response(p, "202 Accepted", req);
         append_history(p, "Xfer", "Refer failed. Bad extension.");
         transmit_notify_with_sipfrag(p, seqno, "404 Not found", TRUE);
         ast_clear_flag(&p->flags[0], SIP_GOTREFER);
         if (req->debug) {
            ast_debug(1, "SIP transfer to bad extension: %s\n", p->refer->refer_to);
         }
         break;
      }
      res = 0;
      goto handle_refer_cleanup;
   }
   if (ast_strlen_zero(p->context)) {
      ast_string_field_set(p, context, sip_cfg.default_context);
   }

   /* If we do not support SIP domains, all transfers are local */
   if (sip_cfg.allow_external_domains && check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      p->refer->localtransfer = 1;
      if (sipdebug) {
         ast_debug(3, "This SIP transfer is local : %s\n", p->refer->refer_to_domain);
      }
   } else if (AST_LIST_EMPTY(&domain_list) || check_sip_domain(p->refer->refer_to_domain, NULL, 0)) {
      /* This PBX doesn't bother with SIP domains or domain is local, so this transfer is local */
      p->refer->localtransfer = 1;
   } else if (sipdebug) {
      ast_debug(3, "This SIP transfer is to a remote SIP extension (remote domain %s)\n", p->refer->refer_to_domain);
   }

   /* Is this a repeat of a current request? Ignore it */
   /* Don't know what else to do right now. */
   if (req->ignore) {
      goto handle_refer_cleanup;
   }

   /* If this is a blind transfer, we have the following
   channels to work with:
   - chan1, chan2: The current call between transferer and transferee (2 channels)
   - target_channel: A new call from the transferee to the target (1 channel)
   We need to stay tuned to what happens in order to be able
   to bring back the call to the transferer */

   /* If this is a attended transfer, we should have all call legs within reach:
   - chan1, chan2: The call between the transferer and transferee (2 channels)
   - target_channel, targetcall_pvt: The call between the transferer and the target (2 channels)
   We want to bridge chan2 with targetcall_pvt!
   
   The replaces call id in the refer message points
   to the call leg between Asterisk and the transferer.
   So we need to connect the target and the transferee channel
   and hangup the two other channels silently
   
   If the target is non-local, the call ID could be on a remote
   machine and we need to send an INVITE with replaces to the
   target. We basically handle this as a blind transfer
   and let the sip_call function catch that we need replaces
   header in the INVITE.
   */

   /* Get the transferer's channel */
   chans[0] = current.chan1 = p->owner;

   /* Find the other part of the bridge (2) - transferee */
   chans[1] = current.chan2 = ast_bridged_channel(current.chan1);

   ast_channel_ref(current.chan1);
   if (current.chan2) {
      ast_channel_ref(current.chan2);
   }

   if (sipdebug) {
      ast_debug(3, "SIP %s transfer: Transferer channel %s, transferee channel %s\n",
         p->refer->attendedtransfer ? "attended" : "blind",
         ast_channel_name(current.chan1),
         current.chan2 ? ast_channel_name(current.chan2) : "<none>");
   }

   if (!current.chan2 && !p->refer->attendedtransfer) {
      /* No bridged channel, propably IVR or echo or similar... */
      /* Guess we should masquerade or something here */
      /* Until we figure it out, refuse transfer of such calls */
      if (sipdebug) {
         ast_debug(3, "Refused SIP transfer on non-bridged channel.\n");
      }
      p->refer->status = REFER_FAILED;
      append_history(p, "Xfer", "Refer failed. Non-bridged channel.");
      transmit_response(p, "603 Declined", req);
      res = -1;
      goto handle_refer_cleanup;
   }

   if (current.chan2) {
      if (sipdebug) {
         ast_debug(4, "Got SIP transfer, applying to bridged peer '%s'\n", ast_channel_name(current.chan2));
      }
      ast_queue_control(current.chan1, AST_CONTROL_UNHOLD);
   }

   ast_set_flag(&p->flags[0], SIP_GOTREFER);

   /* From here on failures will be indicated with NOTIFY requests */
   transmit_response(p, "202 Accepted", req);

   /* Attended transfer: Find all call legs and bridge transferee with target*/
   if (p->refer->attendedtransfer) {
      /* both p and p->owner _MUST_ be locked while calling local_attended_transfer */
      if ((res = local_attended_transfer(p, &current, req, seqno, nounlock))) {
         goto handle_refer_cleanup; /* We're done with the transfer */
      }
      /* Fall through for remote transfers that we did not find locally */
      if (sipdebug) {
         ast_debug(4, "SIP attended transfer: Still not our call - generating INVITE with replaces\n");
      }
      /* Fallthrough if we can't find the call leg internally */
   }

   /* Copy data we can not safely access after letting the pvt lock go. */
   refer_to = ast_strdupa(p->refer->refer_to);
   refer_to_domain = ast_strdupa(p->refer->refer_to_domain);
   refer_to_context = ast_strdupa(p->refer->refer_to_context);
   referred_by = ast_strdupa(p->refer->referred_by);
   callid = ast_strdupa(p->callid);
   localtransfer = p->refer->localtransfer;
   attendedtransfer = p->refer->attendedtransfer;

   if (!*nounlock) {
      ast_channel_unlock(p->owner);
      *nounlock = 1;
   }
   sip_pvt_unlock(p);

   /* Parking a call.  DO NOT hold any locks while calling ast_parking_ext_valid() */
   if (localtransfer && ast_parking_ext_valid(refer_to, current.chan1, refer_to_context)) {
      sip_pvt_lock(p);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);
      p->refer->status = REFER_200OK;
      append_history(p, "Xfer", "REFER to call parking.");
      sip_pvt_unlock(p);

      ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans,
         "TransferMethod: SIP\r\n"
         "TransferType: Blind\r\n"
         "Channel: %s\r\n"
         "Uniqueid: %s\r\n"
         "SIP-Callid: %s\r\n"
         "TargetChannel: %s\r\n"
         "TargetUniqueid: %s\r\n"
         "TransferExten: %s\r\n"
         "Transfer2Parking: Yes\r\n",
         ast_channel_name(current.chan1),
         ast_channel_uniqueid(current.chan1),
         callid,
         ast_channel_name(current.chan2),
         ast_channel_uniqueid(current.chan2),
         refer_to);

      if (sipdebug) {
         ast_debug(4, "SIP transfer to parking: trying to park %s. Parked by %s\n", ast_channel_name(current.chan2), ast_channel_name(current.chan1));
      }

      /* DO NOT hold any locks while calling sip_park */
      if (sip_park(current.chan2, current.chan1, req, seqno, refer_to, refer_to_context)) {
         sip_pvt_lock(p);
         transmit_notify_with_sipfrag(p, seqno, "500 Internal Server Error", TRUE);
      } else {
         sip_pvt_lock(p);
      }
      goto handle_refer_cleanup;
   }

   /* Blind transfers and remote attended xfers.
    * Locks should not be held while calling pbx_builtin_setvar_helper. This function
    * locks the channel being passed into it.*/
   if (current.chan1 && current.chan2) {
      ast_debug(3, "chan1->name: %s\n", ast_channel_name(current.chan1));
      pbx_builtin_setvar_helper(current.chan1, "BLINDTRANSFER", ast_channel_name(current.chan2));
   }

   if (current.chan2) {
      pbx_builtin_setvar_helper(current.chan2, "BLINDTRANSFER", ast_channel_name(current.chan1));
      pbx_builtin_setvar_helper(current.chan2, "SIPDOMAIN", refer_to_domain);
      pbx_builtin_setvar_helper(current.chan2, "SIPTRANSFER", "yes");
      /* One for the new channel */
      pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER", "yes");
      /* Attended transfer to remote host, prepare headers for the INVITE */
      if (!ast_strlen_zero(referred_by)) {
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REFERER", referred_by);
      }

      /* When a call is transferred to voicemail from a Digium phone, there may be
       * a Diversion header present in the REFER with an appropriate reason parameter
       * set. We need to update the redirecting information appropriately.
       */
      ast_channel_lock(p->owner);
      sip_pvt_lock(p);
      ast_party_redirecting_init(&redirecting);
      memset(&update_redirecting, 0, sizeof(update_redirecting));
      change_redirecting_information(p, req, &redirecting, &update_redirecting, FALSE);

      /* Do not hold the pvt lock during a call that causes an indicate or an async_goto.
       * Those functions lock channels which will invalidate locking order if the pvt lock
       * is held.*/
      sip_pvt_unlock(p);
      ast_channel_unlock(p->owner);
      ast_channel_update_redirecting(current.chan2, &redirecting, &update_redirecting);
      ast_party_redirecting_free(&redirecting);
   }

   sip_pvt_lock(p);
   /* Generate a Replaces string to be used in the INVITE during attended transfer */
   if (!ast_strlen_zero(p->refer->replaces_callid)) {
      char tempheader[SIPBUFSIZE];
      snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid,
         p->refer->replaces_callid_totag ? ";to-tag=" : "",
         p->refer->replaces_callid_totag,
         p->refer->replaces_callid_fromtag ? ";from-tag=" : "",
         p->refer->replaces_callid_fromtag);

      if (current.chan2) {
         sip_pvt_unlock(p);
         pbx_builtin_setvar_helper(current.chan2, "_SIPTRANSFER_REPLACES", tempheader);
         sip_pvt_lock(p);
      }
   }

   /* Connect the call */

   /* FAKE ringing if not attended transfer */
   if (!p->refer->attendedtransfer) {
      transmit_notify_with_sipfrag(p, seqno, "180 Ringing", FALSE);
   }

   /* For blind transfer, this will lead to a new call */
   /* For attended transfer to remote host, this will lead to
      a new SIP call with a replaces header, if the dial plan allows it
   */
   if (!current.chan2) {
      /* We have no bridge, so we're talking with Asterisk somehow */
      /* We need to masquerade this call */
      /* What to do to fix this situation:
         * Set up the new call in a new channel
         * Let the new channel masq into this channel
         Please add that code here :-)
      */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable (can't handle one-legged xfers)", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);  
      append_history(p, "Xfer", "Refer failed (only bridged calls).");
      res = -1;
      goto handle_refer_cleanup;
   }
   ast_set_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */

   sip_pvt_unlock(p);

   /* For blind transfers, move the call to the new extensions. For attended transfers on multiple
    * servers - generate an INVITE with Replaces. Either way, let the dial plan decided
    * indicate before masquerade so the indication actually makes it to the real channel
    * when using local channels with MOH passthru */
   ast_indicate(current.chan2, AST_CONTROL_UNHOLD);
   res = ast_async_goto(current.chan2, refer_to_context, refer_to, 1);

   if (!res) {
      ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans,
         "TransferMethod: SIP\r\n"
         "TransferType: Blind\r\n"
         "Channel: %s\r\n"
         "Uniqueid: %s\r\n"
         "SIP-Callid: %s\r\n"
         "TargetChannel: %s\r\n"
         "TargetUniqueid: %s\r\n"
         "TransferExten: %s\r\n"
         "TransferContext: %s\r\n",
         ast_channel_name(current.chan1),
         ast_channel_uniqueid(current.chan1),
         callid,
         ast_channel_name(current.chan2),
         ast_channel_uniqueid(current.chan2),
         refer_to,
         refer_to_context);
      /* Success  - we have a new channel */
      ast_debug(3, "%s transfer succeeded. Telling transferer.\n", attendedtransfer? "Attended" : "Blind");

      /* XXX - what to we put in CEL 'extra' for attended transfers to external systems? NULL for now */
      ast_channel_lock(current.chan1);
      ast_cel_report_event(current.chan1, p->refer->attendedtransfer? AST_CEL_ATTENDEDTRANSFER : AST_CEL_BLINDTRANSFER, NULL, p->refer->attendedtransfer ? NULL : p->refer->refer_to, current.chan2);
      ast_channel_unlock(current.chan1);

      sip_pvt_lock(p);
      transmit_notify_with_sipfrag(p, seqno, "200 Ok", TRUE);
      if (p->refer->localtransfer) {
         p->refer->status = REFER_200OK;
      }
      if (p->owner) {
         ast_channel_hangupcause_set(p->owner, AST_CAUSE_NORMAL_CLEARING);
      }
      append_history(p, "Xfer", "Refer succeeded.");
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);
      /* Do not hangup call, the other side do that when we say 200 OK */
      /* We could possibly implement a timer here, auto congestion */
      res = 0;
   } else {
      sip_pvt_lock(p);
      ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Don't delay hangup */
      ast_debug(3, "%s transfer failed. Resuming original call.\n", p->refer->attendedtransfer? "Attended" : "Blind");
      append_history(p, "Xfer", "Refer failed.");
      /* Failure of some kind */
      p->refer->status = REFER_FAILED;
      transmit_notify_with_sipfrag(p, seqno, "503 Service Unavailable", TRUE);
      ast_clear_flag(&p->flags[0], SIP_GOTREFER);
      res = -1;
   }

handle_refer_cleanup:
   if (current.chan1) {
      ast_channel_unref(current.chan1);
   }
   if (current.chan2) {
      ast_channel_unref(current.chan2);
   }

   /* Make sure we exit with the pvt locked */
   return res;
}

static int handle_request_register	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	sin,
		const char *	e
	)		[static]

Handle incoming REGISTER request.

Definition at line 28095 of file chan_sip.c.

References append_history, ast_debug, ast_log(), ast_sockaddr_stringify(), check_via(), copy_request(), LOG_NOTICE, LOG_WARNING, register_verify(), sip_get_header(), sip_methods, sip_scheddestroy(), and cfsip_methods::text.

Referenced by handle_incoming().

{
   enum check_auth_result res;

   /* If this is not the intial request, and the initial request isn't
    * a register, something screwy happened, so bail */
   if (p->initreq.headers && p->initreq.method != SIP_REGISTER) {
      ast_log(LOG_WARNING, "Ignoring spurious REGISTER with Call-ID: %s\n", p->callid);
      return -1;
   }

   /* Use this as the basis */
   copy_request(&p->initreq, req);
   if (sipdebug)
      ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   check_via(p, req);

   if ((res = register_verify(p, addr, req, e)) < 0) {
      const char *reason;

      switch (res) {
      case AUTH_SECRET_FAILED:
         reason = "Wrong password";
         break;
      case AUTH_USERNAME_MISMATCH:
         reason = "Username/auth name mismatch";
         break;
      case AUTH_NOT_FOUND:
         reason = "No matching peer found";
         break;
      case AUTH_UNKNOWN_DOMAIN:
         reason = "Not a local domain";
         break;
      case AUTH_PEER_NOT_DYNAMIC:
         reason = "Peer is not supposed to register";
         break;
      case AUTH_ACL_FAILED:
         reason = "Device does not match ACL";
         break;
      case AUTH_BAD_TRANSPORT:
         reason = "Device not configured to use this transport type";
         break;
      case AUTH_RTP_FAILED:
         reason = "RTP initialization failed";
         break;
      default:
         reason = "Unknown failure";
         break;
      }
      ast_log(LOG_NOTICE, "Registration from '%s' failed for '%s' - %s\n",
         sip_get_header(req, "To"), ast_sockaddr_stringify(addr),
         reason);
      append_history(p, "RegRequest", "Failed : Account %s : %s", sip_get_header(req, "To"), reason);
   } else {
      req->authenticated = 1;
      append_history(p, "RegRequest", "Succeeded : Account %s", sip_get_header(req, "To"));
   }

   if (res != AUTH_CHALLENGE_SENT) {
      /* Destroy the session, but keep us around for just a bit in case they don't
         get our 200 OK */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }

   return res;
}

static int handle_request_subscribe	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		uint32_t	seqno,
		const char *	e
	)		[static]

Handle incoming SUBSCRIBE request.

Definition at line 27687 of file chan_sip.c.

References __get_header(), add_peer_mwi_subs(), allow_notify_user_presence(), ao2_cleanup, ao2_lock, ao2_ref, ao2_unlock, append_history, ast_channel_creationtime(), ast_debug, AST_EXTENSION_RINGING, ast_extension_state2str(), ast_extension_state_add_destroy_extended(), ast_extension_state_del(), ast_extension_state_extended(), ast_free, ast_hint_presence_state(), AST_LIST_EMPTY, ast_log(), ast_set_flag, ast_sockaddr_stringify(), ast_string_field_build, ast_strlen_zero(), ast_test_flag, ast_verbose(), build_contact(), build_route(), cb_extensionstate(), cb_extensionstate_destroy(), check_user_full(), check_via(), copy_request(), state_notify_data::device_state_info, dialog_unlink_all(), extensionstate_update(), find_ringing_channel(), get_destination(), gettag(), handle_cc_subscribe(), LOG_NOTICE, LOG_WARNING, make_our_tag(), max_subexpiry, NONE, option_debug, parse_ok_contact(), state_notify_data::presence_message, state_notify_data::presence_state, state_notify_data::presence_subtype, pvt_set_needdestroy(), S_OR, set_pvt_allowed_methods(), sip_cancel_destroy(), sip_cfg, sip_get_header(), sip_methods, sip_pvt_lock, sip_pvt_unlock, sip_ref_peer(), sip_scheddestroy(), sip_send_mwi_to_peer(), sip_unref_peer(), state_notify_data::state, cfsip_methods::text, transmit_response(), transmit_response_with_minexpires(), and TRUE.

Referenced by handle_incoming().

{
   int res = 0;
   struct sip_peer *authpeer = NULL;
   char *event = ast_strdupa(sip_get_header(req, "Event")); /* Get Event package name */
   int resubscribe = (p->subscribed != NONE) && !req->ignore;
   char *options;

   if (p->initreq.headers) {
      /* We already have a dialog */
      if (p->initreq.method != SIP_SUBSCRIBE) {
         /* This is a SUBSCRIBE within another SIP dialog, which we do not support */
         /* For transfers, this could happen, but since we haven't seen it happening, let us just refuse this */
         transmit_response(p, "403 Forbidden (within dialog)", req);
         /* Do not destroy session, since we will break the call if we do */
         ast_debug(1, "Got a subscription within the context of another call, can't handle that - %s (Method %s)\n", p->callid, sip_methods[p->initreq.method].text);
         return 0;
      } else if (req->debug) {
         if (resubscribe)
            ast_debug(1, "Got a re-subscribe on existing subscription %s\n", p->callid);
         else
            ast_debug(1, "Got a new subscription %s (possibly with auth) or retransmission\n", p->callid);
      }
   }

   /* Check if we have a global disallow setting on subscriptions.
      if so, we don't have to check peer settings after auth, which saves a lot of processing
   */
   if (!sip_cfg.allowsubscribe) {
      transmit_response(p, "403 Forbidden (policy)", req);
      pvt_set_needdestroy(p, "forbidden");
      return 0;
   }

   if (!req->ignore && !resubscribe) { /* Set up dialog, new subscription */
      const char *to = sip_get_header(req, "To");
      char totag[128];
      set_pvt_allowed_methods(p, req);

      /* Check to see if a tag was provided, if so this is actually a resubscription of a dialog we no longer know about */
      if (!ast_strlen_zero(to) && gettag(req, "To", totag, sizeof(totag))) {
         if (req->debug)
            ast_verbose("Received resubscription for a dialog we no longer know about. Telling remote side to subscribe again.\n");
         transmit_response(p, "481 Subscription does not exist", req);
         pvt_set_needdestroy(p, "subscription does not exist");
         return 0;
      }

      /* Use this as the basis */
      if (req->debug)
         ast_verbose("Creating new subscription\n");

      copy_request(&p->initreq, req);
      if (sipdebug)
         ast_debug(4, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
      check_via(p, req);
      build_route(p, req, 0, 0);
   } else if (req->debug && req->ignore)
      ast_verbose("Ignoring this SUBSCRIBE request\n");

   /* Find parameters to Event: header value and remove them for now */
   if (ast_strlen_zero(event)) {
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: <none>\n");
      pvt_set_needdestroy(p, "unknown event package in subscribe");
      return 0;
   }
   if ((options = strchr(event, ';')) != NULL) {
      *options++ = '\0';
   }

   /* Handle authentication if we're new and not a retransmission. We can't just
    * use if !req->ignore, because then we'll end up sending
    * a 200 OK if someone retransmits without sending auth */
   if (p->subscribed == NONE || resubscribe) {
      res = check_user_full(p, req, SIP_SUBSCRIBE, e, XMIT_UNRELIABLE, addr, &authpeer);

      /* if an authentication response was sent, we are done here */
      if (res == AUTH_CHALLENGE_SENT)  /* authpeer = NULL here */
         return 0;
      if (res != AUTH_SUCCESSFUL) {
         ast_log(LOG_NOTICE, "Failed to authenticate device %s for SUBSCRIBE\n", sip_get_header(req, "From"));
         transmit_response(p, "403 Forbidden", req);

         pvt_set_needdestroy(p, "authentication failed");
         return 0;
      }
   }

   /* At this point, we hold a reference to authpeer (if not NULL).  It
    * must be released when done.
    */

   /* Check if this device  is allowed to subscribe at all */
   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)) {
      transmit_response(p, "403 Forbidden (policy)", req);
      pvt_set_needdestroy(p, "subscription not allowed");
      if (authpeer) {
         sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 1)");
      }
      return 0;
   }

   /* Get full contact header - this needs to be used as a request URI in NOTIFY's */
   parse_ok_contact(p, req);
   build_contact(p);

   /* Initialize tag for new subscriptions */
   if (ast_strlen_zero(p->tag)) {
      make_our_tag(p);
   }

   if (!strcmp(event, "presence") || !strcmp(event, "dialog")) { /* Presence, RFC 3842 */
      int gotdest;
      const char *accept;
      int start = 0;
      enum subscriptiontype subscribed = NONE;
      const char *unknown_accept = NULL;

                /* Get destination right away */
                gotdest = get_destination(p, NULL, NULL);
      if (gotdest != SIP_GET_DEST_EXTEN_FOUND) {
         if (gotdest == SIP_GET_DEST_INVALID_URI) {
            transmit_response(p, "416 Unsupported URI scheme", req);
         } else {
            transmit_response(p, "404 Not Found", req);
         }
         pvt_set_needdestroy(p, "subscription target not found");
         if (authpeer) {
            sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
         }
         return 0;
      }

      /* Header from Xten Eye-beam Accept: multipart/related, application/rlmi+xml, application/pidf+xml, application/xpidf+xml */
      accept = __get_header(req, "Accept", &start);
      while ((subscribed == NONE) && !ast_strlen_zero(accept)) {
         if (strstr(accept, "application/pidf+xml")) {
            if (strstr(p->useragent, "Polycom")) {
               subscribed = XPIDF_XML; /* Older versions of Polycom firmware will claim pidf+xml, but really they only support xpidf+xml */
            } else {
               subscribed = PIDF_XML; /* RFC 3863 format */
            }
         } else if (strstr(accept, "application/dialog-info+xml")) {
            subscribed = DIALOG_INFO_XML;
            /* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
         } else if (strstr(accept, "application/cpim-pidf+xml")) {
            subscribed = CPIM_PIDF_XML;    /* RFC 3863 format */
         } else if (strstr(accept, "application/xpidf+xml")) {
            subscribed = XPIDF_XML;        /* Early pre-RFC 3863 format with MSN additions (Microsoft Messenger) */
         } else {
            unknown_accept = accept;
         }
         /* check to see if there is another Accept header present */
         accept = __get_header(req, "Accept", &start);
      }

      if (!start) {
         if (p->subscribed == NONE) { /* if the subscribed field is not already set, and there is no accept header... */
            transmit_response(p, "489 Bad Event", req);
            ast_log(LOG_WARNING,"SUBSCRIBE failure: no Accept header: pvt: "
               "stateid: %d, laststate: %d, dialogver: %u, subscribecont: "
               "'%s', subscribeuri: '%s'\n",
               p->stateid,
               p->laststate,
               p->dialogver,
               p->subscribecontext,
               p->subscribeuri);
            pvt_set_needdestroy(p, "no Accept header");
            if (authpeer) {
               sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
            }
            return 0;
         }
         /* if p->subscribed is non-zero, then accept is not obligatory; according to rfc 3265 section 3.1.3, at least.
            so, we'll just let it ride, keeping the value from a previous subscription, and not abort the subscription */
      } else if (subscribed == NONE) {
         /* Can't find a format for events that we know about */
         char buf[200];

         if (!ast_strlen_zero(unknown_accept)) {
            snprintf(buf, sizeof(buf), "489 Bad Event (format %s)", unknown_accept);
         } else {
            snprintf(buf, sizeof(buf), "489 Bad Event");
         }
         transmit_response(p, buf, req);
         ast_log(LOG_WARNING,"SUBSCRIBE failure: unrecognized format:"
            "'%s' pvt: subscribed: %d, stateid: %d, laststate: %d,"
            "dialogver: %u, subscribecont: '%s', subscribeuri: '%s'\n",
            unknown_accept,
            (int)p->subscribed,
            p->stateid,
            p->laststate,
            p->dialogver,
            p->subscribecontext,
            p->subscribeuri);
         pvt_set_needdestroy(p, "unrecognized format");
         if (authpeer) {
            sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 2)");
         }
         return 0;
      } else {
         p->subscribed = subscribed;
      }
   } else if (!strcmp(event, "message-summary")) {
      int start = 0;
      int found_supported = 0;
      const char *accept;

      accept = __get_header(req, "Accept", &start);
      while (!found_supported && !ast_strlen_zero(accept)) {
         found_supported = strcmp(accept, "application/simple-message-summary") ? 0 : 1;
         if (!found_supported && (option_debug > 2)) {
            ast_debug(1, "Received SIP mailbox subscription for unknown format: %s\n", accept);
         }
         accept = __get_header(req, "Accept", &start);
      }
      if (start && !found_supported) {
         /* Format requested that we do not support */
         transmit_response(p, "406 Not Acceptable", req);
         ast_debug(2, "Received SIP mailbox subscription for unknown format: %s\n", accept);
         pvt_set_needdestroy(p, "unknown format");
         if (authpeer) {
            sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 3)");
         }
         return 0;
      }
      /* Looks like they actually want a mailbox status
        This version of Asterisk supports mailbox subscriptions
        The subscribed URI needs to exist in the dial plan
        In most devices, this is configurable to the voicemailmain extension you use
      */
      if (!authpeer || AST_LIST_EMPTY(&authpeer->mailboxes)) {
         if (!authpeer) {
            transmit_response(p, "404 Not found", req);
         } else {
            transmit_response(p, "404 Not found (no mailbox)", req);
            ast_log(LOG_NOTICE, "Received SIP subscribe for peer without mailbox: %s\n", S_OR(authpeer->name, ""));
         }
         pvt_set_needdestroy(p, "received 404 response");

         if (authpeer) {
            sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 3)");
         }
         return 0;
      }

      p->subscribed = MWI_NOTIFICATION;
      if (ast_test_flag(&authpeer->flags[1], SIP_PAGE2_SUBSCRIBEMWIONLY)) {
         ao2_unlock(p);
         add_peer_mwi_subs(authpeer);
         ao2_lock(p);
      }
      if (authpeer->mwipvt != p) {  /* Destroy old PVT if this is a new one */
         /* We only allow one subscription per peer */
         if (authpeer->mwipvt) {
            dialog_unlink_all(authpeer->mwipvt);
            authpeer->mwipvt = dialog_unref(authpeer->mwipvt, "unref dialog authpeer->mwipvt");
         }
         authpeer->mwipvt = dialog_ref(p, "setting peers' mwipvt to p");
      }

      if (p->relatedpeer != authpeer) {
         if (p->relatedpeer) {
            sip_unref_peer(p->relatedpeer, "Unref previously stored relatedpeer ptr");
         }
         p->relatedpeer = sip_ref_peer(authpeer, "setting dialog's relatedpeer pointer");
      }
      /* Do not release authpeer here */
   } else if (!strcmp(event, "call-completion")) {
      handle_cc_subscribe(p, req);
   } else { /* At this point, Asterisk does not understand the specified event */
      transmit_response(p, "489 Bad Event", req);
      ast_debug(2, "Received SIP subscribe for unknown event package: %s\n", event);
      pvt_set_needdestroy(p, "unknown event package");
      if (authpeer) {
         sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 5)");
      }
      return 0;
   }

   if (!req->ignore) {
      p->lastinvite = seqno;
   }
   if (!p->needdestroy) {
      p->expiry = atoi(sip_get_header(req, "Expires"));

      /* check if the requested expiry-time is within the approved limits from sip.conf */
      if (p->expiry > max_subexpiry) {
         p->expiry = max_subexpiry;
      } else if (p->expiry < min_subexpiry && p->expiry > 0) {
         transmit_response_with_minexpires(p, "423 Interval too small", req, min_subexpiry);
         ast_log(LOG_WARNING, "Received subscription for extension \"%s\" context \"%s\" "
            "with Expire header less than 'subminexpire' limit. Received \"Expire: %d\" min is %d\n",
            p->exten, p->context, p->expiry, min_subexpiry);
         pvt_set_needdestroy(p, "Expires is less that the min expires allowed.");
         if (authpeer) {
            sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 6)");
         }
         return 0;
      }

      if (sipdebug) {
         const char *action = p->expiry > 0 ? "Adding" : "Removing";
         if (p->subscribed == MWI_NOTIFICATION && p->relatedpeer) {
            ast_debug(2, "%s subscription for mailbox notification - peer %s\n",
                  action, p->relatedpeer->name);
         } else if (p->subscribed == CALL_COMPLETION) {
            ast_debug(2, "%s CC subscription for peer %s\n", action, p->username);
         } else {
            ast_debug(2, "%s subscription for extension %s context %s for peer %s\n",
                  action, p->exten, p->context, p->username);
         }
      }
      if (p->autokillid > -1 && sip_cancel_destroy(p))   /* Remove subscription expiry for renewals */
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      if (p->expiry > 0)
         sip_scheddestroy(p, (p->expiry + 10) * 1000);   /* Set timer for destruction of call at expiration */

      if (p->subscribed == MWI_NOTIFICATION) {
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         if (p->relatedpeer) {   /* Send first notification */
            struct sip_peer *peer = p->relatedpeer;
            sip_ref_peer(peer, "ensure a peer ref is held during MWI sending");
            ao2_unlock(p);
            sip_send_mwi_to_peer(peer, 0);
            ao2_lock(p);
            sip_unref_peer(peer, "release a peer ref now that MWI is sent");
         }
      } else if (p->subscribed != CALL_COMPLETION) {
         struct state_notify_data data = { 0, };
         char *subtype = NULL;
         char *message = NULL;
         struct ao2_container *device_state_info = NULL;

         if (p->expiry > 0 && !resubscribe) {
            /* Add subscription for extension state from the PBX core */
            if (p->stateid != -1) {
               ast_extension_state_del(p->stateid, cb_extensionstate);
            }
            dialog_ref(p, "copying dialog ptr into extension state struct");
            p->stateid = ast_extension_state_add_destroy_extended(p->context, p->exten, cb_extensionstate, cb_extensionstate_destroy, p);
            if (p->stateid == -1) {
               dialog_unref(p, "copying dialog ptr into extension state struct failed");
            }
         }

         sip_pvt_unlock(p);
         data.state = ast_extension_state_extended(NULL, p->context, p->exten, &device_state_info);
         sip_pvt_lock(p);

         if (data.state < 0) {
            ao2_cleanup(device_state_info);
            if (p->expiry > 0) {
               ast_log(LOG_NOTICE, "Got SUBSCRIBE for extension %s@%s from %s, but there is no hint for that extension.\n", p->exten, p->context, ast_sockaddr_stringify(&p->sa));
            }
            transmit_response(p, "404 Not found", req);
            pvt_set_needdestroy(p, "no extension for SUBSCRIBE");
            if (authpeer) {
               sip_unref_peer(authpeer, "sip_unref_peer, from handle_request_subscribe (authpeer 6)");
            }
            return 0;
         }
         if (allow_notify_user_presence(p)) {
            data.presence_state = ast_hint_presence_state(NULL, p->context, p->exten, &subtype, &message);
            data.presence_subtype = subtype;
            data.presence_message = message;
         }
         ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         transmit_response(p, "200 OK", req);
         /* RFC 3265: A notification must be sent on every subscribe, so force it */
         data.device_state_info = device_state_info;
         if (data.state & AST_EXTENSION_RINGING) {
            /* save last_ringing_channel_time if this state really contains a ringing channel
             * because extensionstate_update() doesn't do it if forced
             */
            struct ast_channel *ringing = find_ringing_channel(data.device_state_info, p);
            if (ringing) {
               p->last_ringing_channel_time = ast_channel_creationtime(ringing);
               ao2_ref(ringing, -1);
            }
            /* If there is no channel, this likely indicates that the ringing indication
             * is due to a custom device state. These do not have associated channels.
             */
         }
         extensionstate_update(p->context, p->exten, &data, p, TRUE);
         append_history(p, "Subscribestatus", "%s", ast_extension_state2str(data.state));
         /* hide the 'complete' exten/context in the refer_to field for later display */
         ast_string_field_build(p, subscribeuri, "%s@%s", p->exten, p->context);
         /* Deleted the slow iteration of all sip dialogs to find old subscribes from this peer for exten@context */

         ao2_cleanup(device_state_info);
         ast_free(subtype);
         ast_free(message);
      }
      if (!p->expiry) {
         pvt_set_needdestroy(p, "forcing expiration");
      }
   }

   if (authpeer) {
      sip_unref_peer(authpeer, "unref pointer into (*authpeer)");
   }
   return 1;
}

static int handle_request_update	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

bare-bones support for SIP UPDATE

XXX This is not even close to being RFC 3311-compliant. We don't advertise that we support the UPDATE method, so no one should ever try sending us an UPDATE anyway. However, Asterisk can send an UPDATE to change connected line information, so we need to be prepared to handle this. The way we distinguish such an UPDATE is through the X-Asterisk-rpid-update header.

Actually updating the media session may be some future work.

Definition at line 25047 of file chan_sip.c.

References ast_channel_queue_connected_line_update(), AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, ast_party_connected_line_init(), ast_set_party_id_all(), ast_strlen_zero(), connected, get_rpid(), ast_party_connected_line::id, ast_set_party_connected_line::id, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, ast_party_name::presentation, ast_party_number::presentation, ast_set_party_connected_line::priv, sip_get_header(), ast_party_connected_line::source, ast_party_name::str, ast_party_number::str, ast_party_id::tag, transmit_response(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_incoming().

{
   if (ast_strlen_zero(sip_get_header(req, "X-Asterisk-rpid-update"))) {
      transmit_response(p, "501 Method Not Implemented", req);
      return 0;
   }
   if (!p->owner) {
      transmit_response(p, "481 Call/Transaction Does Not Exist", req);
      return 0;
   }
   if (get_rpid(p, req)) {
      struct ast_party_connected_line connected;
      struct ast_set_party_connected_line update_connected;

      ast_party_connected_line_init(&connected);
      memset(&update_connected, 0, sizeof(update_connected));

      update_connected.id.number = 1;
      connected.id.number.valid = 1;
      connected.id.number.str = (char *) p->cid_num;
      connected.id.number.presentation = p->callingpres;

      update_connected.id.name = 1;
      connected.id.name.valid = 1;
      connected.id.name.str = (char *) p->cid_name;
      connected.id.name.presentation = p->callingpres;

      /* Invalidate any earlier private connected id representation */
      ast_set_party_id_all(&update_connected.priv);

      connected.id.tag = (char *) p->cid_tag;
      connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
      ast_channel_queue_connected_line_update(p->owner, &connected, &update_connected);
   }
   transmit_response(p, "200 OK", req);
   return 0;
}

static void handle_response	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Handle SIP response in dialogue.

Note:

only called by handle_incoming

Definition at line 23810 of file chan_sip.c.

References __sip_ack(), __sip_semi_ack(), append_history, AST_CC_CCBS, ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_name(), ast_channel_set_redirecting(), ast_clear_flag, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, ast_debug, ast_log(), ast_party_redirecting_free(), ast_party_redirecting_init(), ast_queue_control(), ast_queue_hangup_with_cause(), ast_set_flag, ast_skip_blanks(), ast_skip_nonblanks(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, ast_verbose(), change_redirecting_information(), do_proxy_auth(), FALSE, find_sdp(), find_sip_method(), gettag(), handle_response_info(), handle_response_invite(), handle_response_message(), handle_response_notify(), handle_response_peerpoke(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), handle_response_update(), hangup_sip2cause(), LOG_NOTICE, LOG_WARNING, mark_method_allowed(), mark_method_unallowed(), process_sdp(), pvt_set_needdestroy(), rh, sip_alreadygone(), sip_cancel_destroy(), sip_get_header(), sip_handle_cc(), sip_methods, stop_media_flows(), text, transmit_request(), TRUE, and update_redirecting().

Referenced by handle_incoming().

{
   struct ast_channel *owner;
   int sipmethod;
   const char *c = sip_get_header(req, "Cseq");
   /* GCC 4.2 complains if I try to cast c as a char * when passing it to ast_skip_nonblanks, so make a copy of it */
   char *c_copy = ast_strdupa(c);
   /* Skip the Cseq and its subsequent spaces */
   const char *msg = ast_skip_blanks(ast_skip_nonblanks(c_copy));

   if (!msg)
      msg = "";

   sipmethod = find_sip_method(msg);

   owner = p->owner;
   if (owner) {
      const char *rp = NULL, *rh = NULL;

      ast_channel_hangupcause_set(owner, 0);
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON) && (rh = sip_get_header(req, "Reason"))) {
         rh = ast_skip_blanks(rh);
         if (!strncasecmp(rh, "Q.850", 5)) {
            int cause = ast_channel_hangupcause(owner);
            rp = strstr(rh, "cause=");
            if (rp && sscanf(rp + 6, "%30d", &cause) == 1) {
               ast_channel_hangupcause_set(owner, cause & 0x7f);
               if (req->debug)
                  ast_verbose("Using Reason header for cause code: %d\n", ast_channel_hangupcause(owner));
            }
         }
      }

      if (!ast_channel_hangupcause(owner))
         ast_channel_hangupcause_set(owner, hangup_sip2cause(resp));
   }

   if (p->socket.type == SIP_TRANSPORT_UDP) {
      int ack_res = FALSE;

      /* Acknowledge whatever it is destined for */
      if ((resp >= 100) && (resp <= 199)) {
         /* NON-INVITE messages do not ack a 1XX response. RFC 3261 section 17.1.2.2 */
         if (sipmethod == SIP_INVITE) {
            ack_res = __sip_semi_ack(p, seqno, 0, sipmethod);
         }
      } else {
         ack_res = __sip_ack(p, seqno, 0, sipmethod);
      }

      if (ack_res == FALSE) {
         /* RFC 3261 13.2.2.4 and 17.1.1.2 - We must re-send ACKs to re-transmitted final responses */
         if (sipmethod == SIP_INVITE && resp >= 200) {
            transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, resp < 300 ? TRUE: FALSE);
         }

         append_history(p, "Ignore", "Ignoring this retransmit\n");
         return;
      }
   }

   /* If this is a NOTIFY for a subscription clear the flag that indicates that we have a NOTIFY pending */
   if (!p->owner && sipmethod == SIP_NOTIFY && p->pendinginvite) {
      p->pendinginvite = 0;
   }

   /* Get their tag if we haven't already */
   if (ast_strlen_zero(p->theirtag) || (resp >= 200)) {
      char tag[128];

      gettag(req, "To", tag, sizeof(tag));
      ast_string_field_set(p, theirtag, tag);
   } else {
      /* Store theirtag to track for changes when 200 responses to invites are received without SDP */
      ast_string_field_set(p, theirprovtag, p->theirtag);
   }

   /* This needs to be configurable on a channel/peer level,
      not mandatory for all communication. Sadly enough, NAT implementations
      are not so stable so we can always rely on these headers.
      Temporarily disabled, while waiting for fix.
      Fix assigned to Rizzo :-)
   */
   /* check_via_response(p, req); */

   /* RFC 3261 Section 15 specifies that if we receive a 408 or 481
    * in response to a BYE, then we should end the current dialog
    * and session.  It is known that at least one phone manufacturer
    * potentially will send a 404 in response to a BYE, so we'll be
    * liberal in what we accept and end the dialog and session if we
    * receive any of those responses to a BYE.
    */
   if ((resp == 404 || resp == 408 || resp == 481) && sipmethod == SIP_BYE) {
      pvt_set_needdestroy(p, "received 4XX response to a BYE");
      return;
   }

   if (p->relatedpeer && sipmethod == SIP_OPTIONS) {
      /* We don't really care what the response is, just that it replied back.
         Well, as long as it's not a 100 response...  since we might
         need to hang around for something more "definitive" */
      if (resp != 100)
         handle_response_peerpoke(p, resp, req);
   } else if (sipmethod == SIP_REFER && resp >= 200) {
      handle_response_refer(p, resp, rest, req, seqno);
   } else if (sipmethod == SIP_PUBLISH) {
      /* SIP PUBLISH transcends this morass of doodoo and instead
       * we just always call the response handler. Good gravy!
       */
      handle_response_publish(p, resp, rest, req, seqno);
   } else if (sipmethod == SIP_INFO) {
      /* More good gravy! */
      handle_response_info(p, resp, rest, req, seqno);
   } else if (sipmethod == SIP_MESSAGE) {
      /* More good gravy! */
      handle_response_message(p, resp, rest, req, seqno);
   } else if (sipmethod == SIP_NOTIFY) {
      /* The gravy train continues to roll */
      handle_response_notify(p, resp, rest, req, seqno);
   } else if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      switch(resp) {
      case 100:   /* 100 Trying */
      case 101:   /* 101 Dialog establishment */
      case 183:   /* 183 Session Progress */
      case 180:   /* 180 Ringing */
      case 182:   /* 182 Queued */
      case 181:   /* 181 Call Is Being Forwarded */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 200:   /* 200 OK */
         p->authtries = 0; /* Reset authentication counter */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_REGISTER) {
            handle_response_register(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_SUBSCRIBE) {
            ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
            handle_response_subscribe(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {     /* Ok, we're ready to go */
            pvt_set_needdestroy(p, "received 200 response");
            ast_clear_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
         }
         break;
      case 401: /* Not www-authorized on SIP method */
      case 407: /* Proxy auth required */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER)
            handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_UPDATE) {
            handle_response_update(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            if (p->options)
               p->options->auth_type = resp;
            if (ast_strlen_zero(p->authname)) {
               ast_log(LOG_WARNING, "Asked to authenticate %s, to %s but we have no matching peer!\n",
                     msg, ast_sockaddr_stringify(&p->recv));
               pvt_set_needdestroy(p, "unable to authenticate BYE");
            } else if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp,  sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, sip_get_header(&p->initreq, "From"));
               pvt_set_needdestroy(p, "failed to authenticate BYE");
            }
         } else {
            ast_log(LOG_WARNING, "Got authentication request (%d) on %s to '%s'\n", resp, sip_methods[sipmethod].text, sip_get_header(req, "To"));
            pvt_set_needdestroy(p, "received 407 response");
         }
         break;
      case 403: /* Forbidden - we failed authentication */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (p->registry && sipmethod == SIP_REGISTER)
            handle_response_register(p, resp, rest, req, seqno);
         else {
            ast_log(LOG_WARNING, "Forbidden - maybe wrong password on authentication for %s\n", msg);
            pvt_set_needdestroy(p, "received 403 response");
         }
         break;
      case 404: /* Not found */
         if (p->registry && sipmethod == SIP_REGISTER)
            handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_SUBSCRIBE)
            handle_response_subscribe(p, resp, rest, req, seqno);
         else if (owner)
            ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
         break;
      case 423: /* Interval too brief */
         if (sipmethod == SIP_REGISTER)
            handle_response_register(p, resp, rest, req, seqno);
         break;
      case 408: /* Request timeout - terminate dialog */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_REGISTER)
            handle_response_register(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "received 408 response");
            ast_debug(4, "Got timeout on bye. Thanks for the answer. Now, kill this call\n");
         } else {
            if (owner)
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            pvt_set_needdestroy(p, "received 408 response");
         }
         break;

      case 428:
      case 422: /* Session-Timers: Session Interval Too Small */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         }
         break;

      case 481: /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_SUBSCRIBE) {
            handle_response_subscribe(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            /* The other side has no transaction to bye,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else if (sipmethod == SIP_CANCEL) {
            /* The other side has no transaction to cancel,
            just assume it's all right then */
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
         } else {
            ast_log(LOG_WARNING, "Remote host can't match request %s to call '%s'. Giving up.\n", sip_methods[sipmethod].text, p->callid);
            /* Guessing that this is not an important request */
         }
         break;
      case 487:
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 415: /* Unsupported media type */
      case 488: /* Not acceptable here - codec error */
      case 606: /* Not Acceptable */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      case 491: /* Pending */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else {
            ast_debug(1, "Got 491 on %s, unsupported. Call ID %s\n", sip_methods[sipmethod].text, p->callid);
            pvt_set_needdestroy(p, "received 491 response");
         }
         break;
      case 405: /* Method not allowed */
      case 501: /* Not Implemented */
         mark_method_unallowed(&p->allowed_methods, sipmethod);
         if (p->relatedpeer) {
            mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
         }
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else
            ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n", ast_sockaddr_stringify(&p->sa), msg);
         break;
      default:
         if ((resp >= 200) && (resp < 300)) { /* on any 2XX response do the following */
            if (sipmethod == SIP_INVITE) {
               handle_response_invite(p, resp, rest, req, seqno);
            }
         } else if ((resp >= 300) && (resp < 700)) {
            /* Fatal response */
            if ((resp != 487))
               ast_verb(3, "Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
   
            if (sipmethod == SIP_INVITE)
               stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */

            /* XXX Locking issues?? XXX */
            switch(resp) {
            case 300: /* Multiple Choices */
            case 301: /* Moved permanently */
            case 302: /* Moved temporarily */
            case 305: /* Use Proxy */
               if (p->owner) {
                  struct ast_party_redirecting redirecting;
                  struct ast_set_party_redirecting update_redirecting;

                  ast_party_redirecting_init(&redirecting);
                  memset(&update_redirecting, 0, sizeof(update_redirecting));
                  change_redirecting_information(p, req, &redirecting,
                     &update_redirecting, TRUE);
                  ast_channel_set_redirecting(p->owner, &redirecting,
                     &update_redirecting);
                  ast_party_redirecting_free(&redirecting);
               }
               /* Fall through */
            case 486: /* Busy here */
            case 600: /* Busy everywhere */
            case 603: /* Decline */
               if (p->owner) {
                  sip_handle_cc(p, req, AST_CC_CCBS);
                  ast_queue_control(p->owner, AST_CONTROL_BUSY);
               }
               break;
            case 482: /* Loop Detected */
            case 480: /* Temporarily Unavailable */
            case 404: /* Not Found */
            case 410: /* Gone */
            case 400: /* Bad Request */
            case 500: /* Server error */
               if (sipmethod == SIP_SUBSCRIBE) {
                  handle_response_subscribe(p, resp, rest, req, seqno);
                  break;
               }
               /* Fall through */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server Timeout */
               if (owner)
                  ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
               break;
            case 484: /* Address Incomplete */
               if (owner && sipmethod != SIP_BYE) {
                  switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
                  case SIP_PAGE2_ALLOWOVERLAP_YES:
                     ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
                     break;
                  default:
                     ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(404));
                     break;
                  }
               }
               break;
            default:
               /* Send hangup */ 
               if (owner && sipmethod != SIP_BYE)
                  ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
               break;
            }
            /* ACK on invite */
            if (sipmethod == SIP_INVITE)
               transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
            sip_alreadygone(p);
            if (!p->owner) {
               pvt_set_needdestroy(p, "transaction completed");
            }
         } else if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               if (find_sdp(req))
                  process_sdp(p, req, SDP_T38_NONE);
               if (p->owner) {
                  /* Queue a progress frame */
                  ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
               }
            }
         } else
            ast_log(LOG_NOTICE, "Don't know how to handle a %d %s response from %s\n", resp, rest, p->owner ? ast_channel_name(p->owner) : ast_sockaddr_stringify(&p->sa));
      }
   } else { 
      /* Responses to OUTGOING SIP requests on INCOMING calls
         get handled here. As well as out-of-call message responses */
      if (req->debug)
         ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);

      if (sipmethod == SIP_INVITE && resp == 200) {
         /* Tags in early session is replaced by the tag in 200 OK, which is
         the final reply to our INVITE */
         char tag[128];

         gettag(req, "To", tag, sizeof(tag));
         ast_string_field_set(p, theirtag, tag);
      }

      switch(resp) {
      case 200:
         if (sipmethod == SIP_INVITE) {
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_CANCEL) {
            ast_debug(1, "Got 200 OK on CANCEL\n");

            /* Wait for 487, then destroy */
         } else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "transaction completed");
         }
         break;
      case 401:   /* www-auth */
      case 407:
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         else if (sipmethod == SIP_BYE) {
            if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, sipmethod, 0)) {
               ast_log(LOG_NOTICE, "Failed to authenticate on %s to '%s'\n", msg, sip_get_header(&p->initreq, "From"));
               pvt_set_needdestroy(p, "failed to authenticate BYE");
            }
         }
         break;
      case 481:   /* Call leg does not exist */
         if (sipmethod == SIP_INVITE) {
            /* Re-invite failed */
            handle_response_invite(p, resp, rest, req, seqno);
         } else if (sipmethod == SIP_BYE) {
            pvt_set_needdestroy(p, "received 481 response");
         } else if (sipdebug) {
            ast_debug(1, "Remote host can't match request %s to call '%s'. Giving up\n", sip_methods[sipmethod].text, p->callid);
         }
         break;
      case 501: /* Not Implemented */
         if (sipmethod == SIP_INVITE)
            handle_response_invite(p, resp, rest, req, seqno);
         break;
      default: /* Errors without handlers */
         if ((resp >= 100) && (resp < 200)) {
            if (sipmethod == SIP_INVITE) {   /* re-invite */
               if (!req->ignore && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
            }
         } else if ((resp >= 200) && (resp < 300)) { /* on any unrecognized 2XX response do the following */
            if (sipmethod == SIP_INVITE) {
               handle_response_invite(p, resp, rest, req, seqno);
            }
         } else if ((resp >= 300) && (resp < 700)) {
            if ((resp != 487))
               ast_verb(3, "Incoming call: Got SIP response %d \"%s\" back from %s\n", resp, rest, ast_sockaddr_stringify(&p->sa));
            switch(resp) {
            case 415: /* Unsupported media type */
            case 488: /* Not acceptable here - codec error */
            case 603: /* Decline */
            case 500: /* Server error */
            case 502: /* Bad gateway */
            case 503: /* Service Unavailable */
            case 504: /* Server timeout */

               /* re-invite failed */
               if (sipmethod == SIP_INVITE && sip_cancel_destroy(p))
                  ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
               break;
            }
         }
         break;
      }
   }
}

static void handle_response_info	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23676 of file chan_sip.c.

References ast_log(), ast_sockaddr_stringify(), ast_verb, LOG_WARNING, mark_method_allowed(), mark_method_unallowed(), sip_methods, cfsip_methods::text, and text.

Referenced by handle_response().

{
   int sipmethod = SIP_INFO;

   switch (resp) {
   case 401: /* Not www-authorized on SIP method */
   case 407: /* Proxy auth required */
      ast_log(LOG_WARNING, "Host '%s' requests authentication (%d) for '%s'\n",
         ast_sockaddr_stringify(&p->sa), resp, sip_methods[sipmethod].text);
      break;
   case 405: /* Method not allowed */
   case 501: /* Not Implemented */
      mark_method_unallowed(&p->allowed_methods, sipmethod);
      if (p->relatedpeer) {
         mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
      }
      ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
         ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
      break;
   default:
      if (300 <= resp && resp < 700) {
         ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
            sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
      }
      break;
   }
}

static void handle_response_invite	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Handle SIP response to INVITE dialogue.

Definition at line 22701 of file chan_sip.c.

References append_history, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_NORMAL_CLEARING, AST_CC_CCNR, ast_channel_hangupcause_set(), ast_channel_name(), ast_channel_queue_connected_line_update(), ast_channel_queue_redirecting_update(), ast_channel_uniqueid(), AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, AST_CONTROL_ANSWER, AST_CONTROL_CONGESTION, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, AST_CONTROL_UPDATE_RTP_PEER, ast_debug, ast_log(), ast_null_frame, ast_party_connected_line_init(), ast_party_redirecting_free(), ast_party_redirecting_init(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_queue_control(), ast_queue_frame(), ast_queue_hangup_with_cause(), ast_random(), ast_rtp_instance_activate(), ast_rtp_instance_get_remote_address(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_set_party_id_all(), ast_setstate(), ast_sockaddr_isnull(), AST_STATE_RINGING, AST_STATE_UP, ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_route(), change_redirecting_information(), change_t38_state(), check_pendings(), connected, do_proxy_auth(), EVENT_FLAG_SYSTEM, FALSE, find_sdp(), get_rpid(), hangup_sip2cause(), ast_party_connected_line::id, ast_set_party_connected_line::id, LOG_NOTICE, LOG_WARNING, manager_event, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, parse_ok_contact(), parse_session_expires(), ast_party_name::presentation, ast_party_number::presentation, ast_set_party_connected_line::priv, ast_set_party_redirecting::priv_from, ast_set_party_redirecting::priv_orig, ast_set_party_redirecting::priv_to, proc_422_rsp(), process_sdp(), pvt_set_needdestroy(), set_address_from_contact(), set_pvt_allowed_methods(), sip_alreadygone(), sip_cancel_destroy(), sip_cfg, sip_get_header(), sip_handle_cc(), sip_queue_hangup_cause(), sip_reinvite_retry(), sip_scheddestroy(), ast_party_connected_line::source, st_get_mode(), st_get_se(), start_session_timer(), ast_party_name::str, ast_party_number::str, ast_party_id::tag, transmit_reinvite_with_sdp(), transmit_request(), TRUE, update_call_counter(), update_redirecting(), ast_party_name::valid, and ast_party_number::valid.

Referenced by handle_response().

{
   int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
   int res = 0;
   int xmitres = 0;
   int reinvite = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   char *p_hdrval;
   int rtn;
   struct ast_party_connected_line connected;
   struct ast_set_party_connected_line update_connected;

   if (reinvite) {
      ast_debug(4, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
   } else {
      ast_debug(4, "SIP response %d to standard invite\n", resp);
   }

   if (p->alreadygone) { /* This call is already gone */
      ast_debug(1, "Got response on call that is already terminated: %s (ignoring)\n", p->callid);
      return;
   }

   /* Acknowledge sequence number - This only happens on INVITE from SIP-call */
   /* Don't auto congest anymore since we've gotten something useful back */
   AST_SCHED_DEL_UNREF(sched, p->initid, dialog_unref(p, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));

   /* RFC3261 says we must treat every 1xx response (but not 100)
      that we don't recognize as if it was 183.
   */
   if (resp > 100 && resp < 200 && resp!=101 && resp != 180 && resp != 181 && resp != 182 && resp != 183) {
      resp = 183;
   }

   /* For INVITE, treat all 2XX responses as we would a 200 response */
   if ((resp >= 200) && (resp < 300)) {
      resp = 200;
   }

   /* Any response between 100 and 199 is PROCEEDING */
   if (resp >= 100 && resp < 200 && p->invitestate == INV_CALLING) {
      p->invitestate = INV_PROCEEDING;
   }

   /* Final response, not 200 ? */
   if (resp >= 300 && (p->invitestate == INV_CALLING || p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA )) {
      p->invitestate = INV_COMPLETED;
   }
   
   if ((resp >= 200 && reinvite)) {
      p->ongoing_reinvite = 0;
      if (p->reinviteid > -1) {
         AST_SCHED_DEL_UNREF(sched, p->reinviteid, dialog_unref(p, "unref dialog for reinvite timeout because of a final response"));
      }
   }

   /* Final response, clear out pending invite */
   if ((resp == 200 || resp >= 300) && p->pendinginvite && seqno == p->pendinginvite) {
      p->pendinginvite = 0;
   }

   /* If this is a response to our initial INVITE, we need to set what we can use
    * for this peer.
    */
   if (!reinvite) {
      set_pvt_allowed_methods(p, req);
   }

   switch (resp) {
   case 100:   /* Trying */
   case 101:   /* Dialog establishment */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      }
      check_pendings(p);
      break;

   case 180:   /* 180 Ringing */
   case 182:       /* 182 Queued */
      if (!req->ignore && p->invitestate != INV_CANCELLED && sip_cancel_destroy(p)) {
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      }
      /* Store Route-set from provisional SIP responses so
       * early-dialog request can be routed properly
       * */
      parse_ok_contact(p, req);
      if (!reinvite) {
         build_route(p, req, 1, resp);
      }
      if (!req->ignore && p->owner) {
         if (get_rpid(p, req)) {
            /* Queue a connected line update */
            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));

            update_connected.id.number = 1;
            connected.id.number.valid = 1;
            connected.id.number.str = (char *) p->cid_num;
            connected.id.number.presentation = p->callingpres;

            update_connected.id.name = 1;
            connected.id.name.valid = 1;
            connected.id.name.str = (char *) p->cid_name;
            connected.id.name.presentation = p->callingpres;

            /* Invalidate any earlier private connected id representation */
            ast_set_party_id_all(&update_connected.priv);

            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         sip_handle_cc(p, req, AST_CC_CCNR);
         ast_queue_control(p->owner, AST_CONTROL_RINGING);
         if (ast_channel_state(p->owner) != AST_STATE_UP) {
            ast_setstate(p->owner, AST_STATE_RINGING);
         }
      }
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED) {
            p->invitestate = INV_EARLY_MEDIA;
         }
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame only if we have SDP in 180 or 182 */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
         ast_rtp_instance_activate(p->rtp);
      }
      check_pendings(p);
      break;

   case 181:   /* Call Is Being Forwarded */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      /* Store Route-set from provisional SIP responses so
       * early-dialog request can be routed properly
       * */
      parse_ok_contact(p, req);
      if (!reinvite) {
         build_route(p, req, 1, resp);
      }
      if (!req->ignore && p->owner) {
         struct ast_party_redirecting redirecting;
         struct ast_set_party_redirecting update_redirecting;

         ast_party_redirecting_init(&redirecting);
         memset(&update_redirecting, 0, sizeof(update_redirecting));
         change_redirecting_information(p, req, &redirecting, &update_redirecting,
            FALSE);

         /* Invalidate any earlier private redirecting id representations */
         ast_set_party_id_all(&update_redirecting.priv_orig);
         ast_set_party_id_all(&update_redirecting.priv_from);
         ast_set_party_id_all(&update_redirecting.priv_to);

         ast_channel_queue_redirecting_update(p->owner, &redirecting,
            &update_redirecting);
         ast_party_redirecting_free(&redirecting);
         sip_handle_cc(p, req, AST_CC_CCNR);
      }
      check_pendings(p);
      break;

   case 183:   /* Session progress */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      }
      /* Store Route-set from provisional SIP responses so
       * early-dialog request can be routed properly
       * */
      parse_ok_contact(p, req);
      if (!reinvite) {
         build_route(p, req, 1, resp);
      }
      if (!req->ignore && p->owner) {
         if (get_rpid(p, req)) {
            /* Queue a connected line update */
            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));

            update_connected.id.number = 1;
            connected.id.number.valid = 1;
            connected.id.number.str = (char *) p->cid_num;
            connected.id.number.presentation = p->callingpres;

            update_connected.id.name = 1;
            connected.id.name.valid = 1;
            connected.id.name.str = (char *) p->cid_name;
            connected.id.name.presentation = p->callingpres;

            /* Invalidate any earlier private connected id representation */
            ast_set_party_id_all(&update_connected.priv);

            connected.id.tag = (char *) p->cid_tag;
            connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
            ast_channel_queue_connected_line_update(p->owner, &connected,
               &update_connected);
         }
         sip_handle_cc(p, req, AST_CC_CCNR);
      }
      if (find_sdp(req)) {
         if (p->invitestate != INV_CANCELLED) {
            p->invitestate = INV_EARLY_MEDIA;
         }
         res = process_sdp(p, req, SDP_T38_NONE);
         if (!req->ignore && p->owner) {
            /* Queue a progress frame */
            ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
         }
         ast_rtp_instance_activate(p->rtp);
      } else {
         /* Alcatel PBXs are known to send 183s with no SDP after sending
          * a 100 Trying response. We're just going to treat this sort of thing
          * the same as we would treat a 180 Ringing
          */
         if (!req->ignore && p->owner) {
            ast_queue_control(p->owner, AST_CONTROL_RINGING);
         }
      }
      check_pendings(p);
      break;

   case 200:   /* 200 OK on invite - someone's answering our call */
      if (!req->ignore && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) {
         ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
      }
      p->authtries = 0;
      if (find_sdp(req)) {
         if ((res = process_sdp(p, req, SDP_T38_ACCEPT)) && !req->ignore) {
            if (!reinvite) {
               /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
               /* For re-invites, we try to recover */
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
               p->hangupcause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
               if (p->owner) {
                  ast_channel_hangupcause_set(p->owner, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
                  sip_queue_hangup_cause(p, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
               }
            }
         }
         ast_rtp_instance_activate(p->rtp);
      } else if (!reinvite) {
         struct ast_sockaddr remote_address = {{0,}};

         ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
         if (ast_sockaddr_isnull(&remote_address) || (!ast_strlen_zero(p->theirprovtag) && strcmp(p->theirtag, p->theirprovtag))) {
            ast_log(LOG_WARNING, "Received response: \"200 OK\" from '%s' without SDP\n", p->relatedpeer->name);
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
            ast_rtp_instance_activate(p->rtp);
         }
      }

      if (!req->ignore && p->owner) {
         int rpid_changed;

         rpid_changed = get_rpid(p, req);
         if (rpid_changed || !reinvite) {
            /* Queue a connected line update */
            ast_party_connected_line_init(&connected);
            memset(&update_connected, 0, sizeof(update_connected));
            if (rpid_changed
               || !ast_strlen_zero(p->cid_num)
               || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
               update_connected.id.number = 1;
               connected.id.number.valid = 1;
               connected.id.number.str = (char *) p->cid_num;
               connected.id.number.presentation = p->callingpres;
            }
            if (rpid_changed
               || !ast_strlen_zero(p->cid_name)
               || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
               update_connected.id.name = 1;
               connected.id.name.valid = 1;
               connected.id.name.str = (char *) p->cid_name;
               connected.id.name.presentation = p->callingpres;
            }
            if (update_connected.id.number || update_connected.id.name) {
               /* Invalidate any earlier private connected id representation */
               ast_set_party_id_all(&update_connected.priv);

               connected.id.tag = (char *) p->cid_tag;
               connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
               ast_channel_queue_connected_line_update(p->owner, &connected,
                  &update_connected);
            }
         }
      }

      /* Parse contact header for continued conversation */
      /* When we get 200 OK, we know which device (and IP) to contact for this call */
      /* This is important when we have a SIP proxy between us and the phone */
      if (outgoing) {
         update_call_counter(p, DEC_CALL_RINGING);
         parse_ok_contact(p, req);
         /* Save Record-Route for any later requests we make on this dialogue */
         if (!reinvite) {
            build_route(p, req, 1, resp);
         }
         if(set_address_from_contact(p)) {
            /* Bad contact - we don't know how to reach this device */
            /* We need to ACK, but then send a bye */
            if (!p->route && !req->ignore) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            }
         }

      }

      if (!req->ignore && p->owner) {
         if (!reinvite && !res) {
            ast_queue_control(p->owner, AST_CONTROL_ANSWER);
            if (sip_cfg.callevents) {
               manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
                  "Channel: %s\r\nChanneltype: %s\r\nUniqueid: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
                  ast_channel_name(p->owner), "SIP", ast_channel_uniqueid(p->owner), p->callid, p->fullcontact, p->peername);
            }
         } else { /* RE-invite */
            if (p->t38.state == T38_DISABLED || p->t38.state == T38_REJECTED) {
               ast_queue_control(p->owner, AST_CONTROL_UPDATE_RTP_PEER);
            } else {
               ast_queue_frame(p->owner, &ast_null_frame);
            }
         }
      } else {
          /* It's possible we're getting an 200 OK after we've tried to disconnect
              by sending CANCEL */
         /* First send ACK, then send bye */
         if (!req->ignore) {
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
         }
      }

      /* Check for Session-Timers related headers */
      if (st_get_mode(p, 0) != SESSION_TIMER_MODE_REFUSE) {
         p_hdrval = (char*)sip_get_header(req, "Session-Expires");
         if (!ast_strlen_zero(p_hdrval)) {
            /* UAS supports Session-Timers */
            enum st_refresher_param st_ref_param;
            int tmp_st_interval = 0;
            rtn = parse_session_expires(p_hdrval, &tmp_st_interval, &st_ref_param);
            if (rtn != 0) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            } else if (tmp_st_interval < st_get_se(p, FALSE)) {
               ast_log(LOG_WARNING, "Got Session-Expires less than local Min-SE in 200 OK, tearing down call\n");
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
            }
            if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAC) {
               p->stimer->st_ref = SESSION_TIMER_REFRESHER_US;
            } else if (st_ref_param == SESSION_TIMER_REFRESHER_PARAM_UAS) {
               p->stimer->st_ref = SESSION_TIMER_REFRESHER_THEM;
            } else {
               ast_log(LOG_WARNING, "Unknown refresher on %s\n", p->callid);
            }
            if (tmp_st_interval) {
               p->stimer->st_interval = tmp_st_interval;
            }
            p->stimer->st_active = TRUE;
            p->stimer->st_active_peer_ua = TRUE;
            start_session_timer(p);
         } else {
            /* UAS doesn't support Session-Timers */
            if (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE) {
               p->stimer->st_ref = SESSION_TIMER_REFRESHER_US;
               p->stimer->st_active_peer_ua = FALSE;
               start_session_timer(p);
            }
         }
      }


      /* If I understand this right, the branch is different for a non-200 ACK only */
      p->invitestate = INV_TERMINATED;
      ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, TRUE);
      check_pendings(p);
      break;

   case 407: /* Proxy authentication */
   case 401: /* Www auth */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->options) {
         p->options->auth_type = resp;
      }

      /* Then we AUTH */
      ast_string_field_set(p, theirtag, NULL);  /* forget their old tag, so we don't match tags when getting response */
      if (!req->ignore) {
         if (p->authtries < MAX_AUTHTRIES) {
            p->invitestate = INV_CALLING;
         }
         if (p->authtries == MAX_AUTHTRIES || do_proxy_auth(p, req, resp, SIP_INVITE, 1)) {
            ast_log(LOG_NOTICE, "Failed to authenticate on INVITE to '%s'\n", sip_get_header(&p->initreq, "From"));
            pvt_set_needdestroy(p, "failed to authenticate on INVITE");
            sip_alreadygone(p);
            if (p->owner) {
               ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
            }
         }
      }
      break;

   case 403: /* Forbidden */
      /* First we ACK */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_log(LOG_WARNING, "Received response: \"Forbidden\" from '%s'\n", sip_get_header(&p->initreq, "From"));
      if (!req->ignore && p->owner) {
         sip_queue_hangup_cause(p, hangup_sip2cause(resp));
      }
      break;

   case 404: /* Not found */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         sip_queue_hangup_cause(p, hangup_sip2cause(resp));
      }
      break;

   case 481: /* Call leg does not exist */
      /* Could be REFER caused INVITE with replaces */
      ast_log(LOG_WARNING, "Re-invite to non-existing call leg on other UA. SIP dialog '%s'. Giving up.\n", p->callid);
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner) {
         ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
      }
      break;

   case 422: /* Session-Timers: Session interval too small */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      ast_string_field_set(p, theirtag, NULL);
      proc_422_rsp(p, req);
      break;

   case 428: /* Use identity header - rfc 4474 - not supported by Asterisk yet */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      append_history(p, "Identity", "SIP identity is required. Not supported by Asterisk.");
      ast_log(LOG_WARNING, "SIP identity required by proxy. SIP dialog '%s'. Giving up.\n", p->callid);
      if (p->owner && !req->ignore) {
         ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
      }
      break;

   case 487: /* Cancelled transaction */
      /* We have sent CANCEL on an outbound INVITE
         This transaction is already scheduled to be killed by sip_hangup().
      */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         ast_queue_hangup_with_cause(p->owner, AST_CAUSE_NORMAL_CLEARING);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us. Queued AST hangup request");
      } else if (!req->ignore) {
         update_call_counter(p, DEC_CALL_LIMIT);
         append_history(p, "Hangup", "Got 487 on CANCEL request from us on call without owner. Killing this dialog.");
      }
      check_pendings(p);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      break;
   case 415: /* Unsupported media type */
   case 488: /* Not acceptable here */
   case 606: /* Not Acceptable */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
         change_t38_state(p, T38_REJECTED);
         /* Try to reset RTP timers */
         /* XXX Why is this commented away??? */
         //ast_rtp_set_rtptimers_onhold(p->rtp);

         /* Trigger a reinvite back to audio */
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else {
         /* We can't set up this call, so give up */
         if (p->owner && !req->ignore) {
            ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
         }
      }
      break;
   case 491: /* Pending */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner && !req->ignore) {
         if (ast_channel_state(p->owner) != AST_STATE_UP) {
            ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
         } else {
            /* This is a re-invite that failed. */
            /* Reset the flag after a while
             */
            int wait;
            /* RFC 3261, if owner of call, wait between 2.1 to 4 seconds,
             * if not owner of call, wait 0 to 2 seconds */
            if (p->outgoing_call) {
               wait = 2100 + ast_random() % 2000;
            } else {
               wait = ast_random() % 2000;
            }
            p->waitid = ast_sched_add(sched, wait, sip_reinvite_retry, dialog_ref(p, "passing dialog ptr into sched structure based on waitid for sip_reinvite_retry."));
            ast_debug(2, "Reinvite race. Scheduled sip_reinvite_retry in %d secs in handle_response_invite (waitid %d, dialog '%s')\n",
                  wait, p->waitid, p->callid);
         }
      }
      break;

   case 408: /* Request timeout */
   case 405: /* Not allowed */
   case 501: /* Not implemented */
      xmitres = transmit_request(p, SIP_ACK, seqno, XMIT_UNRELIABLE, FALSE);
      if (p->owner) {
         ast_queue_hangup_with_cause(p->owner, hangup_sip2cause(resp));
      }
      break;
   }
   if (xmitres == XMIT_ERROR) {
      ast_log(LOG_WARNING, "Could not transmit message in dialog %s\n", p->callid);
   }
}

static void handle_response_message	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23750 of file chan_sip.c.

References ast_log(), ast_sockaddr_stringify(), ast_test_flag, ast_verb, do_message_auth(), LOG_WARNING, mark_method_allowed(), mark_method_unallowed(), pvt_set_needdestroy(), sip_methods, cfsip_methods::text, and text.

Referenced by handle_response().

{
   int sipmethod = SIP_MESSAGE;
   int in_dialog = ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);

   switch (resp) {
   case 401: /* Not www-authorized on SIP method */
   case 407: /* Proxy auth required */
      if (do_message_auth(p, resp, rest, req, seqno) && !in_dialog) {
         pvt_set_needdestroy(p, "MESSAGE authentication failed");
      }
      break;
   case 405: /* Method not allowed */
   case 501: /* Not Implemented */
      mark_method_unallowed(&p->allowed_methods, sipmethod);
      if (p->relatedpeer) {
         mark_method_allowed(&p->relatedpeer->disallowed_methods, sipmethod);
      }
      ast_log(LOG_WARNING, "Host '%s' does not implement '%s'\n",
         ast_sockaddr_stringify(&p->sa), sip_methods[sipmethod].text);
      if (!in_dialog) {
         pvt_set_needdestroy(p, "MESSAGE not implemented or allowed");
      }
      break;
   default:
      if (100 <= resp && resp < 200) {
         /* Must allow provisional responses for out-of-dialog requests. */
      } else if (200 <= resp && resp < 300) {
         p->authtries = 0; /* Reset authentication counter */
         if (!in_dialog) {
            pvt_set_needdestroy(p, "MESSAGE delivery accepted");
         }
      } else if (300 <= resp && resp < 700) {
         ast_verb(3, "Got SIP %s response %d \"%s\" back from host '%s'\n",
            sip_methods[sipmethod].text, resp, rest, ast_sockaddr_stringify(&p->sa));
         if (!in_dialog) {
            pvt_set_needdestroy(p, (300 <= resp && resp < 600)
               ? "MESSAGE delivery failed" : "MESSAGE delivery refused");
         }
      }
      break;
   }
}

static void handle_response_notify	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23219 of file chan_sip.c.

References ast_channel_name(), ast_clear_flag, ast_debug, ast_log(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, do_proxy_auth(), extensionstate_update(), LOG_NOTICE, LOG_WARNING, NONE, pvt_set_needdestroy(), sip_get_header(), state_notify_data::state, and TRUE.

Referenced by handle_response().

{
   switch (resp) {
   case 200:   /* Notify accepted */
      /* They got the notify, this is the end */
      if (p->owner) {
         if (p->refer) {
            ast_log(LOG_NOTICE, "Got OK on REFER Notify message\n");
         } else {
            ast_log(LOG_WARNING, "Notify answer on an owned channel? - %s\n", ast_channel_name(p->owner));
         }
      } else {
         if (p->subscribed == NONE && !p->refer) {
            ast_debug(4, "Got 200 accepted on NOTIFY %s\n", p->callid);
            pvt_set_needdestroy(p, "received 200 response");
         }
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE)) {
            struct state_notify_data data = {
               .state = p->laststate,
               .device_state_info = p->last_device_state_info,
               .presence_state = p->last_presence_state,
               .presence_subtype = p->last_presence_subtype,
               .presence_message = p->last_presence_message,
            };
            /* Ready to send the next state we have on queue */
            ast_clear_flag(&p->flags[1], SIP_PAGE2_STATECHANGEQUEUE);
            extensionstate_update(p->context, p->exten, &data, p, TRUE);
         }
      }
      break;
   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (!p->notify) {
         break; /* Only device notify can use NOTIFY auth */
      }
      ast_string_field_set(p, theirtag, NULL);
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate NOTIFY to %s but we have no matching peer or realm auth!\n", ast_sockaddr_stringify(&p->recv));
         pvt_set_needdestroy(p, "unable to authenticate NOTIFY");
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_NOTIFY, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on NOTIFY to '%s'\n", sip_get_header(&p->initreq, "From"));
         pvt_set_needdestroy(p, "failed to authenticate NOTIFY");
      }
      break;
   case 481: /* Call leg does not exist */
      pvt_set_needdestroy(p, "Received 481 response for NOTIFY");
      break;
   }
}

static void handle_response_peerpoke	(	struct sip_pvt *	p,
		int	resp,
		struct sip_request *	req
	)		[static]

Handle qualification responses (OPTIONS)

Definition at line 23609 of file chan_sip.c.

References ast_check_realtime(), AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_tvdiff_ms(), ast_tvnow(), ast_update_realtime(), DEFAULT_FREQ_NOTOK, EVENT_FLAG_SYSTEM, LOG_NOTICE, manager_event, pvt_set_needdestroy(), register_peer_exten(), SENTINEL, sip_cfg, sip_poke_peer_s(), sip_ref_peer(), sip_unref_peer(), and TRUE.

Referenced by handle_response().

{
   struct sip_peer *peer = /* sip_ref_peer( */ p->relatedpeer /* , "bump refcount on p, as it is being used in this function(handle_response_peerpoke)")*/ ; /* hope this is already refcounted! */
   int statechanged, is_reachable, was_reachable;
   int pingtime = ast_tvdiff_ms(ast_tvnow(), peer->ps);

   /*
    * Compute the response time to a ping (goes in peer->lastms.)
    * -1 means did not respond, 0 means unknown,
    * 1..maxms is a valid response, >maxms means late response.
    */
   if (pingtime < 1) {  /* zero = unknown, so round up to 1 */
      pingtime = 1;
   }

   if (!peer->maxms) { /* this should never happens */
      pvt_set_needdestroy(p, "got OPTIONS response but qualify is not enabled");
      return;
   }

   /* Now determine new state and whether it has changed.
    * Use some helper variables to simplify the writing
    * of the expressions.
    */
   was_reachable = peer->lastms > 0 && peer->lastms <= peer->maxms;
   is_reachable = pingtime <= peer->maxms;
   statechanged = peer->lastms == 0 /* yes, unknown before */
      || was_reachable != is_reachable;

   peer->lastms = pingtime;
   peer->call = dialog_unref(peer->call, "unref dialog peer->call");
   if (statechanged) {
      const char *s = is_reachable ? "Reachable" : "Lagged";
      char str_lastms[20];
      snprintf(str_lastms, sizeof(str_lastms), "%d", pingtime);

      ast_log(LOG_NOTICE, "Peer '%s' is now %s. (%dms / %dms)\n",
         peer->name, s, pingtime, peer->maxms);
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
      if (sip_cfg.peer_rtupdate) {
         ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", str_lastms, SENTINEL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
         "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: %s\r\nTime: %d\r\n",
         peer->name, s, pingtime);
      if (is_reachable && sip_cfg.regextenonqualify)
         register_peer_exten(peer, TRUE);
   }

   pvt_set_needdestroy(p, "got OPTIONS response");

   /* Try again eventually */
   AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
         is_reachable ? peer->qualifyfreq : DEFAULT_FREQ_NOTOK,
         sip_poke_peer_s, peer,
         sip_unref_peer(_data, "removing poke peer ref"),
         sip_unref_peer(peer, "removing poke peer ref"),
         sip_ref_peer(peer, "adding poke peer ref"));
}

static void handle_response_publish	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 22615 of file chan_sip.c.

References ast_assert, ast_copy_string(), ast_log(), ast_string_field_set, ast_strlen_zero(), do_proxy_auth(), LOG_NOTICE, mark_method_unallowed(), pvt_set_needdestroy(), sip_alreadygone(), and sip_get_header().

Referenced by handle_response().

{
   struct sip_epa_entry *epa_entry = p->epa_entry;
   const char *etag = sip_get_header(req, "Sip-ETag");

   ast_assert(epa_entry != NULL);

   if (resp == 401 || resp == 407) {
      ast_string_field_set(p, theirtag, NULL);
      if (p->options) {
         p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
      }
      if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_PUBLISH, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on PUBLISH to '%s'\n", sip_get_header(&p->initreq, "From"));
         pvt_set_needdestroy(p, "Failed to authenticate on PUBLISH");
         sip_alreadygone(p);
      }
      return;
   }

   if (resp == 501 || resp == 405) {
      mark_method_unallowed(&p->allowed_methods, SIP_PUBLISH);
   }

   if (resp == 200) {
      p->authtries = 0;
      /* If I've read section 6, item 6 of RFC 3903 correctly,
       * an ESC will only generate a new etag when it sends a 200 OK
       */
      if (!ast_strlen_zero(etag)) {
         ast_copy_string(epa_entry->entity_tag, etag, sizeof(epa_entry->entity_tag));
      }
      /* The nominal case. Everything went well. Everybody is happy.
       * Each EPA will have a specific action to take as a result of this
       * development, so ... callbacks!
       */
      if (epa_entry->static_data->handle_ok) {
         epa_entry->static_data->handle_ok(p, req, epa_entry);
      }
   } else {
      /* Rather than try to make individual callbacks for each error
       * type, there is just a single error callback. The callback
       * can distinguish between error messages and do what it needs to
       */
      if (epa_entry->static_data->handle_error) {
         epa_entry->static_data->handle_error(p, resp, req, epa_entry);
      }
   }
}

static void handle_response_refer	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23357 of file chan_sip.c.

References AST_CONTROL_CONGESTION, AST_CONTROL_TRANSFER, ast_debug, ast_log(), ast_queue_control(), ast_queue_control_data(), ast_sockaddr_stringify(), ast_strlen_zero(), AST_TRANSFER_FAILED, do_proxy_auth(), LOG_NOTICE, LOG_WARNING, pvt_set_needdestroy(), and sip_get_header().

Referenced by handle_response().

{
   enum ast_control_transfer message = AST_TRANSFER_FAILED;

   /* If no refer structure exists, then do nothing */
   if (!p->refer)
      return;

   switch (resp) {
   case 202:   /* Transfer accepted */
      /* We need  to do something here */
      /* The transferee is now sending INVITE to target */
      p->refer->status = REFER_ACCEPTED;
      /* Now wait for next message */
      ast_debug(3, "Got 202 accepted on transfer\n");
      /* We should hang along, waiting for NOTIFY's here */
      break;

   case 401:   /* Not www-authorized on SIP method */
   case 407:   /* Proxy auth */
      if (ast_strlen_zero(p->authname)) {
         ast_log(LOG_WARNING, "Asked to authenticate REFER to %s but we have no matching peer or realm auth!\n",
            ast_sockaddr_stringify(&p->recv));
         if (p->owner) {
            ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
         }
         pvt_set_needdestroy(p, "unable to authenticate REFER");
      }
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_REFER, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REFER to '%s'\n", sip_get_header(&p->initreq, "From"));
         p->refer->status = REFER_NOAUTH;
         if (p->owner) {
            ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
         }
         pvt_set_needdestroy(p, "failed to authenticate REFER");
      }
      break;
   
   case 405:   /* Method not allowed */
      /* Return to the current call onhold */
      /* Status flag needed to be reset */
      ast_log(LOG_NOTICE, "SIP transfer to %s failed, REFER not allowed. \n", p->refer->refer_to);
      pvt_set_needdestroy(p, "received 405 response");
      p->refer->status = REFER_FAILED;
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;

   case 481: /* Call leg does not exist */

      /* A transfer with Replaces did not work */
      /* OEJ: We should Set flag, cancel the REFER, go back
      to original call - but right now we can't */
      ast_log(LOG_WARNING, "Remote host can't match REFER request to call '%s'. Giving up.\n", p->callid);
      if (p->owner)
         ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
      pvt_set_needdestroy(p, "received 481 response");
      break;

   case 500:   /* Server error */
   case 501:   /* Method not implemented */
      /* Return to the current call onhold */
      /* Status flag needed to be reset */
      ast_log(LOG_NOTICE, "SIP transfer to %s failed, call miserably fails. \n", p->refer->refer_to);
      pvt_set_needdestroy(p, "received 500/501 response");
      p->refer->status = REFER_FAILED;
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   case 603:   /* Transfer declined */
      ast_log(LOG_NOTICE, "SIP transfer to %s declined, call miserably fails. \n", p->refer->refer_to);
      p->refer->status = REFER_FAILED;
      pvt_set_needdestroy(p, "received 603 response");
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   default:
      /* We should treat unrecognized 9xx as 900.  400 is actually
         specified as a possible response, but any 4-6xx is 
         theoretically possible. */

      if (resp < 299) { /* 1xx cases don't get here */
         ast_log(LOG_WARNING, "SIP transfer to %s had unxpected 2xx response (%d), confusion is possible. \n", p->refer->refer_to, resp);
      } else {
         ast_log(LOG_WARNING, "SIP transfer to %s with response (%d). \n", p->refer->refer_to, resp);
      }

      p->refer->status = REFER_FAILED;
      pvt_set_needdestroy(p, "received failure response");
      if (p->owner) {
         ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
      }
      break;
   }
}

static int handle_response_register	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Handle responses on REGISTER to services.

Definition at line 23457 of file chan_sip.c.

References __get_header(), ast_debug, ast_log(), AST_SCHED_DEL_UNREF, AST_SCHED_REPLACE_UNREF, ast_string_field_set, ast_strlen_zero(), ast_tvnow(), default_expiry, do_register_auth(), EVENT_FLAG_SYSTEM, LOG_NOTICE, LOG_WARNING, manager_event, MAX, pvt_set_needdestroy(), REG_STATE_NOAUTH, REG_STATE_REGISTERED, REG_STATE_REJECTED, REG_STATE_UNREGISTERED, registry_addref(), registry_unref(), regstate2str(), S_OR, sip_get_header(), sip_reregister(), and transmit_register().

Referenced by handle_response().

{
   int expires, expires_ms;
   struct sip_registry *r;
   r=p->registry;
   
   switch (resp) {
   case 401:   /* Unauthorized */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s@%s' (Tries %d)\n", p->registry->username, p->registry->hostname, p->authtries);
         pvt_set_needdestroy(p, "failed to authenticate REGISTER");
      }
      break;
   case 403:   /* Forbidden */
      if (global_reg_retry_403) {
         ast_log(LOG_NOTICE, "Treating 403 response to REGISTER as non-fatal for %s@%s\n",
            p->registry->username, p->registry->hostname);
         ast_string_field_set(r, nonce, "");
         ast_string_field_set(p, nonce, "");
         break;
      }
      ast_log(LOG_WARNING, "Forbidden - wrong password on authentication for REGISTER for '%s' to '%s'\n", p->registry->username, p->registry->hostname);
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 403"));
      r->regstate = REG_STATE_NOAUTH;
      pvt_set_needdestroy(p, "received 403 response");
      break;
   case 404:   /* Not found */
      ast_log(LOG_WARNING, "Got 404 Not found on SIP register to service %s@%s, giving up\n", p->registry->username, p->registry->hostname);
      pvt_set_needdestroy(p, "received 404 response");
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 404");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 404"));
      break;
   case 407:   /* Proxy auth */
      if (p->authtries == MAX_AUTHTRIES || do_register_auth(p, req, resp)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on REGISTER to '%s' (tries '%d')\n", sip_get_header(&p->initreq, "From"), p->authtries);
         pvt_set_needdestroy(p, "failed to authenticate REGISTER");
      }
      break;
   case 408:   /* Request timeout */
      /* Got a timeout response, so reset the counter of failed responses */
      if (r) {
         r->regattempts = 0;
      } else {
         ast_log(LOG_WARNING, "Got a 408 response to our REGISTER on call %s after we had destroyed the registry object\n", p->callid);
      }
      break;
   case 423:   /* Interval too brief */
      r->expiry = atoi(sip_get_header(req, "Min-Expires"));
      ast_log(LOG_WARNING, "Got 423 Interval too brief for service %s@%s, minimum is %d seconds\n", p->registry->username, p->registry->hostname, r->expiry);
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 423"));
      if (r->call) {
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 423");
         pvt_set_needdestroy(p, "received 423 response");
      }
      if (r->expiry > max_expiry) {
         ast_log(LOG_WARNING, "Required expiration time from %s@%s is too high, giving up\n", p->registry->username, p->registry->hostname);
         r->expiry = r->configured_expiry;
         r->regstate = REG_STATE_REJECTED;
      } else {
         r->regstate = REG_STATE_UNREGISTERED;
         transmit_register(r, SIP_REGISTER, NULL, NULL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
      break;
   case 479:   /* SER: Not able to process the URI - address is wrong in register*/
      ast_log(LOG_WARNING, "Got error 479 on register to %s@%s, giving up (check config)\n", p->registry->username, p->registry->hostname);
      pvt_set_needdestroy(p, "received 479 response");
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 479");
      r->regstate = REG_STATE_REJECTED;
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 479"));
      break;
   case 200:   /* 200 OK */
      if (!r) {
         ast_log(LOG_WARNING, "Got 200 OK on REGISTER, but there isn't a registry entry for '%s' (we probably already got the OK)\n", S_OR(p->peername, p->username));
         pvt_set_needdestroy(p, "received erroneous 200 response");
         return 0;
      }

      r->regstate = REG_STATE_REGISTERED;
      r->regtime = ast_tvnow();     /* Reset time of last successful registration */
      manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
      r->regattempts = 0;
      ast_debug(1, "Registration successful\n");
      if (r->timeout > -1) {
         ast_debug(1, "Cancelling timeout %d\n", r->timeout);
      }
      AST_SCHED_DEL_UNREF(sched, r->timeout, registry_unref(r, "reg ptr unref from handle_response_register 200"));
      if (r->call)
         r->call = dialog_unref(r->call, "unsetting registry->call pointer-- case 200");
      p->registry = registry_unref(p->registry, "unref registry entry p->registry");

      /* destroy dialog now to avoid interference with next register */
      pvt_set_needdestroy(p, "Registration successfull");

      /* set us up for re-registering
       * figure out how long we got registered for
       * according to section 6.13 of RFC, contact headers override
       * expires headers, so check those first */
      expires = 0;

      /* XXX todo: try to save the extra call */
      if (!ast_strlen_zero(sip_get_header(req, "Contact"))) {
         const char *contact = NULL;
         const char *tmptmp = NULL;
         int start = 0;
         for(;;) {
            contact = __get_header(req, "Contact", &start);
            /* this loop ensures we get a contact header about our register request */
            if(!ast_strlen_zero(contact)) {
               if( (tmptmp=strstr(contact, p->our_contact))) {
                  contact=tmptmp;
                  break;
               }
            } else
               break;
         }
         tmptmp = strcasestr(contact, "expires=");
         if (tmptmp) {
            if (sscanf(tmptmp + 8, "%30d", &expires) != 1) {
               expires = 0;
            }
         }
         
      }
      if (!expires)
         expires=atoi(sip_get_header(req, "expires"));
      if (!expires)
         expires=default_expiry;
      
      expires_ms = expires * 1000;
      if (expires <= EXPIRY_GUARD_LIMIT)
         expires_ms -= MAX((expires_ms * EXPIRY_GUARD_PCT), EXPIRY_GUARD_MIN);
      else
         expires_ms -= EXPIRY_GUARD_SECS * 1000;
      if (sipdebug)
         ast_log(LOG_NOTICE, "Outbound Registration: Expiry for %s is %d sec (Scheduling reregistration in %d s)\n", r->hostname, expires, expires_ms/1000);
      
      r->refresh= (int) expires_ms / 1000;
      
      /* Schedule re-registration before we expire */
      AST_SCHED_REPLACE_UNREF(r->expire, sched, expires_ms, sip_reregister, r,
                        registry_unref(_data,"unref in REPLACE del fail"),
                        registry_unref(r,"unref in REPLACE add fail"),
                        registry_addref(r,"The Addition side of REPLACE"));
   }
   return 1;
}

static void handle_response_subscribe	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Definition at line 23271 of file chan_sip.c.

References ao2_callback, ao2_ref, ast_cc_monitor_failed(), ast_debug, ast_free, ast_log(), ast_sched_add(), ast_string_field_set, ASTOBJ_REF, ASTOBJ_UNREF, do_proxy_auth(), find_sip_monitor_instance_by_subscription_pvt(), LOG_NOTICE, LOG_WARNING, pvt_set_needdestroy(), set_pvt_allowed_methods(), sip_alreadygone(), sip_get_header(), sip_subscribe_mwi_destroy(), sip_subscribe_mwi_do(), and transmit_response_with_date().

Referenced by handle_response().

{
   if (p->subscribed == CALL_COMPLETION) {
      struct sip_monitor_instance *monitor_instance;

      if (resp < 300) {
         return;
      }

      /* Final failure response received. */
      monitor_instance = ao2_callback(sip_monitor_instances, 0,
         find_sip_monitor_instance_by_subscription_pvt, p);
      if (monitor_instance) {
         ast_cc_monitor_failed(monitor_instance->core_id,
            monitor_instance->device_name,
            "Received error response to our SUBSCRIBE");
      }
      return;
   }

   if (p->subscribed != MWI_NOTIFICATION) {
      return;
   }
   if (!p->mwi) {
      return;
   }

   switch (resp) {
   case 200: /* Subscription accepted */
      ast_debug(3, "Got 200 OK on subscription for MWI\n");
      set_pvt_allowed_methods(p, req);
      if (p->options) {
         if (p->options->outboundproxy) {
            ao2_ref(p->options->outboundproxy, -1);
         }
         ast_free(p->options);
         p->options = NULL;
      }
      p->mwi->subscribed = 1;
      if ((p->mwi->resub = ast_sched_add(sched, mwi_expiry * 1000, sip_subscribe_mwi_do, ASTOBJ_REF(p->mwi))) < 0) {
         ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      }
      break;
   case 401:
   case 407:
      ast_string_field_set(p, theirtag, NULL);
      if (p->authtries > 1 || do_proxy_auth(p, req, resp, SIP_SUBSCRIBE, 0)) {
         ast_log(LOG_NOTICE, "Failed to authenticate on SUBSCRIBE to '%s'\n", sip_get_header(&p->initreq, "From"));
         p->mwi->call = NULL;
         ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
         pvt_set_needdestroy(p, "failed to authenticate SUBSCRIBE");
      }
      break;
   case 403:
      transmit_response_with_date(p, "200 OK", req);
      ast_log(LOG_WARNING, "Authentication failed while trying to subscribe for MWI.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 403 response");
      sip_alreadygone(p);
      break;
   case 404:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that a mailbox may not have been configured.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 404 response");
      break;
   case 481:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side said that our dialog did not exist.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 481 response");
      break;
   case 500:
   case 501:
      ast_log(LOG_WARNING, "Subscription failed for MWI. The remote side may have suffered a heart attack.\n");
      p->mwi->call = NULL;
      ASTOBJ_UNREF(p->mwi, sip_subscribe_mwi_destroy);
      pvt_set_needdestroy(p, "received 500/501 response");
      break;
   }
}

static void handle_response_update	(	struct sip_pvt *	p,
		int	resp,
		const char *	rest,
		struct sip_request *	req,
		uint32_t	seqno
	)		[static]

Handle authentication challenge for SIP UPDATE.

This function is only called upon the receipt of a 401/407 response to an UPDATE.

Definition at line 22555 of file chan_sip.c.

References ast_log(), do_proxy_auth(), LOG_NOTICE, and sip_get_header().

Referenced by handle_response().

{
   if (p->options) {
      p->options->auth_type = (resp == 401 ? WWW_AUTH : PROXY_AUTH);
   }
   if ((p->authtries == MAX_AUTHTRIES) || do_proxy_auth(p, req, resp, SIP_UPDATE, 1)) {
      ast_log(LOG_NOTICE, "Failed to authenticate on UPDATE to '%s'\n", sip_get_header(&p->initreq, "From"));
   }
}

static int handle_sip_publish_initial	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const int	expires
	)		[static]

Definition at line 27419 of file chan_sip.c.

References ao2_ref, event_state_compositor::callbacks, create_esc_entry(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = create_esc_entry(esc, req, expires);
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "503 Internal Server Failure", req);
      return -1;
   }

   if (esc->callbacks->initial_handler) {
      res = esc->callbacks->initial_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 0);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_modify	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag,
		const int	expires
	)		[static]

Definition at line 27469 of file chan_sip.c.

References ao2_ref, AST_SCHED_REPLACE_UNREF, event_state_compositor::callbacks, get_esc_entry(), publish_expire(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int expires_ms = expires * 1000;
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
         ao2_ref(_data, -1),
         ao2_ref(esc_entry, -1),
         ao2_ref(esc_entry, +1));

   if (esc->callbacks->modify_handler) {
      res = esc->callbacks->modify_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_refresh	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag,
		const int	expires
	)		[static]

Definition at line 27441 of file chan_sip.c.

References ao2_ref, AST_SCHED_REPLACE_UNREF, event_state_compositor::callbacks, get_esc_entry(), publish_expire(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int expires_ms = expires * 1000;
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_REPLACE_UNREF(esc_entry->sched_id, sched, expires_ms, publish_expire, esc_entry,
         ao2_ref(_data, -1),
         ao2_ref(esc_entry, -1),
         ao2_ref(esc_entry, +1));

   if (esc->callbacks->refresh_handler) {
      res = esc->callbacks->refresh_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_sip_publish_remove	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct event_state_compositor *	esc,
		const char *const	etag
	)		[static]

Definition at line 27497 of file chan_sip.c.

References ao2_ref, ao2_unlink, AST_SCHED_DEL, event_state_compositor::callbacks, event_state_compositor::compositor, get_esc_entry(), transmit_response(), and transmit_response_with_sip_etag().

Referenced by handle_request_publish().

{
   struct sip_esc_entry *esc_entry = get_esc_entry(etag, esc);
   int res = 0;

   if (!esc_entry) {
      transmit_response(p, "412 Conditional Request Failed", req);
      return -1;
   }

   AST_SCHED_DEL(sched, esc_entry->sched_id);
   /* Scheduler's ref of the esc_entry */
   ao2_ref(esc_entry, -1);

   if (esc->callbacks->remove_handler) {
      res = esc->callbacks->remove_handler(p, req, esc, esc_entry);
   }

   if (!res) {
      transmit_response_with_sip_etag(p, "200 OK", req, esc_entry, 1);
   }

   /* Ref from finding the esc_entry earlier in function */
   ao2_unlink(esc->compositor, esc_entry);
   ao2_ref(esc_entry, -1);
   return res;
}

static int handle_t38_options	(	struct ast_flags *	flags,
		struct ast_flags *	mask,
		struct ast_variable *	v,
		int *	maxdatagram
	)		[static]

Handle T.38 configuration options common to users and peers.

Returns:

non-zero if any config options were handled, zero otherwise

Definition at line 30072 of file chan_sip.c.

References ast_clear_flag, ast_log(), ast_set2_flag, ast_set_flag, ast_true(), global_t38_maxdatagram, ast_variable::lineno, LOG_WARNING, ast_variable::name, ast_variable::value, and word.

Referenced by build_peer(), and reload_config().

{
   int res = 1;

   if (!strcasecmp(v->name, "t38pt_udptl")) {
      char *buf = ast_strdupa(v->value);
      char *word, *next = buf;

      ast_set_flag(&mask[1], SIP_PAGE2_T38SUPPORT);

      while ((word = strsep(&next, ","))) {
         if (ast_true(word) || !strcasecmp(word, "fec")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_FEC);
         } else if (!strcasecmp(word, "redundancy")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY);
         } else if (!strcasecmp(word, "none")) {
            ast_clear_flag(&flags[1], SIP_PAGE2_T38SUPPORT);
            ast_set_flag(&flags[1], SIP_PAGE2_T38SUPPORT_UDPTL);
         } else if (!strncasecmp(word, "maxdatagram=", 12)) {
            if (sscanf(&word[12], "%30u", maxdatagram) != 1) {
               ast_log(LOG_WARNING, "Invalid maxdatagram '%s' at line %d of %s\n", v->value, v->lineno, config);
               *maxdatagram = global_t38_maxdatagram;
            }
         }
      }
   } else if (!strcasecmp(v->name, "t38pt_usertpsource")) {
      ast_set_flag(&mask[1], SIP_PAGE2_UDPTL_DESTINATION);
      ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_UDPTL_DESTINATION);
   } else {
      res = 0;
   }

   return res;
}

const char* hangup_cause2sip

(

int

cause

)

Convert Asterisk hangup causes to SIP codes.

 Possible values from causes.h
        AST_CAUSE_NOTDEFINED    AST_CAUSE_NORMAL        AST_CAUSE_BUSY
        AST_CAUSE_FAILURE       AST_CAUSE_CONGESTION    AST_CAUSE_UNALLOCATED

	In addition to these, a lot of PRI codes is defined in causes.h
	...should we take care of them too ?

	Quote RFC 3398

   ISUP Cause value                        SIP response
   ----------------                        ------------
   1  unallocated number                   404 Not Found
   2  no route to network                  404 Not found
   3  no route to destination              404 Not found
   16 normal call clearing                 --- (*)
   17 user busy                            486 Busy here
   18 no user responding                   408 Request Timeout
   19 no answer from the user              480 Temporarily unavailable
   20 subscriber absent                    480 Temporarily unavailable
   21 call rejected                        403 Forbidden (+)
   22 number changed (w/o diagnostic)      410 Gone
   22 number changed (w/ diagnostic)       301 Moved Permanently
   23 redirection to new destination       410 Gone
   26 non-selected user clearing           404 Not Found (=)
   27 destination out of order             502 Bad Gateway
   28 address incomplete                   484 Address incomplete
   29 facility rejected                    501 Not implemented
   31 normal unspecified                   480 Temporarily unavailable

Definition at line 6959 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CHAN_NOT_IMPLEMENTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_ROUTE_TRANSIT_NET, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL_UNSPECIFIED, AST_CAUSE_NOTDEFINED, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNALLOCATED, AST_CAUSE_UNREGISTERED, AST_CAUSE_USER_BUSY, and ast_debug.

Referenced by sip_hangup().

{
   switch (cause) {
      case AST_CAUSE_UNALLOCATED:      /* 1 */
      case AST_CAUSE_NO_ROUTE_DESTINATION:   /* 3 IAX2: Can't find extension in context */
      case AST_CAUSE_NO_ROUTE_TRANSIT_NET:   /* 2 */
         return "404 Not Found";
      case AST_CAUSE_CONGESTION:    /* 34 */
      case AST_CAUSE_SWITCH_CONGESTION:   /* 42 */
         return "503 Service Unavailable";
      case AST_CAUSE_NO_USER_RESPONSE: /* 18 */
         return "408 Request Timeout";
      case AST_CAUSE_NO_ANSWER:     /* 19 */
      case AST_CAUSE_UNREGISTERED:        /* 20 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_CALL_REJECTED:    /* 21 */
         return "403 Forbidden";
      case AST_CAUSE_NUMBER_CHANGED:      /* 22 */
         return "410 Gone";
      case AST_CAUSE_NORMAL_UNSPECIFIED:  /* 31 */
         return "480 Temporarily unavailable";
      case AST_CAUSE_INVALID_NUMBER_FORMAT:
         return "484 Address incomplete";
      case AST_CAUSE_USER_BUSY:
         return "486 Busy here";
      case AST_CAUSE_FAILURE:
         return "500 Server internal failure";
      case AST_CAUSE_FACILITY_REJECTED:   /* 29 */
         return "501 Not Implemented";
      case AST_CAUSE_CHAN_NOT_IMPLEMENTED:
         return "503 Service Unavailable";
      /* Used in chan_iax2 */
      case AST_CAUSE_DESTINATION_OUT_OF_ORDER:
         return "502 Bad Gateway";
      case AST_CAUSE_BEARERCAPABILITY_NOTAVAIL: /* Can't find codec to connect to host */
         return "488 Not Acceptable Here";
      case AST_CAUSE_INTERWORKING:  /* Unspecified Interworking issues */
         return "500 Network error";

      case AST_CAUSE_NOTDEFINED:
      default:
         ast_debug(1, "AST hangup cause %d (no match found in SIP)\n", cause);
         return NULL;
   }

   /* Never reached */
   return 0;
}

int hangup_sip2cause

(

int

cause

)

Convert SIP hangup causes to Asterisk hangup causes.

Definition at line 6837 of file chan_sip.c.

References AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_BUSY, AST_CAUSE_CALL_REJECTED, AST_CAUSE_CONGESTION, AST_CAUSE_DESTINATION_OUT_OF_ORDER, AST_CAUSE_FACILITY_REJECTED, AST_CAUSE_FAILURE, AST_CAUSE_INTERWORKING, AST_CAUSE_INVALID_NUMBER_FORMAT, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NO_USER_RESPONSE, AST_CAUSE_NORMAL, AST_CAUSE_NORMAL_TEMPORARY_FAILURE, AST_CAUSE_NUMBER_CHANGED, AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE, AST_CAUSE_UNALLOCATED, and AST_CAUSE_USER_BUSY.

Referenced by __transmit_response(), handle_incoming(), handle_response(), and handle_response_invite().

{
   /* Possible values taken from causes.h */

   switch(cause) {
      case 401:   /* Unauthorized */
         return AST_CAUSE_CALL_REJECTED;
      case 403:   /* Not found */
         return AST_CAUSE_CALL_REJECTED;
      case 404:   /* Not found */
         return AST_CAUSE_UNALLOCATED;
      case 405:   /* Method not allowed */
         return AST_CAUSE_INTERWORKING;
      case 407:   /* Proxy authentication required */
         return AST_CAUSE_CALL_REJECTED;
      case 408:   /* No reaction */
         return AST_CAUSE_NO_USER_RESPONSE;
      case 409:   /* Conflict */
         return AST_CAUSE_NORMAL_TEMPORARY_FAILURE;
      case 410:   /* Gone */
         return AST_CAUSE_NUMBER_CHANGED;
      case 411:   /* Length required */
         return AST_CAUSE_INTERWORKING;
      case 413:   /* Request entity too large */
         return AST_CAUSE_INTERWORKING;
      case 414:   /* Request URI too large */
         return AST_CAUSE_INTERWORKING;
      case 415:   /* Unsupported media type */
         return AST_CAUSE_INTERWORKING;
      case 420:   /* Bad extension */
         return AST_CAUSE_NO_ROUTE_DESTINATION;
      case 480:   /* No answer */
         return AST_CAUSE_NO_ANSWER;
      case 481:   /* No answer */
         return AST_CAUSE_INTERWORKING;
      case 482:   /* Loop detected */
         return AST_CAUSE_INTERWORKING;
      case 483:   /* Too many hops */
         return AST_CAUSE_NO_ANSWER;
      case 484:   /* Address incomplete */
         return AST_CAUSE_INVALID_NUMBER_FORMAT;
      case 485:   /* Ambiguous */
         return AST_CAUSE_UNALLOCATED;
      case 486:   /* Busy everywhere */
         return AST_CAUSE_BUSY;
      case 487:   /* Request terminated */
         return AST_CAUSE_INTERWORKING;
      case 488:   /* No codecs approved */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      case 491:   /* Request pending */
         return AST_CAUSE_INTERWORKING;
      case 493:   /* Undecipherable */
         return AST_CAUSE_INTERWORKING;
      case 500:   /* Server internal failure */
         return AST_CAUSE_FAILURE;
      case 501:   /* Call rejected */
         return AST_CAUSE_FACILITY_REJECTED;
      case 502:
         return AST_CAUSE_DESTINATION_OUT_OF_ORDER;
      case 503:   /* Service unavailable */
         return AST_CAUSE_CONGESTION;
      case 504:   /* Gateway timeout */
         return AST_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
      case 505:   /* SIP version not supported */
         return AST_CAUSE_INTERWORKING;
      case 600:   /* Busy everywhere */
         return AST_CAUSE_USER_BUSY;
      case 603:   /* Decline */
         return AST_CAUSE_CALL_REJECTED;
      case 604:   /* Does not exist anywhere */
         return AST_CAUSE_UNALLOCATED;
      case 606:   /* Not acceptable */
         return AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;
      default:
         if (cause < 500 && cause >= 400) {
            /* 4xx class error that is unknown - someting wrong with our request */
            return AST_CAUSE_INTERWORKING;
         } else if (cause < 600 && cause >= 500) {
            /* 5xx class error - problem in the remote end */
            return AST_CAUSE_CONGESTION;
         } else if (cause < 700 && cause >= 600) {
            /* 6xx - global errors in the 4xx class */
            return AST_CAUSE_INTERWORKING;
         }
         return AST_CAUSE_NORMAL;
   }
   /* Never reached */
   return 0;
}

static int has_media_stream	(	struct sip_pvt *	p,
		enum media_type	m
	)		[static]

Check the media stream list to see if the given type already exists.

Definition at line 9892 of file chan_sip.c.

References AST_LIST_TRAVERSE.

Referenced by process_sdp().

{
   struct offered_media *offer = NULL;
   AST_LIST_TRAVERSE(&p->offered_media, offer, next) {
      if (m == offer->type) {
         return 1;
      }
   }
   return 0;
}

static int init_req	(	struct sip_request *	req,
		int	sipmethod,
		const char *	recip
	)		[static]

Initialize SIP request.

Definition at line 11703 of file chan_sip.c.

References ast_free, ast_str_create(), ast_str_set(), sip_methods, and cfsip_methods::text.

Referenced by initreqprep(), reqprep(), and transmit_register().

{
   /* Initialize a request */
   memset(req, 0, sizeof(*req));
   if (!(req->data = ast_str_create(SIP_MIN_PACKET)))
      goto e_return;
   if (!(req->content = ast_str_create(SIP_MIN_PACKET)))
      goto e_free_data;
   req->method = sipmethod;
   req->header[0] = 0;
   ast_str_set(&req->data, 0, "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
   req->headers++;
   return 0;

e_free_data:
   ast_free(req->data);
   req->data = NULL;
e_return:
   return -1;
}

static int init_resp	(	struct sip_request *	resp,
		const char *	msg
	)		[static]

Initialize SIP response, based on SIP request.

Definition at line 11681 of file chan_sip.c.

References ast_free, ast_str_create(), and ast_str_set().

Referenced by respprep().

{
   /* Initialize a response */
   memset(resp, 0, sizeof(*resp));
   resp->method = SIP_RESPONSE;
   if (!(resp->data = ast_str_create(SIP_MIN_PACKET)))
      goto e_return;
   if (!(resp->content = ast_str_create(SIP_MIN_PACKET)))
      goto e_free_data;
   resp->header[0] = 0;
   ast_str_set(&resp->data, 0, "SIP/2.0 %s\r\n", msg);
   resp->headers++;
   return 0;

e_free_data:
   ast_free(resp->data);
   resp->data = NULL;
e_return:
   return -1;
}

static int initialize_escs

(

void

)

[static]

Definition at line 1126 of file chan_sip.c.

References ao2_container_alloc, ARRAY_LEN, esc_cmp_fn(), esc_hash_fn(), and event_state_compositors.

Referenced by load_module().

{
   int i, res = 0;
   for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
      if (!((event_state_compositors[i].compositor) =
               ao2_container_alloc(ESC_MAX_BUCKETS, esc_hash_fn, esc_cmp_fn))) {
         res = -1;
      }
   }
   return res;
}

static void initialize_initreq	(	struct sip_pvt *	p,
		struct sip_request *	req
	)		[static]

Initialize the initital request packet in the pvt structure. This packet is used for creating replies and future requests in a dialog.

Definition at line 3514 of file chan_sip.c.

References ast_debug, ast_verbose(), copy_request(), parse_request(), sip_methods, and cfsip_methods::text.

Referenced by transmit_invite(), transmit_message(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), transmit_reinvite_with_sdp(), and update_connectedline().

{
   if (p->initreq.headers) {
      ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
   } else {
      ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
   }
   /* Use this as the basis */
   copy_request(&p->initreq, req);
   parse_request(&p->initreq);
   if (req->debug) {
      ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
   }
}

static int initialize_udptl

(

struct sip_pvt *

p

)

[static]

Definition at line 7619 of file chan_sip.c.

References ast_channel_set_fd(), ast_clear_flag, ast_debug, ast_log(), AST_LOG_WARNING, ast_test_flag, ast_udptl_fd(), ast_udptl_new_with_bindaddr(), ast_udptl_setnat(), ast_udptl_setqos(), bindaddr, global_t38_maxdatagram, and set_t38_capabilities().

Referenced by process_sdp(), process_sdp_a_image(), and sip_indicate().

{
   int natflags = ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP);

   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) {
      return 1;
   }

   /* If we've already initialized T38, don't take any further action */
   if (p->udptl) {
      return 0;
   }

   /* T38 can be supported by this dialog, create it and set the derived properties */
   if ((p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, &bindaddr))) {
      if (p->owner) {
         ast_channel_set_fd(p->owner, 5, ast_udptl_fd(p->udptl));
      }

      ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
      p->t38_maxdatagram = p->relatedpeer ? p->relatedpeer->t38_maxdatagram : global_t38_maxdatagram;
      set_t38_capabilities(p);

      ast_debug(1, "Setting NAT on UDPTL to %s\n", natflags ? "On" : "Off");
      ast_udptl_setnat(p->udptl, natflags);
   } else {
      ast_log(AST_LOG_WARNING, "UDPTL creation failed - disabling T38 for this dialog\n");
      ast_clear_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT);
      return 1;
   }

   return 0;
}

static void initreqprep	(	struct sip_request *	req,
		struct sip_pvt *	p,
		int	sipmethod,
		const char *const	explicit_uri
	)		[static]

Initiate new SIP request to peer/user.

Todo:

Need to add back the VXML URL here at some point, possibly use build_string for all this junk

Definition at line 13809 of file chan_sip.c.

References add_header(), add_max_forwards(), add_route(), ast_channel_connected_effective_id(), ast_copy_string(), AST_DIGIT_ANYNUM, ast_escape_quoted(), ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_sockaddr_port, ast_sockaddr_stringify_host_remote(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_str_set(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_uri_encode(), ast_uri_sip_user, build_contact(), default_callerid, exten, init_req(), ast_party_id::name, ast_party_id::number, ourport, sip_cfg, sip_methods, sip_standard_port(), ast_party_name::str, ast_party_number::str, cfsip_methods::text, ast_party_name::valid, and ast_party_number::valid.

Referenced by transmit_invite(), transmit_message(), and transmit_notify_with_mwi().

{
   struct ast_str *invite = ast_str_alloca(256);
   char from[256];
   char to[256];
   char tmp_n[SIPBUFSIZE/2];  /* build a local copy of 'n' if needed */
   char tmp_l[SIPBUFSIZE/2];  /* build a local copy of 'l' if needed */
   const char *l = NULL;   /* XXX what is this, exactly ? */
   const char *n = NULL;   /* XXX what is this, exactly ? */
   const char *d = NULL;   /* domain in from header */
   const char *urioptions = "";
   int ourport;
   int cid_has_name = 1;
   int cid_has_num = 1;
   struct ast_party_id connected_id;

   if (ast_test_flag(&p->flags[0], SIP_USEREQPHONE)) {
      const char *s = p->username;  /* being a string field, cannot be NULL */

      /* Test p->username against allowed characters in AST_DIGIT_ANY
         If it matches the allowed characters list, then sipuser = ";user=phone"
         If not, then sipuser = ""
      */
      /* + is allowed in first position in a tel: uri */
      if (*s == '+')
         s++;
      for (; *s; s++) {
         if (!strchr(AST_DIGIT_ANYNUM, *s) )
            break;
      }
      /* If we have only digits, add ;user=phone to the uri */
      if (!*s)
         urioptions = ";user=phone";
   }


   snprintf(p->lastmsg, sizeof(p->lastmsg), "Init: %s", sip_methods[sipmethod].text);

   if (ast_strlen_zero(p->fromdomain)) {
      d = ast_sockaddr_stringify_host_remote(&p->ourip);
   }
   if (p->owner) {
      connected_id = ast_channel_connected_effective_id(p->owner);

      if ((ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) == AST_PRES_ALLOWED) {
         if (connected_id.number.valid) {
            l = connected_id.number.str;
         }
         if (connected_id.name.valid) {
            n = connected_id.name.str;
         }
      } else {
         /* Even if we are using RPID, we shouldn't leak information in the From if the user wants
          * their callerid restricted */
         l = "anonymous";
         n = CALLERID_UNKNOWN;
         d = FROMDOMAIN_INVALID;
      }
   }

   /* Hey, it's a NOTIFY! See if they've configured a mwi_from.
    * XXX Right now, this logic works because the only place that mwi_from
    * is set on the sip_pvt is in sip_send_mwi_to_peer. If things changed, then
    * we might end up putting the mwi_from setting into other types of NOTIFY
    * messages as well.
    */
   if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->mwi_from)) {
      l = p->mwi_from;
   }

   if (ast_strlen_zero(l)) {
      cid_has_num = 0;
      l = default_callerid;
   }
   if (ast_strlen_zero(n)) {
      cid_has_name = 0;
      n = l;
   }

   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromuser))
      l = p->fromuser;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromuser, l);

   /* Allow user to be overridden */
   if (!ast_strlen_zero(p->fromname))
      n = p->fromname;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromname, n);

   /* Allow domain to be overridden */
   if (!ast_strlen_zero(p->fromdomain))
      d = p->fromdomain;
   else /* Save for any further attempts */
      ast_string_field_set(p, fromdomain, d);

   ast_copy_string(tmp_l, l, sizeof(tmp_l));
   if (sip_cfg.pedanticsipchecking) {
      ast_escape_quoted(n, tmp_n, sizeof(tmp_n));
      n = tmp_n;
      ast_uri_encode(l, tmp_l, sizeof(tmp_l), ast_uri_sip_user);
   }

   ourport = (p->fromdomainport) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);

   /* If a caller id name was specified, add a display name. */
   if (cid_has_name || !cid_has_num) {
      snprintf(from, sizeof(from), "\"%s\" ", n);
   } else {
      from[0] = '\0';
   }

   if (!sip_standard_port(p->socket.type, ourport)) {
      size_t offset = strlen(from);
      snprintf(&from[offset], sizeof(from) - offset, "<sip:%s@%s:%d>;tag=%s", tmp_l, d, ourport, p->tag);
   } else {
      size_t offset = strlen(from);
      snprintf(&from[offset], sizeof(from) - offset, "<sip:%s@%s>;tag=%s", tmp_l, d, p->tag);
   }

   if (!ast_strlen_zero(explicit_uri)) {
      ast_str_set(&invite, 0, "%s", explicit_uri);
   } else {
      /* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
      if (!ast_strlen_zero(p->fullcontact)) {
         /* If we have full contact, trust it */
         ast_str_append(&invite, 0, "%s", p->fullcontact);
      } else {
         /* Otherwise, use the username while waiting for registration */
         ast_str_append(&invite, 0, "sip:");
         if (!ast_strlen_zero(p->username)) {
            n = p->username;
            if (sip_cfg.pedanticsipchecking) {
               ast_uri_encode(n, tmp_n, sizeof(tmp_n), ast_uri_sip_user);
               n = tmp_n;
            }
            ast_str_append(&invite, 0, "%s@", n);
         }
         ast_str_append(&invite, 0, "%s", p->tohost);
         if (p->portinuri) {
            ast_str_append(&invite, 0, ":%d", ast_sockaddr_port(&p->sa));
         }
         ast_str_append(&invite, 0, "%s", urioptions);
      }
   }

   /* If custom URI options have been provided, append them */
   if (p->options && !ast_strlen_zero(p->options->uri_options))
      ast_str_append(&invite, 0, ";%s", p->options->uri_options);
   
   /* This is the request URI, which is the next hop of the call
      which may or may not be the destination of the call
   */
   ast_string_field_set(p, uri, ast_str_buffer(invite));

   if (!ast_strlen_zero(p->todnid)) {
      /*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
      if (!strchr(p->todnid, '@')) {
         /* We have no domain in the dnid */
         snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      } else {
         snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
      }
   } else {
      if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
         /* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
         snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
      } else if (p->options && p->options->vxml_url) {
         /* If there is a VXML URL append it to the SIP URL */
         snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
      } else {
         snprintf(to, sizeof(to), "<%s>", p->uri);
      }
   }

   init_req(req, sipmethod, p->uri);
   /* now tmp_n is available so reuse it to build the CSeq */
   snprintf(tmp_n, sizeof(tmp_n), "%u %s", ++p->ocseq, sip_methods[sipmethod].text);

   add_header(req, "Via", p->via);
   add_max_forwards(p, req);
   /* This will be a no-op most of the time. However, under certain circumstances,
    * NOTIFY messages will use this function for preparing the request and should
    * have Route headers present.
    */
   add_route(req, p->route);

   add_header(req, "From", from);
   add_header(req, "To", to);
   ast_string_field_set(p, exten, l);
   build_contact(p);
   add_header(req, "Contact", p->our_contact);
   add_header(req, "Call-ID", p->callid);
   add_header(req, "CSeq", tmp_n);
   if (!ast_strlen_zero(global_useragent)) {
      add_header(req, "User-Agent", global_useragent);
   }
}

static const char * insecure2str

(

int

mode

)

[static]

Convert Insecure setting to printable string.

Definition at line 19439 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer().

{
   return map_x_s(insecurestr, mode, "<error>");
}

static int interpret_t38_parameters	(	struct sip_pvt *	p,
		const struct ast_control_t38_parameters *	parameters
	)		[static]

Helper function which updates T.38 capability information and triggers a reinvite.

Definition at line 7525 of file chan_sip.c.

References AST_CONTROL_T38_PARAMETERS, ast_queue_control_data(), AST_SCHED_DEL_UNREF, ast_set_flag, AST_T38_NEGOTIATED, AST_T38_REFUSED, AST_T38_REQUEST_NEGOTIATE, AST_T38_REQUEST_PARMS, AST_T38_REQUEST_TERMINATE, AST_T38_TERMINATED, ast_test_flag, ast_udptl_get_far_max_ifp(), ast_udptl_set_local_max_ifp(), change_t38_state(), FALSE, ast_control_t38_parameters::fill_bit_removal, ast_control_t38_parameters::max_ifp, MIN, ast_control_t38_parameters::request_response, transmit_reinvite_with_sdp(), transmit_response_reliable(), transmit_response_with_t38_sdp(), and TRUE.

Referenced by sip_indicate().

{
   int res = 0;

   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) || !p->udptl) {
      return -1;
   }
   switch (parameters->request_response) {
   case AST_T38_NEGOTIATED:
   case AST_T38_REQUEST_NEGOTIATE:         /* Request T38 */
      /* Negotiation can not take place without a valid max_ifp value. */
      if (!parameters->max_ifp) {
         if (p->t38.state == T38_PEER_REINVITE) {
            AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
            transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
         }
         change_t38_state(p, T38_REJECTED);
         break;
      } else if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
         p->t38.our_parms = *parameters;
         /* modify our parameters to conform to the peer's parameters,
          * based on the rules in the ITU T.38 recommendation
          */
         if (!p->t38.their_parms.fill_bit_removal) {
            p->t38.our_parms.fill_bit_removal = FALSE;
         }
         if (!p->t38.their_parms.transcoding_mmr) {
            p->t38.our_parms.transcoding_mmr = FALSE;
         }
         if (!p->t38.their_parms.transcoding_jbig) {
            p->t38.our_parms.transcoding_jbig = FALSE;
         }
         p->t38.our_parms.version = MIN(p->t38.our_parms.version, p->t38.their_parms.version);
         p->t38.our_parms.rate_management = p->t38.their_parms.rate_management;
         ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
         change_t38_state(p, T38_ENABLED);
         transmit_response_with_t38_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL);
      } else if (p->t38.state != T38_ENABLED) {
         p->t38.our_parms = *parameters;
         ast_udptl_set_local_max_ifp(p->udptl, p->t38.our_parms.max_ifp);
         change_t38_state(p, T38_LOCAL_REINVITE);
         if (!p->pendinginvite) {
            transmit_reinvite_with_sdp(p, TRUE, FALSE);
         } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
            ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
         }
      }
      break;
   case AST_T38_TERMINATED:
   case AST_T38_REFUSED:
   case AST_T38_REQUEST_TERMINATE:         /* Shutdown T38 */
      if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
         change_t38_state(p, T38_REJECTED);
         transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
      } else if (p->t38.state == T38_ENABLED)
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      break;
   case AST_T38_REQUEST_PARMS: {    /* Application wants remote's parameters re-sent */
      struct ast_control_t38_parameters parameters = p->t38.their_parms;

      if (p->t38.state == T38_PEER_REINVITE) {
         AST_SCHED_DEL_UNREF(sched, p->t38id, dialog_unref(p, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
         parameters.max_ifp = ast_udptl_get_far_max_ifp(p->udptl);
         parameters.request_response = AST_T38_REQUEST_NEGOTIATE;
         if (p->owner) {
            ast_queue_control_data(p->owner, AST_CONTROL_T38_PARAMETERS, &parameters, sizeof(parameters));
         }
         /* we need to return a positive value here, so that applications that
          * send this request can determine conclusively whether it was accepted or not...
          * older versions of chan_sip would just silently accept it and return zero.
          */
         res = AST_T38_REQUEST_PARMS;
      }
      break;
   }
   default:
      res = -1;
      break;
   }

   return res;
}

static int is_method_allowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)		[static]

Check if method is allowed for a device or a dialog.

Definition at line 9462 of file chan_sip.c.

Referenced by sip_sendtext(), and update_connectedline().

{
   return ((*allowed_methods) >> method) & 1;
}

static void list_route

(

struct sip_route *

route

)

[static]

List all routes - mostly for debugging.

Definition at line 16224 of file chan_sip.c.

References ast_verbose().

Referenced by build_route().

{
   if (!route) {
      ast_verbose("list_route: no route\n");
   } else {
      for (;route; route = route->next)
         ast_verbose("list_route: hop: <%s>\n", route->hop);
   }
}

static int load_module

(

void

)

[static]

PBX load module - initialization.

Definition at line 34583 of file chan_sip.c.

References ao2_container_alloc, ao2_t_container_alloc, ao2_t_ref, ARRAY_LEN, ast_cc_agent_register(), ast_cc_monitor_register(), ast_channel_register(), ast_check_realtime(), ast_clear_flag, ast_cli_register_multiple(), ast_custom_function_register, ast_data_register_multiple, ast_format_cap_add_all_by_type(), ast_format_cap_alloc(), AST_FORMAT_TYPE_AUDIO, ast_log(), ast_manager_register_xml, AST_MODULE_LOAD_DECLINE, AST_MODULE_LOAD_FAILURE, AST_MODULE_LOAD_SUCCESS, ast_msg_tech_register(), ast_realtime_require_field(), ast_register_application_xml, ast_rtp_glue_register, ast_sched_context_create(), ast_sched_context_destroy(), ast_sip_api_provider_register(), ast_sip_api_provider_unregister(), ast_string_field_set, AST_TEST_REGISTER, ast_udptl_proto_register(), ast_verbose(), ast_websocket_add_protocol(), ASTOBJ_CONTAINER_INIT, BOGUS_PEER_MD5SECRET, ast_channel_tech::capabilities, CHANNEL_MODULE_LOAD, dialog_cmp_cb(), dialog_hash_cb(), EVENT_FLAG_REPORTING, EVENT_FLAG_SYSTEM, initialize_escs(), io_context_create(), io_context_destroy(), LOG_ERROR, manager_show_registry(), manager_sip_peer_status(), manager_sip_qualify_peer(), manager_sip_show_peer(), manager_sip_show_peers(), manager_sipnotify(), network_change_event_subscribe(), peer_cmp_cb(), peer_hash_cb(), peer_ipcmp_cb(), peer_iphash_cb(), regl, reload_config(), restart_monitor(), RQ_CHAR, RQ_INTEGER4, RQ_UINTEGER2, ast_channel_tech::send_digit_begin, SENTINEL, sip_addheader(), sip_cfg, sip_dtmfmode(), sip_epa_register(), sip_is_xml_parsable(), sip_keepalive_all_peers(), sip_monitor_instance_cmp_fn(), sip_monitor_instance_hash_fn(), sip_poke_all_peers(), sip_register_tests(), sip_removeheader(), sip_reqresp_parser_init(), sip_send_all_mwi_subscriptions(), sip_send_all_registers(), sip_tech_info, sip_websocket_callback(), submwil, temp_peer(), threadt_cmp_cb(), and threadt_hash_cb().

{
   ast_verbose("SIP channel loading...\n");

   if (!(sip_tech.capabilities = ast_format_cap_alloc())) {
      return AST_MODULE_LOAD_FAILURE;
   }

   if (ast_sip_api_provider_register(&chan_sip_api_provider)) {
      return AST_MODULE_LOAD_FAILURE;
   }

   /* the fact that ao2_containers can't resize automatically is a major worry! */
   /* if the number of objects gets above MAX_XXX_BUCKETS, things will slow down */
   peers = ao2_t_container_alloc(HASH_PEER_SIZE, peer_hash_cb, peer_cmp_cb, "allocate peers");
   peers_by_ip = ao2_t_container_alloc(HASH_PEER_SIZE, peer_iphash_cb, peer_ipcmp_cb, "allocate peers_by_ip");
   dialogs = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs");
   dialogs_needdestroy = ao2_t_container_alloc(1, NULL, NULL, "allocate dialogs_needdestroy");
   dialogs_rtpcheck = ao2_t_container_alloc(HASH_DIALOG_SIZE, dialog_hash_cb, dialog_cmp_cb, "allocate dialogs for rtpchecks");
   threadt = ao2_t_container_alloc(HASH_DIALOG_SIZE, threadt_hash_cb, threadt_cmp_cb, "allocate threadt table");
   if (!peers || !peers_by_ip || !dialogs || !dialogs_needdestroy || !dialogs_rtpcheck
      || !threadt) {
      ast_log(LOG_ERROR, "Unable to create primary SIP container(s)\n");
      return AST_MODULE_LOAD_FAILURE;
   }

   if (!(sip_cfg.caps = ast_format_cap_alloc())) {
      return AST_MODULE_LOAD_FAILURE;
   }
   ast_format_cap_add_all_by_type(sip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);

   ASTOBJ_CONTAINER_INIT(&regl); /* Registry object list -- not searched for anything */
   ASTOBJ_CONTAINER_INIT(&submwil); /* MWI subscription object list */

   if (!(sched = ast_sched_context_create())) {
      ast_log(LOG_ERROR, "Unable to create scheduler context\n");
      return AST_MODULE_LOAD_FAILURE;
   }

   if (!(io = io_context_create())) {
      ast_log(LOG_ERROR, "Unable to create I/O context\n");
      return AST_MODULE_LOAD_FAILURE;
   }

   sip_reloadreason = CHANNEL_MODULE_LOAD;

   can_parse_xml = sip_is_xml_parsable();
   if (reload_config(sip_reloadreason)) { /* Load the configuration from sip.conf */
      ast_sip_api_provider_unregister();
      return AST_MODULE_LOAD_DECLINE;
   }

   /* Initialize bogus peer. Can be done first after reload_config() */
   if (!(bogus_peer = temp_peer("(bogus_peer)"))) {
      ast_log(LOG_ERROR, "Unable to create bogus_peer for authentication\n");
      io_context_destroy(io);
      ast_sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }
   /* Make sure the auth will always fail. */
   ast_string_field_set(bogus_peer, md5secret, BOGUS_PEER_MD5SECRET);
   ast_clear_flag(&bogus_peer->flags[0], SIP_INSECURE);

   /* Prepare the version that does not require DTMF BEGIN frames.
    * We need to use tricks such as memcpy and casts because the variable
    * has const fields.
    */
   memcpy(&sip_tech_info, &sip_tech, sizeof(sip_tech));
   memset((void *) &sip_tech_info.send_digit_begin, 0, sizeof(sip_tech_info.send_digit_begin));

   if (ast_msg_tech_register(&sip_msg_tech)) {
      /* LOAD_FAILURE stops Asterisk, so cleanup is a moot point. */
      return AST_MODULE_LOAD_FAILURE;
   }

   /* Make sure we can register our sip channel type */
   if (ast_channel_register(&sip_tech)) {
      ast_log(LOG_ERROR, "Unable to register channel type 'SIP'\n");
      ao2_t_ref(bogus_peer, -1, "unref the bogus_peer");
      io_context_destroy(io);
      ast_sched_context_destroy(sched);
      return AST_MODULE_LOAD_FAILURE;
   }

#ifdef TEST_FRAMEWORK
   AST_TEST_REGISTER(test_sip_peers_get);
   AST_TEST_REGISTER(test_sip_mwi_subscribe_parse);
   AST_TEST_REGISTER(test_tcp_message_fragmentation);
   AST_TEST_REGISTER(get_in_brackets_const_test);
#endif

   /* Register AstData providers */
   ast_data_register_multiple(sip_data_providers, ARRAY_LEN(sip_data_providers));

   /* Register all CLI functions for SIP */
   ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));

   /* Tell the UDPTL subdriver that we're here */
   ast_udptl_proto_register(&sip_udptl);

   /* Tell the RTP engine about our RTP glue */
   ast_rtp_glue_register(&sip_rtp_glue);

   /* Register dialplan applications */
   ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
   ast_register_application_xml(app_sipaddheader, sip_addheader);
   ast_register_application_xml(app_sipremoveheader, sip_removeheader);
#ifdef TEST_FRAMEWORK
   ast_register_application_xml(app_sipsendcustominfo, sip_sendcustominfo);
#endif

   /* Register dialplan functions */
   ast_custom_function_register(&sip_header_function);
   ast_custom_function_register(&sippeer_function);
   ast_custom_function_register(&sipchaninfo_function);
   ast_custom_function_register(&checksipdomain_function);

   /* Register manager commands */
   ast_manager_register_xml("SIPpeers", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peers);
   ast_manager_register_xml("SIPshowpeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_show_peer);
   ast_manager_register_xml("SIPqualifypeer", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_sip_qualify_peer);
   ast_manager_register_xml("SIPshowregistry", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, manager_show_registry);
   ast_manager_register_xml("SIPnotify", EVENT_FLAG_SYSTEM, manager_sipnotify);
   ast_manager_register_xml("SIPpeerstatus", EVENT_FLAG_SYSTEM, manager_sip_peer_status);
   sip_poke_all_peers();
   sip_keepalive_all_peers();
   sip_send_all_registers();
   sip_send_all_mwi_subscriptions();
   initialize_escs();

   if (sip_epa_register(&cc_epa_static_data)) {
      ast_sip_api_provider_unregister();
      return AST_MODULE_LOAD_DECLINE;
   }

   if (sip_reqresp_parser_init() == -1) {
      ast_log(LOG_ERROR, "Unable to initialize the SIP request and response parser\n");
      ast_sip_api_provider_unregister();
      return AST_MODULE_LOAD_DECLINE;
   }

   if (can_parse_xml) {
      /* SIP CC agents require the ability to parse XML PIDF bodies
       * in incoming PUBLISH requests
       */
      if (ast_cc_agent_register(&sip_cc_agent_callbacks)) {
         ast_sip_api_provider_unregister();
         return AST_MODULE_LOAD_DECLINE;
      }
   }
   if (ast_cc_monitor_register(&sip_cc_monitor_callbacks)) {
      ast_sip_api_provider_unregister();
      return AST_MODULE_LOAD_DECLINE;
   }
   if (!(sip_monitor_instances = ao2_container_alloc(37, sip_monitor_instance_hash_fn, sip_monitor_instance_cmp_fn))) {
      ast_sip_api_provider_unregister();
      return AST_MODULE_LOAD_DECLINE;
   }

   /* And start the monitor for the first time */
   restart_monitor();

   ast_realtime_require_field(ast_check_realtime("sipregs") ? "sipregs" : "sippeers",
      "name", RQ_CHAR, 10,
      "ipaddr", RQ_CHAR, INET6_ADDRSTRLEN - 1,
      "port", RQ_UINTEGER2, 5,
      "regseconds", RQ_INTEGER4, 11,
      "defaultuser", RQ_CHAR, 10,
      "fullcontact", RQ_CHAR, 35,
      "regserver", RQ_CHAR, 20,
      "useragent", RQ_CHAR, 20,
      "lastms", RQ_INTEGER4, 11,
      SENTINEL);


   sip_register_tests();
   network_change_event_subscribe();

   ast_websocket_add_protocol("sip", sip_websocket_callback);

   return AST_MODULE_LOAD_SUCCESS;
}

static int local_attended_transfer	(	struct sip_pvt *	transferer,
		struct sip_dual *	current,
		struct sip_request *	req,
		uint32_t	seqno,
		int *	nounlock
	)		[static]

Find all call legs and bridge transferee with target called from handle_request_refer.

Note:

this function assumes two locks to begin with, sip_pvt transferer and current.chan1 (the pvt's owner)... 2 additional locks are held at the beginning of the function, targetcall_pvt, and targetcall_pvt's owner channel (which is stored in target.chan1). These 2 locks _MUST_ be let go by the end of the function. Do not be confused into thinking a pvt's owner is the same thing as the channels locked at the beginning of this function, after the masquerade this may not be true. Be consistent and unlock only the exact same pointers that were locked to begin with.

If this function is successful, only the transferer pvt lock will remain on return. Setting nounlock indicates to handle_request_do() that the pvt's owner it locked does not require an unlock.

Definition at line 26043 of file chan_sip.c.

References ao2_t_ref, append_history, ast_alloca, ast_bridged_channel(), AST_CEL_ATTENDEDTRANSFER, ast_cel_report_event(), ast_channel_connected(), ast_channel_language(), ast_channel_linkedid(), ast_channel_name(), ast_channel_queue_connected_line_update(), ast_channel_ref, ast_channel_uniqueid(), ast_channel_unlock, ast_channel_unref, ast_clear_flag, ast_connected_line_build_data(), AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER, AST_CONTROL_READ_ACTION, AST_CONTROL_RINGING, AST_CONTROL_UNHOLD, ast_copy_string(), ast_debug, ast_do_masquerade(), AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO, ast_indicate(), ast_manager_event_multichan, ast_party_connected_line_copy(), ast_party_connected_line_free(), ast_party_connected_line_init(), ast_party_id_reset(), ast_queue_control_data(), ast_set_flag, ast_state2str(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_streamfile(), ast_strlen_zero(), ast_waitstream(), attempt_transfer(), EVENT_FLAG_CALL, frame_size, get_sip_pvt_byid_locked(), pbx_builtin_getvar_helper(), ast_party_connected_line::priv, sip_pvt_lock, sip_pvt_unlock, ast_party_connected_line::source, transmit_notify_with_sipfrag(), TRUE, and xfersound.

Referenced by handle_request_refer().

{
   struct sip_dual target;    /* Chan 1: Call from tranferer to Asterisk */
               /* Chan 2: Call from Asterisk to target */
   int res = 0;
   struct sip_pvt *targetcall_pvt;
   struct ast_party_connected_line connected_to_transferee;
   struct ast_party_connected_line connected_to_target;
   char transferer_linkedid[32];
   struct ast_channel *chans[2];

   /* Check if the call ID of the replaces header does exist locally */
   if (!(targetcall_pvt = get_sip_pvt_byid_locked(transferer->refer->replaces_callid, transferer->refer->replaces_callid_totag,
      transferer->refer->replaces_callid_fromtag))) {
      if (transferer->refer->localtransfer) {
         /* We did not find the refered call. Sorry, can't accept then */
         /* Let's fake a response from someone else in order
            to follow the standard */
         transmit_notify_with_sipfrag(transferer, seqno, "481 Call leg/transaction does not exist", TRUE);
         append_history(transferer, "Xfer", "Refer failed");
         ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
         transferer->refer->status = REFER_FAILED;
         return -1;
      }
      /* Fall through for remote transfers that we did not find locally */
      ast_debug(3, "SIP attended transfer: Not our call - generating INVITE with replaces\n");
      return 0;
   }

   /* Ok, we can accept this transfer */
   append_history(transferer, "Xfer", "Refer accepted");
   if (!targetcall_pvt->owner) { /* No active channel */
      ast_debug(4, "SIP attended transfer: Error: No owner of target call\n");
      /* Cancel transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "503 Service Unavailable", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      ast_clear_flag(&transferer->flags[0], SIP_GOTREFER);
      transferer->refer->status = REFER_FAILED;
      sip_pvt_unlock(targetcall_pvt);
      if (targetcall_pvt)
         ao2_t_ref(targetcall_pvt, -1, "Drop targetcall_pvt pointer");
      return -1;
   }

   /* We have a channel, find the bridge */
   target.chan1 = ast_channel_ref(targetcall_pvt->owner);            /* Transferer to Asterisk */
   target.chan2 = ast_bridged_channel(targetcall_pvt->owner);  /* Asterisk to target */
   if (target.chan2) {
      ast_channel_ref(target.chan2);
   }

   if (!target.chan2 || !(ast_channel_state(target.chan2) == AST_STATE_UP || ast_channel_state(target.chan2) == AST_STATE_RINGING) ) {
      /* Wrong state of new channel */
      if (target.chan2)
         ast_debug(4, "SIP attended transfer: Error: Wrong state of target call: %s\n", ast_state2str(ast_channel_state(target.chan2)));
      else if (ast_channel_state(target.chan1) != AST_STATE_RING)
         ast_debug(4, "SIP attended transfer: Error: No target channel\n");
      else
         ast_debug(4, "SIP attended transfer: Attempting transfer in ringing state\n");
   }

   /* Transfer */
   if (sipdebug) {
      if (current->chan2)  /* We have two bridges */
         ast_debug(4, "SIP attended transfer: trying to bridge %s and %s\n", ast_channel_name(target.chan1), ast_channel_name(current->chan2));
      else        /* One bridge, propably transfer of IVR/voicemail etc */
         ast_debug(4, "SIP attended transfer: trying to make %s take over (masq) %s\n", ast_channel_name(target.chan1), ast_channel_name(current->chan1));
   }

   ast_set_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);   /* Delay hangup */

   ast_copy_string(transferer_linkedid, ast_channel_linkedid(transferer->owner), sizeof(transferer_linkedid));

   /* Perform the transfer */
   chans[0] = transferer->owner;
   chans[1] = target.chan1;
   ast_manager_event_multichan(EVENT_FLAG_CALL, "Transfer", 2, chans,
      "TransferMethod: SIP\r\n"
      "TransferType: Attended\r\n"
      "Channel: %s\r\n"
      "Uniqueid: %s\r\n"
      "SIP-Callid: %s\r\n"
      "TargetChannel: %s\r\n"
      "TargetUniqueid: %s\r\n",
      ast_channel_name(transferer->owner),
      ast_channel_uniqueid(transferer->owner),
      transferer->callid,
      ast_channel_name(target.chan1),
      ast_channel_uniqueid(target.chan1));
   ast_party_connected_line_init(&connected_to_transferee);
   ast_party_connected_line_init(&connected_to_target);
   /* No need to lock current->chan1 here since it was locked in sipsock_read */
   ast_party_connected_line_copy(&connected_to_transferee, ast_channel_connected(current->chan1));
   /* No need to lock target.chan1 here since it was locked in get_sip_pvt_byid_locked */
   ast_party_connected_line_copy(&connected_to_target, ast_channel_connected(target.chan1));
   /* Reset any earlier private connected id representation */
   ast_party_id_reset(&connected_to_transferee.priv);
   ast_party_id_reset(&connected_to_target.priv);
   connected_to_target.source = connected_to_transferee.source = AST_CONNECTED_LINE_UPDATE_SOURCE_TRANSFER;
   res = attempt_transfer(current, &target);
   if (res) {
      /* Failed transfer */
      transmit_notify_with_sipfrag(transferer, seqno, "486 Busy Here", TRUE);
      append_history(transferer, "Xfer", "Refer failed");
      ast_clear_flag(&transferer->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
      /* if transfer failed, go ahead and unlock targetcall_pvt and it's owner channel */
      sip_pvt_unlock(targetcall_pvt);
      ast_channel_unlock(target.chan1);
   } else {
      /* Transfer succeeded! */
      const char *xfersound = pbx_builtin_getvar_helper(target.chan1, "ATTENDED_TRANSFER_COMPLETE_SOUND");

      /* target.chan1 was locked in get_sip_pvt_byid_locked, do not unlock target.chan1 before this */
      ast_cel_report_event(target.chan1, AST_CEL_ATTENDEDTRANSFER, NULL, transferer_linkedid, target.chan2);

      /* Tell transferer that we're done. */
      transmit_notify_with_sipfrag(transferer, seqno, "200 OK", TRUE);
      append_history(transferer, "Xfer", "Refer succeeded");
      transferer->refer->status = REFER_200OK;
      if (target.chan2 && !ast_strlen_zero(xfersound) && ast_streamfile(target.chan2, xfersound, ast_channel_language(target.chan2)) >= 0) {
         ast_waitstream(target.chan2, "");
      }

      /* By forcing the masquerade, we know that target.chan1 and target.chan2 are bridged. We then
       * can queue connected line updates where they need to go.
       *
       * before a masquerade, all channel and pvt locks must be unlocked.  Any recursive
       * channel locks held before this function invalidates channel container locking order.
       * Since we are unlocking both the pvt (transferer) and its owner channel (current.chan1)
       * it is possible for current.chan1 to be destroyed in the pbx thread.  To prevent this
       * we must give c a reference before any unlocking takes place.
       */

      ast_channel_ref(current->chan1);
      ast_channel_unlock(current->chan1); /* current.chan1 is p->owner before the masq, it was locked by socket_read()*/
      ast_channel_unlock(target.chan1);
      *nounlock = 1;  /* we just unlocked the dialog's channel and have no plans of locking it again. */
      sip_pvt_unlock(targetcall_pvt);
      sip_pvt_unlock(transferer);

      ast_do_masquerade(target.chan1);

      if (target.chan2) {
         ast_indicate(target.chan2, AST_CONTROL_UNHOLD);
      }

      if (current->chan2 && ast_channel_state(current->chan2) == AST_STATE_RING) {
         ast_indicate(target.chan1, AST_CONTROL_RINGING);
      }

      if (target.chan2) {
         ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);
         ast_channel_queue_connected_line_update(target.chan2, &connected_to_target, NULL);
      } else {
         /* Since target.chan1 isn't actually connected to another channel, there is no way for us
          * to queue a frame so that its connected line status will be updated.
          *
          * Instead, we use the somewhat hackish approach of using a special control frame type that
          * instructs ast_read to perform a specific action. In this case, the frame we queue tells
          * ast_read to call the connected line interception macro configured for target.chan1.
          */
         struct ast_control_read_action_payload *frame_payload;
         int payload_size;
         int frame_size;
         unsigned char connected_line_data[1024];
         payload_size = ast_connected_line_build_data(connected_line_data,
            sizeof(connected_line_data), &connected_to_target, NULL);
         frame_size = payload_size + sizeof(*frame_payload);
         if (payload_size != -1) {
            frame_payload = ast_alloca(frame_size);
            frame_payload->payload_size = payload_size;
            memcpy(frame_payload->payload, connected_line_data, payload_size);
            frame_payload->action = AST_FRAME_READ_ACTION_CONNECTED_LINE_MACRO;
            ast_queue_control_data(target.chan1, AST_CONTROL_READ_ACTION, frame_payload, frame_size);
         }
         /* In addition to queueing the read action frame so that target.chan1's connected line info
          * will be updated, we also are going to queue a plain old connected line update on target.chan1. This
          * way, either Dial or Queue can apply this connected line update to the outgoing ringing channel.
          */
         ast_channel_queue_connected_line_update(target.chan1, &connected_to_transferee, NULL);

      }
      sip_pvt_lock(transferer); /* the transferer pvt is expected to remain locked on return */

      ast_channel_unref(current->chan1);
   }

   /* at this point if the transfer is successful only the transferer pvt should be locked. */
   ast_party_connected_line_free(&connected_to_target);
   ast_party_connected_line_free(&connected_to_transferee);
   ast_channel_unref(target.chan1);
   if (target.chan2) {
      ast_channel_unref(target.chan2);
   }
   if (targetcall_pvt)
      ao2_t_ref(targetcall_pvt, -1, "drop targetcall_pvt");
   return 1;
}

static void lws2sws

(

struct ast_str *

msgbuf

)

[static]

Parse multiline SIP headers into one header This is enabled if pedanticsipchecking is enabled.

Definition at line 9550 of file chan_sip.c.

References ast_str_strlen(), and len().

Referenced by handle_request_do(), and read_raw_content_length().

{
   char *msgbuf = data->str;
   int len = ast_str_strlen(data);
   int h = 0, t = 0;
   int lws = 0;

   for (; h < len;) {
      /* Eliminate all CRs */
      if (msgbuf[h] == '\r') {
         h++;
         continue;
      }
      /* Check for end-of-line */
      if (msgbuf[h] == '\n') {
         /* Check for end-of-message */
         if (h + 1 == len)
            break;
         /* Check for a continuation line */
         if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') {
            /* Merge continuation line */
            h++;
            continue;
         }
         /* Propagate LF and start new line */
         msgbuf[t++] = msgbuf[h++];
         lws = 0;
         continue;
      }
      if (msgbuf[h] == ' ' || msgbuf[h] == '\t') {
         if (lws) {
            h++;
            continue;
         }
         msgbuf[t++] = msgbuf[h++];
         lws = 1;
         continue;
      }
      msgbuf[t++] = msgbuf[h++];
      if (lws)
         lws = 0;
   }
   msgbuf[t] = '\0';
   data->used = t;
}

static void make_our_tag

(

struct sip_pvt *

pvt

)

[static]

Make our SIP dialog tag.

Definition at line 8593 of file chan_sip.c.

References ast_random(), and ast_string_field_build.

Referenced by handle_request_invite(), handle_request_subscribe(), sip_alloc(), and transmit_response_using_temp().

{
   ast_string_field_build(pvt, tag, "as%08lx", ast_random());
}

static int manager_show_registry	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Show SIP registrations in the manager API.

Definition at line 19013 of file chan_sip.c.

References ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, regl, regstate2str(), S_OR, and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Registrations will follow", "start");

   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_RDLOCK(iterator);
      astman_append(s,
         "Event: RegistryEntry\r\n"
         "%s"
         "Host: %s\r\n"
         "Port: %d\r\n"
         "Username: %s\r\n"
         "Domain: %s\r\n"
         "DomainPort: %d\r\n"
         "Refresh: %d\r\n"
         "State: %s\r\n"
         "RegistrationTime: %ld\r\n"
         "\r\n",
         idtext,
         iterator->hostname,
         iterator->portno ? iterator->portno : STANDARD_SIP_PORT,
         iterator->username,
         S_OR(iterator->regdomain,iterator->hostname),
         iterator->regdomainport ? iterator->regdomainport : STANDARD_SIP_PORT,
         iterator->refresh,
         regstate2str(iterator->regstate),
         (long) iterator->regtime.tv_sec);
      ASTOBJ_UNLOCK(iterator);
      total++;
   } while(0));

   astman_append(s,
      "Event: RegistrationsComplete\r\n"
      "EventList: Complete\r\n"
      "ListItems: %d\r\n"
      "%s"
      "\r\n", total, idtext);
   
   return 0;
}

static int manager_sip_peer_status	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Show SIP peers in the manager API.

Definition at line 19850 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_ack(), astman_send_error(), FALSE, send_manager_peer_status(), sip_find_peer(), sip_unref_peer(), and TRUE.

Referenced by load_module().

{
   const char *id = astman_get_header(m,"ActionID");
   const char *peer_name = astman_get_header(m,"Peer");
   char idText[256] = "";
   struct sip_peer *peer = NULL;

   if (!ast_strlen_zero(id)) {
      snprintf(idText, sizeof(idText), "ActionID: %s\r\n", id);
   }

   if (!ast_strlen_zero(peer_name)) {
      /* strip SIP/ from the begining of the peer name */
      if (strlen(peer_name) >= 4 && !strncasecmp("SIP/", peer_name, 4)) {
         peer_name += 4;
      }

      peer = sip_find_peer(peer_name, NULL, TRUE, FINDPEERS, FALSE, 0);
      if (!peer) {
         astman_send_error(s, m, "No such peer");
         return 0;
      }
   }

   astman_send_ack(s, m, "Peer status will follow");

   if (!peer) {
      struct ao2_iterator i = ao2_iterator_init(peers, 0);
      while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table for SIPpeerstatus"))) {
         ao2_lock(peer);
         send_manager_peer_status(s, peer, idText);
         ao2_unlock(peer);
         sip_unref_peer(peer, "unref peer for SIPpeerstatus");
      }
      ao2_iterator_destroy(&i);
   } else {
      ao2_lock(peer);
      send_manager_peer_status(s, peer, idText);
      ao2_unlock(peer);
      sip_unref_peer(peer, "unref peer for SIPpeerstatus");
   }


   astman_append(s,
   "Event: SIPpeerstatusComplete\r\n"
   "%s"
   "\r\n",
   idText);

   return 0;
}

static int manager_sip_qualify_peer	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Qualify SIP peers in the manager API.

Definition at line 19925 of file chan_sip.c.

References _sip_qualify_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), and astman_send_error().

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "qualify";
   a[2] = "peer";
   a[3] = peer;

   _sip_qualify_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n\r\n" );
   return 0;
}

static int manager_sip_show_peer	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Show SIP peers in the manager API.

Definition at line 19780 of file chan_sip.c.

References _sip_show_peer(), ast_strlen_zero(), astman_append(), astman_get_header(), and astman_send_error().

Referenced by load_module().

{
   const char *a[4];
   const char *peer;

   peer = astman_get_header(m, "Peer");
   if (ast_strlen_zero(peer)) {
      astman_send_error(s, m, "Peer: <name> missing.");
      return 0;
   }
   a[0] = "sip";
   a[1] = "show";
   a[2] = "peer";
   a[3] = peer;

   _sip_show_peer(1, -1, s, m, 4, a);
   astman_append(s, "\r\n" );
   return 0;
}

static int manager_sip_show_peers	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Show SIP peers in the manager API.

Definition at line 19063 of file chan_sip.c.

References _sip_show_peers(), ast_strlen_zero(), astman_append(), astman_get_header(), astman_send_listack(), and total.

Referenced by load_module().

{
   const char *id = astman_get_header(m, "ActionID");
   const char *a[] = {"sip", "show", "peers"};
   char idtext[256] = "";
   int total = 0;

   if (!ast_strlen_zero(id))
      snprintf(idtext, sizeof(idtext), "ActionID: %s\r\n", id);

   astman_send_listack(s, m, "Peer status list will follow", "start");
   /* List the peers in separate manager events */
   _sip_show_peers(-1, &total, s, m, 3, a);
   /* Send final confirmation */
   astman_append(s,
   "Event: PeerlistComplete\r\n"
   "EventList: Complete\r\n"
   "ListItems: %d\r\n"
   "%s"
   "\r\n", total, idtext);
   return 0;
}

static int manager_sipnotify	(	struct mansession *	s,
		const struct message *	m
	)		[static]

Definition at line 14926 of file chan_sip.c.

References ast_log(), ast_set_flag, ast_str_append(), ast_str_strlen(), ast_strlen_zero(), ast_variable_new(), ast_variables_destroy(), astman_get_header(), astman_get_variables(), astman_send_ack(), astman_send_error(), create_addr(), dialog_unlink_all(), LOG_WARNING, ast_variable::name, ast_variable::next, sip_alloc(), sip_notify_alloc(), sip_scheddestroy(), transmit_invite(), ast_variable::value, and var.

Referenced by load_module().

{
   const char *channame = astman_get_header(m, "Channel");
   struct ast_variable *vars = astman_get_variables(m);
   struct sip_pvt *p;
   struct ast_variable *header, *var;

   if (ast_strlen_zero(channame)) {
      astman_send_error(s, m, "SIPNotify requires a channel name");
      return 0;
   }

   if (!strncasecmp(channame, "sip/", 4)) {
      channame += 4;
   }

   if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, NULL))) {
      astman_send_error(s, m, "Unable to build sip pvt data for notify (memory/socket error)");
      return 0;
   }

   if (create_addr(p, channame, NULL, 0)) {
      /* Maybe they're not registered, etc. */
      dialog_unlink_all(p);
      dialog_unref(p, "unref dialog inside for loop" );
      /* sip_destroy(p); */
      astman_send_error(s, m, "Could not create address");
      return 0;
   }

   /* Notify is outgoing call */
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
   sip_notify_alloc(p);

   p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");

   for (var = vars; var; var = var->next) {
      if (!strcasecmp(var->name, "Content")) {
         if (ast_str_strlen(p->notify->content))
            ast_str_append(&p->notify->content, 0, "\r\n");
         ast_str_append(&p->notify->content, 0, "%s", var->value);
      } else if (!strcasecmp(var->name, "Content-Length")) {
         ast_log(LOG_WARNING, "it is not necessary to specify Content-Length, ignoring\n");
      } else {
         header->next = ast_variable_new(var->name, var->value, "");
         header = header->next;
      }
   }

   sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
   transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
   dialog_unref(p, "bump down the count of p since we're done with it.");

   astman_send_ack(s, m, "Notify Sent");
   ast_variables_destroy(vars);
   return 0;
}

static int map_s_x	(	const struct _map_x_s *	table,
		const char *	s,
		int	errorvalue
	)		[static]

map from a string to an integer value, case insensitive. If no match is found, return errorvalue.

Definition at line 2369 of file chan_sip.c.

Referenced by str2dtmfmode(), str2stmode(), and str2strefresherparam().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++) {
      if (!strcasecmp(cur->s, s)) {
         return cur->x;
      }
   }
   return errorvalue;
}

static const char* map_x_s	(	const struct _map_x_s *	table,
		int	x,
		const char *	errorstring
	)		[static]

map from an integer value to a string. If no match is found, return errorstring

Definition at line 2354 of file chan_sip.c.

Referenced by allowoverlap2str(), autocreatepeer2str(), dtmfmode2str(), faxec2str(), insecure2str(), referstatus2str(), regstate2str(), stmode2str(), strefresher2str(), strefresherparam2str(), and trust_id_outbound2str().

{
   const struct _map_x_s *cur;

   for (cur = table; cur->s; cur++) {
      if (cur->x == x) {
         return cur->s;
      }
   }
   return errorstring;
}

static void mark_method_allowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)		[static]

Definition at line 9451 of file chan_sip.c.

Referenced by handle_response(), handle_response_info(), handle_response_message(), mark_parsed_methods(), and set_pvt_allowed_methods().

{
   (*allowed_methods) |= (1 << method);
}

static void mark_method_unallowed	(	unsigned int *	allowed_methods,
		enum sipmethod	method
	)		[static]

Definition at line 9456 of file chan_sip.c.

Referenced by handle_response(), handle_response_info(), handle_response_message(), and handle_response_publish().

{
   (*allowed_methods) &= ~(1 << method);
}

static void mark_parsed_methods	(	unsigned int *	methods,
		char *	methods_str
	)		[static]

Definition at line 9467 of file chan_sip.c.

References ast_skip_blanks(), ast_strlen_zero(), find_sip_method(), and mark_method_allowed().

Referenced by build_peer(), parse_allowed_methods(), and reload_config().

{
   char *method;
   for (method = strsep(&methods_str, ","); !ast_strlen_zero(method); method = strsep(&methods_str, ",")) {
      int id = find_sip_method(ast_skip_blanks(method));
      if (id == SIP_UNKNOWN) {
         continue;
      }
      mark_method_allowed(methods, id);
   }
}

static int match_and_cleanup_peer_sched	(	void *	peerobj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 3314 of file chan_sip.c.

References ast_dnsmgr_release(), CMP_MATCH, peer_sched_cleanup(), SIP_PEERS_ALL, and sip_unref_peer().

Referenced by unlink_peers_from_tables().

{
   struct sip_peer *peer = peerobj;
   peer_unlink_flag_t which = *(peer_unlink_flag_t *)arg;

   if (which == SIP_PEERS_ALL || peer->the_mark) {
      peer_sched_cleanup(peer);
      if (peer->dnsmgr) {
         ast_dnsmgr_release(peer->dnsmgr);
         peer->dnsmgr = NULL;
         sip_unref_peer(peer, "Release peer from dnsmgr");
      }
      return CMP_MATCH;
   }
   return 0;
}

static enum match_req_res match_req_to_dialog	(	struct sip_pvt *	sip_pvt_ptr,
		struct match_req_args *	arg
	)		[static]

Definition at line 8862 of file chan_sip.c.

References ast_strlen_zero(), ast_test_flag, match_req_args::authentication_present, match_req_args::callid, match_req_args::fromtag, match_req_args::method, match_req_args::respid, match_req_args::ruri, match_req_args::seqno, SIP_REQ_FORKED, SIP_REQ_LOOP_DETECTED, SIP_REQ_MATCH, SIP_REQ_NOT_MATCH, sip_uri_cmp(), match_req_args::totag, match_req_args::viabranch, and match_req_args::viasentby.

Referenced by find_call().

{
   const char *init_ruri = NULL;
   if (sip_pvt_ptr->initreq.headers) {
      init_ruri = REQ_OFFSET_TO_STR(&sip_pvt_ptr->initreq, rlpart2);
   }

   /*
    * Match Tags and call-id to Dialog
    */
   if (!ast_strlen_zero(arg->callid) && strcmp(sip_pvt_ptr->callid, arg->callid)) {
      /* call-id does not match. */
      return SIP_REQ_NOT_MATCH;
   }
   if (arg->method == SIP_RESPONSE) {
      /* Verify fromtag of response matches the tag we gave them. */
      if (strcmp(arg->fromtag, sip_pvt_ptr->tag)) {
         /* fromtag from response does not match our tag */
         return SIP_REQ_NOT_MATCH;
      }

      /* Verify totag if we have one stored for this dialog, but never be strict about this for
       * a response until the dialog is established */
      if (!ast_strlen_zero(sip_pvt_ptr->theirtag) && ast_test_flag(&sip_pvt_ptr->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
         if (ast_strlen_zero(arg->totag)) {
            /* missing totag when they already gave us one earlier */
            return SIP_REQ_NOT_MATCH;
         }
         /* compare the totag of response with the tag we have stored for them */
         if (strcmp(arg->totag, sip_pvt_ptr->theirtag)) {
            /* totag did not match what we had stored for them. */
            char invite_branch[32] = { 0, };
            if (sip_pvt_ptr->invite_branch) {
               snprintf(invite_branch, sizeof(invite_branch), "z9hG4bK%08x", (int) sip_pvt_ptr->invite_branch);
            }
            /* Forked Request Detection
             *
             * If this is a 200ok response and the totags do not match, this
             * might be a forked response to an outgoing Request. Detection of
             * a forked response must meet the criteria below.
             *
             * 1. must be a 2xx Response
             * 2. call-d equal to call-id of Request. this is done earlier
             * 3. from-tag equal to from-tag of Request. this is done earlier
             * 4. branch parameter equal to branch of inital Request
             * 5. to-tag _NOT_ equal to previous 2xx response that already established the dialog.
             */
            if ((arg->respid == 200) &&
               !ast_strlen_zero(invite_branch) &&
               !ast_strlen_zero(arg->viabranch) &&
               !strcmp(invite_branch, arg->viabranch)) {
               return SIP_REQ_FORKED;
            }

            /* The totag did not match the one we had stored, and this is not a Forked Request. */
            return SIP_REQ_NOT_MATCH;
         }
      }
   } else {
      /* Verify the fromtag of Request matches the tag they provided earlier.
       * If this is a Request with authentication credentials, forget their old
       * tag as it is not valid after the 401 or 407 response. */
      if (!arg->authentication_present && strcmp(arg->fromtag, sip_pvt_ptr->theirtag)) {
         /* their tag does not match the one was have stored for them */
         return SIP_REQ_NOT_MATCH;
      }
      /* Verify if totag is present in Request, that it matches what we gave them as our tag earlier */
      if (!ast_strlen_zero(arg->totag) && (strcmp(arg->totag, sip_pvt_ptr->tag))) {
         /* totag from Request does not match our tag */
         return SIP_REQ_NOT_MATCH;
      }
   }

   /*
    * Compare incoming request against initial transaction.
    * 
    * This is a best effort attempt at distinguishing forked requests from
    * our initial transaction.  If all the elements are NOT in place to evaluate
    * this, this block is ignored and the dialog match is made regardless.
    * Once the totag is established after the dialog is confirmed, this is not necessary.
    *
    * CRITERIA required for initial transaction matching.
    * 
    * 1. Is a Request
    * 2. Callid and theirtag match (this is done in the dialog matching block)
    * 3. totag is NOT present
    * 4. CSeq matchs our initial transaction's cseq number
    * 5. pvt has init via branch parameter stored
    */
   if ((arg->method != SIP_RESPONSE) &&                 /* must be a Request */
      ast_strlen_zero(arg->totag) &&                   /* must not have a totag */
      (sip_pvt_ptr->init_icseq == arg->seqno) &&       /* the cseq must be the same as this dialogs initial cseq */
      !ast_strlen_zero(sip_pvt_ptr->initviabranch) &&  /* The dialog must have started with a RFC3261 compliant branch tag */
      init_ruri) {                                     /* the dialog must have an initial request uri associated with it */
      /* This Request matches all the criteria required for Loop/Merge detection.
       * Now we must go down the path of comparing VIA's and RURIs. */
      if (ast_strlen_zero(arg->viabranch) ||
         strcmp(arg->viabranch, sip_pvt_ptr->initviabranch) ||
         ast_strlen_zero(arg->viasentby) ||
         strcmp(arg->viasentby, sip_pvt_ptr->initviasentby)) {
         /* At this point, this request does not match this Dialog.*/

         /* if methods are different this is just a mismatch */
         if ((sip_pvt_ptr->method != arg->method)) {
            return SIP_REQ_NOT_MATCH;
         }

         /* If RUIs are different, this is a forked request to a separate URI.
          * Returning a mismatch allows this Request to be processed separately. */
         if (sip_uri_cmp(init_ruri, arg->ruri)) {
            /* not a match, request uris are different */
            return SIP_REQ_NOT_MATCH;
         }

         /* Loop/Merge Detected
          *
          * ---Current Matches to Initial Request---
          * request uri
          * Call-id
          * their-tag
          * no totag present
          * method
          * cseq
          *
          * --- Does not Match Initial Request ---
          * Top Via
          *
          * Without the same Via, this can not match our initial transaction for this dialog,
          * but given that this Request matches everything else associated with that initial
          * Request this is most certainly a Forked request in which we have already received
          * part of the fork.
          */
         return SIP_REQ_LOOP_DETECTED;
      }
   } /* end of Request Via check */

   /* Match Authentication Request.
    *
    * A Request with an Authentication header must come back with the
    * same Request URI.  Otherwise it is not a match.
    */
   if ((arg->method != SIP_RESPONSE) &&      /* Must be a Request type to even begin checking this */
      ast_strlen_zero(arg->totag) &&        /* no totag is present to match */
      arg->authentication_present &&        /* Authentication header is present in Request */
      sip_uri_cmp(init_ruri, arg->ruri)) {  /* Compare the Request URI of both the last Request and this new one */

      /* Authentication was provided, but the Request URI did not match the last one on this dialog. */
      return SIP_REQ_NOT_MATCH;
   }

   return SIP_REQ_MATCH;
}

static int method_match	(	enum sipmethod	id,
		const char *	name
	)		[static]

returns true if 'name' (with optional trailing whitespace) matches the sip method 'id'. Strictly speaking, SIP methods are case SENSITIVE, but we do a case-insensitive comparison to be more tolerant. following Jon Postel's rule: Be gentle in what you accept, strict with what you send

Definition at line 3654 of file chan_sip.c.

References len(), sip_methods, and text.

Referenced by __sip_autodestruct(), __sip_semi_ack(), and find_sip_method().

{
   int len = strlen(sip_methods[id].text);
   int l_name = name ? strlen(name) : 0;
   /* true if the string is long enough, and ends with whitespace, and matches */
   return (l_name >= len && name && name[len] < 33 &&
      !strncasecmp(sip_methods[id].text, name, len));
}

static void mwi_event_cb	(	const struct ast_event *	event,
		void *	userdata
	)		[static]

Receive MWI events that we have subscribed to.

Definition at line 16588 of file chan_sip.c.

References sip_send_mwi_to_peer().

Referenced by add_peer_mwi_subs().

{
   struct sip_peer *peer = userdata;

   sip_send_mwi_to_peer(peer, 0);
}

static void network_change_event_cb	(	const struct ast_event *	event,
		void *	userdata
	)		[static]

Definition at line 16634 of file chan_sip.c.

References ast_debug, ast_sched_add(), and network_change_event_sched_cb().

Referenced by network_change_event_subscribe().

{
   ast_debug(1, "SIP, got a network change event, renewing all SIP registrations.\n");
   if (network_change_event_sched_id == -1) {
      network_change_event_sched_id = ast_sched_add(sched, 1000, network_change_event_sched_cb, NULL);
   }
}

static int network_change_event_sched_cb

(

const void *

data

)

[static]

Definition at line 16626 of file chan_sip.c.

References sip_send_all_mwi_subscriptions(), and sip_send_all_registers().

Referenced by network_change_event_cb().

{
   network_change_event_sched_id = -1;
   sip_send_all_registers();
   sip_send_all_mwi_subscriptions();
   return 0;
}

static void network_change_event_subscribe

(

void

)

[static]

Definition at line 16595 of file chan_sip.c.

References AST_EVENT_IE_END, AST_EVENT_NETWORK_CHANGE, ast_event_subscribe(), and network_change_event_cb().

Referenced by load_module(), and reload_config().

{
   if (!network_change_event_subscription) {
      network_change_event_subscription = ast_event_subscribe(AST_EVENT_NETWORK_CHANGE,
         network_change_event_cb, "SIP Network Change", NULL, AST_EVENT_IE_END);
   }
}

static void network_change_event_unsubscribe

(

void

)

[static]

Definition at line 16603 of file chan_sip.c.

References ast_event_unsubscribe().

Referenced by reload_config(), and unload_module().

{
   if (network_change_event_subscription) {
      network_change_event_subscription = ast_event_unsubscribe(network_change_event_subscription);
   }
}

static struct sip_proxy* obproxy_get	(	struct sip_pvt *	dialog,
		struct sip_peer *	peer
	)		[static, read]

Get default outbound proxy or global proxy.

Definition at line 3619 of file chan_sip.c.

References append_history, ast_debug, and sip_cfg.

Referenced by __sip_subscribe_mwi_do(), create_addr(), create_addr_from_peer(), and transmit_register().

{
   if (dialog && dialog->options && dialog->options->outboundproxy) {
      if (sipdebug) {
         ast_debug(1, "OBPROXY: Applying dialplan set OBproxy to this call\n");
      }
      append_history(dialog, "OBproxy", "Using dialplan obproxy %s", dialog->options->outboundproxy->name);
      return dialog->options->outboundproxy;
   }
   if (peer && peer->outboundproxy) {
      if (sipdebug) {
         ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
      }
      append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
      return peer->outboundproxy;
   }
   if (sip_cfg.outboundproxy.name[0]) {
      if (sipdebug) {
         ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
      }
      append_history(dialog, "OBproxy", "Using global obproxy %s", sip_cfg.outboundproxy.name);
      return &sip_cfg.outboundproxy;
   }
   if (sipdebug) {
      ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
   }
   return NULL;
}

static void offered_media_list_destroy

(

struct sip_pvt *

p

)

[static]

Destroy SDP media offer list.

Definition at line 6463 of file chan_sip.c.

References ast_free, and AST_LIST_REMOVE_HEAD.

Referenced by __sip_destroy(), process_sdp(), transmit_invite(), and transmit_reinvite_with_sdp().

{
   struct offered_media *offer;
   while ((offer = AST_LIST_REMOVE_HEAD(&p->offered_media, next))) {
      ast_free(offer->decline_m_line);
      ast_free(offer);
   }
}

static void on_dns_update_mwi	(	struct ast_sockaddr *	old,
		struct ast_sockaddr *	new,
		void *	data
	)		[static]

Definition at line 14325 of file chan_sip.c.

References ast_debug, ast_sockaddr_copy(), ast_sockaddr_isnull(), and ast_sockaddr_stringify().

Referenced by __sip_subscribe_mwi_do().

{
   struct sip_subscription_mwi *mwi = data;
   const char *old_str;

   /* This shouldn't happen, but just in case */
   if (ast_sockaddr_isnull(new)) {
      ast_debug(1, "Empty sockaddr change...ignoring!\n");
      return;
   }

   old_str = ast_strdupa(ast_sockaddr_stringify(old));
   ast_debug(1, "Changing mwi %s from %s to %s\n", mwi->hostname, old_str, ast_sockaddr_stringify(new));
   ast_sockaddr_copy(&mwi->us, new);
}

static void on_dns_update_peer	(	struct ast_sockaddr *	old,
		struct ast_sockaddr *	new,
		void *	data
	)		[static]

Definition at line 14296 of file chan_sip.c.

References ao2_link, ao2_lock, ao2_unlink, ao2_unlock, ast_debug, ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), and default_sip_port().

Referenced by build_peer(), and transmit_register().

{
   struct sip_peer *peer = data;
   const char *old_str;

   /* This shouldn't happen, but just in case */
   if (ast_sockaddr_isnull(new)) {
      ast_debug(1, "Empty sockaddr change...ignoring!\n");
      return;
   }

   if (!ast_sockaddr_isnull(&peer->addr)) {
      ao2_unlink(peers_by_ip, peer);
   }

   if (!ast_sockaddr_port(new)) {
      ast_sockaddr_set_port(new, default_sip_port(peer->socket.type));
   }

   old_str = ast_strdupa(ast_sockaddr_stringify(old));
   ast_debug(1, "Changing peer %s address from %s to %s\n", peer->name, old_str, ast_sockaddr_stringify(new));

   ao2_lock(peer);
   ast_sockaddr_copy(&peer->addr, new);
   ao2_unlock(peer);

   ao2_link(peers_by_ip, peer);
}

static void on_dns_update_registry	(	struct ast_sockaddr *	old,
		struct ast_sockaddr *	new,
		void *	data
	)		[static]

Definition at line 14275 of file chan_sip.c.

References ast_debug, ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_sockaddr_stringify(), and S_OR.

Referenced by transmit_register().

{
   struct sip_registry *reg = data;
   const char *old_str;

   /* This shouldn't happen, but just in case */
   if (ast_sockaddr_isnull(new)) {
      ast_debug(1, "Empty sockaddr change...ignoring!\n");
      return;
   }

   if (!ast_sockaddr_port(new)) {
      ast_sockaddr_set_port(new, reg->portno);
   }

   old_str = ast_strdupa(ast_sockaddr_stringify(old));

   ast_debug(1, "Changing registry %s from %s to %s\n", S_OR(reg->peername, reg->hostname), old_str, ast_sockaddr_stringify(new));
   ast_sockaddr_copy(&reg->us, new);
}

static unsigned int parse_allowed_methods

(

struct sip_request *

req

)

[static]

parse the Allow header to see what methods the endpoint we are communicating with allows.

We parse the allow header on incoming Registrations and save the result to the SIP peer that is registering. When the registration expires, we clear what we know about the peer's allowed methods. When the peer re-registers, we once again parse to see if the list of allowed methods has changed.

For peers that do not register, we parse the first message we receive during a call to see what is allowed, and save the information for the duration of the call.

Parameters:

req	The SIP request we are parsing

Return values:

The	methods allowed

Definition at line 9494 of file chan_sip.c.

References ast_strip_quoted(), ast_strlen_zero(), mark_parsed_methods(), and sip_get_header().

Referenced by set_pvt_allowed_methods().

{
   char *allow = ast_strdupa(sip_get_header(req, "Allow"));
   unsigned int allowed_methods = SIP_UNKNOWN;

   if (ast_strlen_zero(allow)) {
      /* I have witnessed that REGISTER requests from Polycom phones do not
       * place the phone's allowed methods in an Allow header. Instead, they place the
       * allowed methods in a methods= parameter in the Contact header.
       */
      char *contact = ast_strdupa(sip_get_header(req, "Contact"));
      char *methods = strstr(contact, ";methods=");

      if (ast_strlen_zero(methods)) {
         /* RFC 3261 states:
          *
          * "The absence of an Allow header field MUST NOT be
          * interpreted to mean that the UA sending the message supports no
          * methods.   Rather, it implies that the UA is not providing any
          * information on what methods it supports."
          *
          * For simplicity, we'll assume that the peer allows all known
          * SIP methods if they have no Allow header. We can then clear out the necessary
          * bits if the peer lets us know that we have sent an unsupported method.
          */
         return UINT_MAX;
      }
      allow = ast_strip_quoted(methods + 9, "\"", "\"");
   }
   mark_parsed_methods(&allowed_methods, allow);
   return allowed_methods;
}

static void parse_copy	(	struct sip_request *	dst,
		const struct sip_request *	src
	)		[static]

Copy SIP request, parse it.

Definition at line 4615 of file chan_sip.c.

References copy_request(), and parse_request().

Referenced by send_request(), and send_response().

{
   copy_request(dst, src);
   parse_request(dst);
}

int parse_minse	(	const char *	p_hdrval,
		int *const	p_interval
	)		[static]

Session-Timers: Function for parsing Min-SE header.

Definition at line 29327 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strlen_zero(), and LOG_WARNING.

Referenced by handle_request_invite_st(), and proc_422_rsp().

{
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Min-SE header\n");
      return -1;
   }

   *p_interval = 0;
   p_hdrval = ast_skip_blanks(p_hdrval);
   if (!sscanf(p_hdrval, "%30d", p_interval)) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return -1;
   }

   ast_debug(2, "Received Min-SE: %d\n", *p_interval);
   return 0;
}

static void parse_moved_contact	(	struct sip_pvt *	p,
		struct sip_request *	req,
		char **	name,
		char **	number,
		int	set_call_forward
	)		[static]

Parse 302 Moved temporalily response.

Todo:

XXX Doesn't redirect over TLS on sips: uri's. If we get a redirect to a SIPS: uri, this needs to be going back to the dialplan (this is a request for a secure signalling path). Note that transport=tls is deprecated, but we need to support it on incoming requests.

Definition at line 22365 of file chan_sip.c.

References ao2_ref, ast_copy_string(), ast_debug, ast_log(), ast_strdup, ast_strlen_zero(), ast_test_flag, ast_uri_decode(), ast_uri_sip_user, ast_websocket_unref(), find_closing_quote(), get_in_brackets(), LOG_NOTICE, pbx_builtin_setvar_helper(), remove_uri_parameters(), set_socket_transport(), sip_get_header(), and sip_get_transport().

Referenced by change_redirecting_information().

{
   char contact[SIPBUFSIZE];
   char *contact_name = NULL;
   char *contact_number = NULL;
   char *separator, *trans;
   char *domain;
   enum sip_transport transport = SIP_TRANSPORT_UDP;

   ast_copy_string(contact, sip_get_header(req, "Contact"), sizeof(contact));
   if ((separator = strchr(contact, ',')))
      *separator = '\0';

   contact_number = get_in_brackets(contact);
   if ((trans = strcasestr(contact_number, ";transport="))) {
      trans += 11;

      if ((separator = strchr(trans, ';')))
         *separator = '\0';

      if (!strncasecmp(trans, "tcp", 3))
         transport = SIP_TRANSPORT_TCP;
      else if (!strncasecmp(trans, "tls", 3))
         transport = SIP_TRANSPORT_TLS;
      else {
         if (strncasecmp(trans, "udp", 3))
            ast_debug(1, "received contact with an invalid transport, '%s'\n", contact_number);
         /* This will assume UDP for all unknown transports */
         transport = SIP_TRANSPORT_UDP;
      }
   }
   contact_number = remove_uri_parameters(contact_number);

   if (p->socket.tcptls_session) {
      ao2_ref(p->socket.tcptls_session, -1);
      p->socket.tcptls_session = NULL;
   } else if (p->socket.ws_session) {
      ast_websocket_unref(p->socket.ws_session);
      p->socket.ws_session = NULL;
   }

   set_socket_transport(&p->socket, transport);

   if (set_call_forward && ast_test_flag(&p->flags[0], SIP_PROMISCREDIR)) {
      char *host = NULL;
      if (!strncasecmp(contact_number, "sip:", 4))
         contact_number += 4;
      else if (!strncasecmp(contact_number, "sips:", 5))
         contact_number += 5;
      separator = strchr(contact_number, '/');
      if (separator)
         *separator = '\0';
      if ((host = strchr(contact_number, '@'))) {
         *host++ = '\0';
         ast_debug(2, "Found promiscuous redirection to 'SIP/%s::::%s@%s'\n", contact_number, sip_get_transport(transport), host);
         if (p->owner)
            ast_channel_call_forward_build(p->owner, "SIP/%s::::%s@%s", contact_number, sip_get_transport(transport), host);
      } else {
         ast_debug(2, "Found promiscuous redirection to 'SIP/::::%s@%s'\n", sip_get_transport(transport), contact_number);
         if (p->owner)
            ast_channel_call_forward_build(p->owner, "SIP/::::%s@%s", sip_get_transport(transport), contact_number);
      }
   } else {
      separator = strchr(contact, '@');
      if (separator) {
         *separator++ = '\0';
         domain = separator;
      } else {
         /* No username part */
         domain = contact;
      }
      separator = strchr(contact, '/');   /* WHEN do we hae a forward slash in the URI? */
      if (separator)
         *separator = '\0';

      if (!strncasecmp(contact_number, "sip:", 4))
         contact_number += 4;
      else if (!strncasecmp(contact_number, "sips:", 5))
         contact_number += 5;
      separator = strchr(contact_number, ';');  /* And username ; parameters? */
      if (separator)
         *separator = '\0';
      ast_uri_decode(contact_number, ast_uri_sip_user);
      if (set_call_forward) {
         ast_debug(2, "Received 302 Redirect to extension '%s' (domain %s)\n", contact_number, domain);
         if (p->owner) {
            pbx_builtin_setvar_helper(p->owner, "SIPDOMAIN", domain);
            ast_channel_call_forward_set(p->owner, contact_number);
         }
      }
   }

   /* We've gotten the number for the contact, now get the name */

   if (*contact == '\"') {
      contact_name = contact + 1;
      if (!(separator = (char *)find_closing_quote(contact_name, NULL))) {
         ast_log(LOG_NOTICE, "No closing quote on name in Contact header? %s\n", contact);
      }
      *separator = '\0';
   }

   if (name && !ast_strlen_zero(contact_name)) {
      *name = ast_strdup(contact_name);
   }
   if (number) {
      *number = ast_strdup(contact_number);
   }
}

static int parse_ok_contact	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)		[static]

Save contact header for 200 OK on INVITE.

Definition at line 15892 of file chan_sip.c.

References ast_copy_string(), ast_string_field_set, get_in_brackets(), sip_get_header(), and TRUE.

Referenced by forked_invite_init(), handle_request_invite(), handle_request_subscribe(), and handle_response_invite().

{
   char contact[SIPBUFSIZE];
   char *c;

   /* Look for brackets */
   ast_copy_string(contact, sip_get_header(req, "Contact"), sizeof(contact));
   c = get_in_brackets(contact);

   /* Save full contact to call pvt for later bye or re-invite */
   ast_string_field_set(pvt, fullcontact, c);

   /* Save URI for later ACKs, BYE or RE-invites */
   ast_string_field_set(pvt, okcontacturi, c);

   /* We should return false for URI:s we can't handle,
      like tel:, mailto:,ldap: etc */
   return TRUE;      
}

static void parse_oli	(	struct sip_request *	req,
		struct ast_channel *	chan
	)		[static]

Check for the presence of OLI tag(s) in the From header and set on the channel.

Definition at line 25987 of file chan_sip.c.

References ast_party_caller::ani2, ast_channel_caller(), ast_strlen_zero(), and sip_get_header().

Referenced by handle_request_invite().

{
   const char *from = NULL;
   const char *s = NULL;
   int ani2 = 0;

   if (!chan || !req) {
      /* null pointers are not helpful */
      return;
   }

   from = sip_get_header(req, "From");
   if (ast_strlen_zero(from)) {
      /* no From header */
      return;
   }

   /* Look for the possible OLI tags. */
   if ((s = strcasestr(from, ";isup-oli="))) {
      s += 10;
   } else if ((s = strcasestr(from, ";ss7-oli="))) {
      s += 9;
   } else if ((s = strcasestr(from, ";oli="))) {
      s += 5;
   }

   if (ast_strlen_zero(s)) {
      /* OLI tag is missing, or present with nothing following the '=' sign */
      return;
   }

   /* just in case OLI is quoted */
   if (*s == '\"') {
      s++;
   }

   if (sscanf(s, "%d", &ani2)) {
      ast_channel_caller(chan)->ani2 = ani2;
   }

   return;
}

static enum parse_register_result parse_register_contact	(	struct sip_pvt *	pvt,
		struct sip_peer *	p,
		struct sip_request *	req
	)		[static]

Parse contact header and save registration (peer registration)

Todo:

Check NAPTR/SRV if we have not got a port in the URI

Definition at line 15999 of file chan_sip.c.

References __get_header(), ao2_t_link, ao2_t_unlink, ast_apply_acl(), ast_copy_string(), ast_db_put(), ast_debug, ast_log(), ast_sched_add(), AST_SCHED_DEL_UNREF, ast_sched_when(), AST_SENSE_ALLOW, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_resolve_first_transport(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_string_field_build, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_verb, copy_socket_data(), default_expiry, default_sip_port(), EVENT_FLAG_SYSTEM, expire_register(), FALSE, get_in_brackets(), get_transport_str2enum(), LOG_NOTICE, LOG_WARNING, manager_event, max_expiry, min_expiry, parse_uri_legacy_check(), register_peer_exten(), set_socket_transport(), sip_cfg, sip_get_header(), sip_poke_peer(), sip_pvt_lock, sip_pvt_unlock, sip_ref_peer(), sip_unref_peer(), and TRUE.

Referenced by register_verify().

{
   char contact[SIPBUFSIZE];
   char data[SIPBUFSIZE];
   const char *expires = sip_get_header(req, "Expires");
   int expire = atoi(expires);
   char *curi = NULL, *hostport = NULL, *transport = NULL;
   int transport_type;
   const char *useragent;
   struct ast_sockaddr oldsin, testsa;
   char *firstcuri = NULL;
   int start = 0;
   int wildcard_found = 0;
   int single_binding_found = 0;

   ast_copy_string(contact, __get_header(req, "Contact", &start), sizeof(contact));

   if (ast_strlen_zero(expires)) {  /* No expires header, try look in Contact: */
      char *s = strcasestr(contact, ";expires=");
      if (s) {
         expires = strsep(&s, ";"); /* trim ; and beyond */
         if (sscanf(expires + 9, "%30d", &expire) != 1) {
            expire = default_expiry;
         }
      } else {
         /* Nothing has been specified */
         expire = default_expiry;
      }
   }

   if (expire > max_expiry) {
      expire = max_expiry;
   }
   if (expire < min_expiry && expire != 0) {
      expire = min_expiry;
   }
   pvt->expiry = expire;

   copy_socket_data(&pvt->socket, &req->socket);

   do {
      /* Look for brackets */
      curi = contact;
      if (strchr(contact, '<') == NULL)   /* No <, check for ; and strip it */
         strsep(&curi, ";");  /* This is Header options, not URI options */
      curi = get_in_brackets(contact);
      if (!firstcuri) {
         firstcuri = ast_strdupa(curi);
      }

      if (!strcasecmp(curi, "*")) {
         wildcard_found = 1;
      } else {
         single_binding_found = 1;
      }

      if (wildcard_found && (ast_strlen_zero(expires) || expire != 0 || single_binding_found)) {
         /* Contact header parameter "*" detected, so punt if: Expires header is missing,
          * Expires value is not zero, or another Contact header is present. */
         return PARSE_REGISTER_FAILED;
      }

      ast_copy_string(contact, __get_header(req, "Contact", &start), sizeof(contact));
   } while (!ast_strlen_zero(contact));
   curi = firstcuri;

   /* if they did not specify Contact: or Expires:, they are querying
      what we currently have stored as their contact address, so return
      it
   */
   if (ast_strlen_zero(curi) && ast_strlen_zero(expires)) {
      /* If we have an active registration, tell them when the registration is going to expire */
      if (peer->expire > -1 && !ast_strlen_zero(peer->fullcontact)) {
         pvt->expiry = ast_sched_when(sched, peer->expire);
      }
      return PARSE_REGISTER_QUERY;
   } else if (!strcasecmp(curi, "*") || !expire) { /* Unregister this peer */
      /* This means remove all registrations and return OK */
      AST_SCHED_DEL_UNREF(sched, peer->expire,
            sip_unref_peer(peer, "remove register expire ref"));
      ast_verb(3, "Unregistered SIP '%s'\n", peer->name);
      expire_register(sip_ref_peer(peer,"add ref for explicit expire_register"));
      return PARSE_REGISTER_UPDATE;
   }

   /* Store whatever we got as a contact from the client */
   ast_string_field_set(peer, fullcontact, curi);

   /* For the 200 OK, we should use the received contact */
   ast_string_field_build(pvt, our_contact, "<%s>", curi);

   /* Make sure it's a SIP URL */
   if (ast_strlen_zero(curi) || parse_uri_legacy_check(curi, "sip:,sips:", &curi, NULL, &hostport, &transport)) {
      ast_log(LOG_NOTICE, "Not a valid SIP contact (missing sip:/sips:) trying to use anyway\n");
   }

   /* handle the transport type specified in Contact header. */
   if (!(transport_type = get_transport_str2enum(transport))) {
      transport_type = pvt->socket.type;
   }

   /* if the peer's socket type is different than the Registration
    * transport type, change it.  If it got this far, it is a
    * supported type, but check just in case */
   if ((peer->socket.type != transport_type) && (peer->transports & transport_type)) {
      set_socket_transport(&peer->socket, transport_type);
   }

   oldsin = peer->addr;

   /* If we were already linked into the peers_by_ip container unlink ourselves so nobody can find us */
   if (!ast_sockaddr_isnull(&peer->addr) && (!peer->is_realtime || ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS))) {
      ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
   }

   if ((transport_type != SIP_TRANSPORT_WS) && (transport_type != SIP_TRANSPORT_WSS) &&
       (!ast_test_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT) && !ast_test_flag(&peer->flags[0], SIP_NAT_RPORT_PRESENT))) {
      /* use the data provided in the Contact header for call routing */
      ast_debug(1, "Store REGISTER's Contact header for call routing.\n");
      /* XXX This could block for a long time XXX */
      /*! \todo Check NAPTR/SRV if we have not got a port in the URI */
      if (ast_sockaddr_resolve_first_transport(&testsa, hostport, 0, peer->socket.type)) {
         ast_log(LOG_WARNING, "Invalid hostport '%s'\n", hostport);
         ast_string_field_set(peer, fullcontact, "");
         ast_string_field_set(pvt, our_contact, "");
         return PARSE_REGISTER_FAILED;
      }

      /* If we have a port number in the given URI, make sure we do remember to not check for NAPTR/SRV records.
         The hostport part is actually a host. */
      peer->portinuri = ast_sockaddr_port(&testsa) ? TRUE : FALSE;

      if (!ast_sockaddr_port(&testsa)) {
         ast_sockaddr_set_port(&testsa, default_sip_port(transport_type));
      }

      ast_sockaddr_copy(&peer->addr, &testsa);
   } else {
      /* Don't trust the contact field.  Just use what they came to us
         with */
      ast_debug(1, "Store REGISTER's src-IP:port for call routing.\n");
      peer->addr = pvt->recv;
   }

   /* Check that they're allowed to register at this IP */
   if (ast_apply_acl(sip_cfg.contact_acl, &peer->addr, "SIP contact ACL: ") != AST_SENSE_ALLOW ||
         ast_apply_acl(peer->contactacl, &peer->addr, "SIP contact ACL: ") != AST_SENSE_ALLOW) {
      ast_log(LOG_WARNING, "Domain '%s' disallowed by contact ACL (violating IP %s)\n", hostport,
            ast_sockaddr_stringify_addr(&peer->addr));
      ast_string_field_set(peer, fullcontact, "");
      ast_string_field_set(pvt, our_contact, "");
      return PARSE_REGISTER_DENIED;
   }

   /* if the Contact header information copied into peer->addr matches the
    * received address, and the transport types are the same, then copy socket
    * data into the peer struct */
   if ((peer->socket.type == pvt->socket.type) &&
      !ast_sockaddr_cmp(&peer->addr, &pvt->recv)) {
      copy_socket_data(&peer->socket, &pvt->socket);
   }

   /* Now that our address has been updated put ourselves back into the container for lookups */
   if (!peer->is_realtime || ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      ao2_t_link(peers_by_ip, peer, "ao2_link into peers_by_ip table");
   }

   /* Save SIP options profile */
   peer->sipoptions = pvt->sipoptions;

   if (!ast_strlen_zero(curi) && ast_strlen_zero(peer->username)) {
      ast_string_field_set(peer, username, curi);
   }

   AST_SCHED_DEL_UNREF(sched, peer->expire,
         sip_unref_peer(peer, "remove register expire ref"));

   if (peer->is_realtime && !ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
      peer->expire = -1;
   } else {
      peer->expire = ast_sched_add(sched, (expire + 10) * 1000, expire_register,
            sip_ref_peer(peer, "add registration ref"));
      if (peer->expire == -1) {
         sip_unref_peer(peer, "remote registration ref");
      }
   }
   snprintf(data, sizeof(data), "%s:%d:%s:%s", ast_sockaddr_stringify(&peer->addr),
       expire, peer->username, peer->fullcontact);
   /* We might not immediately be able to reconnect via TCP, but try caching it anyhow */
   if (!peer->rt_fromcontact || !sip_cfg.peer_rtupdate)
      ast_db_put("SIP/Registry", peer->name, data);
   manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\n", peer->name,  ast_sockaddr_stringify(&peer->addr));

   /* Is this a new IP address for us? */
   if (ast_sockaddr_cmp(&peer->addr, &oldsin)) {
      ast_verb(3, "Registered SIP '%s' at %s\n", peer->name,
         ast_sockaddr_stringify(&peer->addr));
   }
   sip_pvt_unlock(pvt);
   sip_poke_peer(peer, 0);
   sip_pvt_lock(pvt);
   register_peer_exten(peer, 1);

   /* Save User agent */
   useragent = sip_get_header(req, "User-Agent");
   if (strcasecmp(useragent, peer->useragent)) {
      ast_string_field_set(peer, useragent, useragent);
      ast_verb(4, "Saved useragent \"%s\" for peer %s\n", peer->useragent, peer->name);
   }
   return PARSE_REGISTER_UPDATE;
}

static int parse_request

(

struct sip_request *

req

)

[static]

Parse a SIP message.

Note:

this function is used both on incoming and outgoing packets

Definition at line 9599 of file chan_sip.c.

References ast_debug, ast_log(), ast_str_buffer(), ast_str_strlen(), ast_strlen_zero(), determine_firstline_parts(), and LOG_WARNING.

Referenced by handle_request_do(), initialize_initreq(), parse_copy(), and sip_tls_read().

{
   char *c = req->data->str;
   ptrdiff_t *dst = req->header;
   int i = 0, lim = SIP_MAX_HEADERS - 1;
   unsigned int skipping_headers = 0;
   ptrdiff_t current_header_offset = 0;
   char *previous_header = "";

   req->header[0] = 0;
   req->headers = -1;   /* mark that we are working on the header */
   for (; *c; c++) {
      if (*c == '\r') {    /* remove \r */
         *c = '\0';
      } else if (*c == '\n') {   /* end of this line */
         *c = '\0';
         current_header_offset = (c + 1) - ast_str_buffer(req->data);
         previous_header = ast_str_buffer(req->data) + dst[i];
         if (skipping_headers) {
            /* check to see if this line is blank; if so, turn off
               the skipping flag, so the next line will be processed
               as a body line */
            if (ast_strlen_zero(previous_header)) {
               skipping_headers = 0;
            }
            dst[i] = current_header_offset; /* record start of next line */
            continue;
         }
         if (sipdebug) {
            ast_debug(4, "%7s %2d [%3d]: %s\n",
                 req->headers < 0 ? "Header" : "Body",
                 i, (int) strlen(previous_header), previous_header);
         }
         if (ast_strlen_zero(previous_header) && req->headers < 0) {
            req->headers = i; /* record number of header lines */
            dst = req->line;  /* start working on the body */
            i = 0;
            lim = SIP_MAX_LINES - 1;
         } else { /* move to next line, check for overflows */
            if (i++ == lim) {
               /* if we're processing headers, then skip any remaining
                  headers and move on to processing the body, otherwise
                  we're done */
               if (req->headers != -1) {
                  break;
               } else {
                  req->headers = i;
                  dst = req->line;
                  i = 0;
                  lim = SIP_MAX_LINES - 1;
                  skipping_headers = 1;
               }
            }
         }
         dst[i] = current_header_offset; /* record start of next line */
      }
   }

   /* Check for last header or body line without CRLF. The RFC for SDP requires CRLF,
      but since some devices send without, we'll be generous in what we accept. However,
      if we've already reached the maximum number of lines for portion of the message
      we were parsing, we can't accept any more, so just ignore it.
   */
   previous_header = ast_str_buffer(req->data) + dst[i];
   if ((i < lim) && !ast_strlen_zero(previous_header)) {
      if (sipdebug) {
         ast_debug(4, "%7s %2d [%3d]: %s\n",
              req->headers < 0 ? "Header" : "Body",
              i, (int) strlen(previous_header), previous_header );
      }
      i++;
   }

   /* update count of header or body lines */
   if (req->headers >= 0) {   /* we are in the body */
      req->lines = i;
   } else {       /* no body */
      req->headers = i;
      req->lines = 0;
      /* req->data->used will be a NULL byte */
      req->line[0] = ast_str_strlen(req->data);
   }

   if (*c) {
      ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
   }

   /* Split up the first line parts */
   return determine_firstline_parts(req);
}

int parse_session_expires	(	const char *	p_hdrval,
		int *const	p_interval,
		enum st_refresher_param *const	p_ref
	)		[static]

Session-Timers: Function for parsing Session-Expires header.

Definition at line 29347 of file chan_sip.c.

References ast_debug, ast_log(), ast_skip_blanks(), ast_strlen_zero(), and LOG_WARNING.

Referenced by handle_request_invite_st(), and handle_response_invite().

{
   char *p_token;
   int  ref_idx;
   char *p_se_hdr;

   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "Null Session-Expires header\n");
      return -1;
   }

   *p_ref = SESSION_TIMER_REFRESHER_PARAM_UNKNOWN;
   *p_interval = 0;

   p_se_hdr = ast_strdupa(p_hdrval);
   p_se_hdr = ast_skip_blanks(p_se_hdr);

   while ((p_token = strsep(&p_se_hdr, ";"))) {
      p_token = ast_skip_blanks(p_token);
      if (!sscanf(p_token, "%30d", p_interval)) {
         ast_log(LOG_WARNING, "Parsing of Session-Expires failed\n");
         return -1;
      }

      ast_debug(2, "Session-Expires: %d\n", *p_interval);

      if (!p_se_hdr)
         continue;

      p_se_hdr = ast_skip_blanks(p_se_hdr);
      ref_idx = strlen("refresher=");
      if (!strncasecmp(p_se_hdr, "refresher=", ref_idx)) {
         p_se_hdr += ref_idx;
         p_se_hdr = ast_skip_blanks(p_se_hdr);

         if (!strncasecmp(p_se_hdr, "uac", strlen("uac"))) {
            *p_ref = SESSION_TIMER_REFRESHER_PARAM_UAC;
            ast_debug(2, "Refresher: UAC\n");
         } else if (!strncasecmp(p_se_hdr, "uas", strlen("uas"))) {
            *p_ref = SESSION_TIMER_REFRESHER_PARAM_UAS;
            ast_debug(2, "Refresher: UAS\n");
         } else {
            ast_log(LOG_WARNING, "Invalid refresher value %s\n", p_se_hdr);
            return -1;
         }
         break;
      }
   }
   return 0;
}

static int parse_uri_legacy_check	(	char *	uri,
		const char *	scheme,
		char **	user,
		char **	pass,
		char **	hostport,
		char **	transport
	)		[static]

parse uri in a way that allows semicolon stripping if legacy mode is enabled

Note:

This calls parse_uri which has the unexpected property that passing more arguments results in more splitting. Most common is to leave out the pass argument, causing user to contain user:pass if available.

Definition at line 15918 of file chan_sip.c.

References parse_uri(), and sip_cfg.

Referenced by __set_address_from_contact(), check_user_full(), get_also_info(), get_destination(), parse_register_contact(), and register_verify().

{
   int ret = parse_uri(uri, scheme, user, pass, hostport, transport);
   if (sip_cfg.legacy_useroption_parsing) { /* if legacy mode is active, strip semis from the user field */
      char *p;
      if ((p = strchr(uri, (int)';'))) {
         *p = '\0';
      }
   }
   return ret;
}

static int peer_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Note:

The only member of the peer used here is the name field

Definition at line 33486 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_peer *peer = obj, *peer2 = arg;

   return !strcasecmp(peer->name, peer2->name) ? CMP_MATCH | CMP_STOP : 0;
}

static int peer_dump_func	(	void *	userobj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 19340 of file chan_sip.c.

References ao2_t_ref, ast_cli(), and ast_cli_args::fd.

Referenced by sip_show_objects().

{
   struct sip_peer *peer = userobj;
   int refc = ao2_t_ref(userobj, 0, "");
   struct ast_cli_args *a = (struct ast_cli_args *) arg;
   
   ast_cli(a->fd, "name: %s\ntype: peer\nobjflags: %d\nrefcount: %d\n\n",
      peer->name, 0, refc);
   return 0;
}

static int peer_hash_cb	(	const void *	obj,
		const int	flags
	)		[static]

Note:

The only member of the peer used here is the name field

Definition at line 33476 of file chan_sip.c.

References ast_str_case_hash().

Referenced by load_module().

{
   const struct sip_peer *peer = obj;

   return ast_str_case_hash(peer->name);
}

static int peer_ipcmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 33567 of file chan_sip.c.

References peer_ipcmp_cb_full().

Referenced by load_module().

{
   return peer_ipcmp_cb_full(obj, arg, NULL, flags);
}

static int peer_ipcmp_cb_full	(	void *	obj,
		void *	arg,
		void *	data,
		int	flags
	)		[static]

Match Peers by IP and Port number.

This function has two modes.

• If the peer arg does not have INSECURE_PORT set, then we will only return a match for a peer that matches both the IP and port.
• If the peer arg does have the INSECURE_PORT flag set, then we will only return a match for a peer that matches the IP and has insecure=port in its configuration.

This callback will be used twice when doing peer matching. There is a first pass for full IP+port matching, and a second pass in case there is a match that meets the insecure=port criteria.

Note:

Connections coming in over TCP or TLS should never be matched by port.

the peer's addr struct provides to fields combined to make a key: the sin_addr.s_addr and sin_port fields.

Definition at line 33534 of file chan_sip.c.

References ast_sockaddr_cmp_addr(), ast_sockaddr_port, ast_strlen_zero(), ast_test_flag, CMP_MATCH, and CMP_STOP.

Referenced by peer_ipcmp_cb(), and sip_find_peer_full().

{
   struct sip_peer *peer = obj, *peer2 = arg;
   char *callback = data;

   if (!ast_strlen_zero(callback) && strcasecmp(peer->callback, callback)) {
      /* We require a callback extension match, but don't have one */
      return 0;
   }

   /* At this point, we match the callback extension if we need to. Carry on. */

   if (ast_sockaddr_cmp_addr(&peer->addr, &peer2->addr)) {
      /* IP doesn't match */
      return 0;
   }

   /* We matched the IP, check to see if we need to match by port as well. */
   if ((peer->transports & peer2->transports) & (SIP_TRANSPORT_TLS | SIP_TRANSPORT_TCP)) {
      /* peer matching on port is not possible with TCP/TLS */
      return CMP_MATCH | CMP_STOP;
   } else if (ast_test_flag(&peer2->flags[0], SIP_INSECURE_PORT)) {
      /* We are allowing match without port for peers configured that
       * way in this pass through the peers. */
      return ast_test_flag(&peer->flags[0], SIP_INSECURE_PORT) ?
            (CMP_MATCH | CMP_STOP) : 0;
   }

   /* Now only return a match if the port matches, as well. */
   return ast_sockaddr_port(&peer->addr) == ast_sockaddr_port(&peer2->addr) ?
         (CMP_MATCH | CMP_STOP) : 0;
}

static int peer_iphash_cb	(	const void *	obj,
		const int	flags
	)		[static]

Hash function based on the the peer's ip address. For IPv6, we use the end of the address.

Todo:

Find a better hashing function

Definition at line 33498 of file chan_sip.c.

References ast_log(), ast_sockaddr_hash(), ast_sockaddr_isnull(), and LOG_ERROR.

Referenced by load_module().

{
   const struct sip_peer *peer = obj;
   int ret = 0;

   if (ast_sockaddr_isnull(&peer->addr)) {
      ast_log(LOG_ERROR, "Empty address\n");
   }

   ret = ast_sockaddr_hash(&peer->addr);

   if (ret < 0) {
      ret = -ret;
   }

   return ret;
}

static void peer_mailboxes_to_str	(	struct ast_str **	mailbox_str,
		struct sip_peer *	peer
	)		[static]

list peer mailboxes to CLI

Definition at line 19963 of file chan_sip.c.

References AST_LIST_NEXT, AST_LIST_TRAVERSE, ast_str_append(), ast_strlen_zero(), mailbox, and S_OR.

Referenced by _sip_show_peer(), function_sippeer(), show_channels_cb(), and sip_send_mwi_to_peer().

{
   struct sip_mailbox *mailbox;

   AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
      ast_str_append(mailbox_str, 0, "%s%s%s%s",
         mailbox->mailbox,
         ast_strlen_zero(mailbox->context) ? "" : "@",
         S_OR(mailbox->context, ""),
         AST_LIST_NEXT(mailbox, entry) ? "," : "");
   }
}

static int peer_markall_autopeers_func	(	void *	device,
		void *	arg,
		int	flags
	)		[static]

Definition at line 31352 of file chan_sip.c.

Referenced by reload_config().

{
   struct sip_peer *peer = device;
   if (peer->selfdestruct) {
      peer->the_mark = 1;
   }
   return 0;
}

static int peer_markall_func	(	void *	device,
		void *	arg,
		int	flags
	)		[static]

Definition at line 31343 of file chan_sip.c.

Referenced by reload_config().

{
   struct sip_peer *peer = device;
   if (!peer->selfdestruct) {
      peer->the_mark = 1;
   }
   return 0;
}

static void peer_sched_cleanup

(

struct sip_peer *

peer

)

[static]

Definition at line 3291 of file chan_sip.c.

References AST_SCHED_DEL_UNREF, and sip_unref_peer().

Referenced by match_and_cleanup_peer_sched().

{
   if (peer->pokeexpire != -1) {
      AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
            sip_unref_peer(peer, "removing poke peer ref"));
   }
   if (peer->expire != -1) {
      AST_SCHED_DEL_UNREF(sched, peer->expire,
            sip_unref_peer(peer, "remove register expire ref"));
   }
   if (peer->keepalivesend != -1) {
      AST_SCHED_DEL_UNREF(sched, peer->keepalivesend,
                sip_unref_peer(peer, "remove keepalive peer ref"));
   }
}

static int peer_status	(	struct sip_peer *	peer,
		char *	status,
		int	statuslen
	)		[static]

Definition at line 18878 of file chan_sip.c.

References ast_copy_string().

Referenced by _sip_show_peer(), _sip_show_peers_one(), and function_sippeer().

{
   int res = 0;
   if (peer->maxms) {
      if (peer->lastms < 0) {
         ast_copy_string(status, "UNREACHABLE", statuslen);
      } else if (peer->lastms > peer->maxms) {
         snprintf(status, statuslen, "LAGGED (%d ms)", peer->lastms);
         res = 1;
      } else if (peer->lastms) {
         snprintf(status, statuslen, "OK (%d ms)", peer->lastms);
         res = 1;
      } else {
         ast_copy_string(status, "UNKNOWN", statuslen);
      }
   } else {
      ast_copy_string(status, "Unmonitored", statuslen);
      /* Checking if port is 0 */
      res = -1;
   }
   return res;
}

int peercomparefunc	(	const void *	a,
		const void *	b
	)

Definition at line 19106 of file chan_sip.c.

Referenced by _sip_show_peers().

{
   struct sip_peer **ap = (struct sip_peer **)a;
   struct sip_peer **bp = (struct sip_peer **)b;
   return strcmp((*ap)->name, (*bp)->name);
}

static int peers_data_provider_get	(	const struct ast_data_search *	search,
		struct ast_data *	data_root
	)		[static]

Definition at line 34457 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_lock, ao2_ref, ao2_unlock, ARRAY_LEN, ast_cdr_flags2str(), ast_data_add_bool(), ast_data_add_codecs(), ast_data_add_int(), ast_data_add_node(), ast_data_add_str(), ast_data_add_structure, ast_data_remove_node(), ast_data_search_match(), ast_describe_caller_presentation(), AST_LIST_TRAVERSE, get_transport_list(), mailbox, text, and transfermode2str().

{
   struct sip_peer *peer;
   struct ao2_iterator i;
   struct ast_data *data_peer, *data_peer_mailboxes = NULL, *data_peer_mailbox, *enum_node;
   struct ast_data *data_sip_options;
   int total_mailboxes, x;
   struct sip_mailbox *mailbox;

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_iterator_next(&i))) {
      ao2_lock(peer);

      data_peer = ast_data_add_node(data_root, "peer");
      if (!data_peer) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }

      ast_data_add_structure(sip_peer, data_peer, peer);

      /* transfer mode */
      enum_node = ast_data_add_node(data_peer, "allowtransfer");
      if (!enum_node) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }
      ast_data_add_str(enum_node, "text", transfermode2str(peer->allowtransfer));
      ast_data_add_int(enum_node, "value", peer->allowtransfer);

      /* transports */
      ast_data_add_str(data_peer, "transports", get_transport_list(peer->transports));

      /* peer type */
      if ((peer->type & SIP_TYPE_USER) && (peer->type & SIP_TYPE_PEER)) {
         ast_data_add_str(data_peer, "type", "friend");
      } else if (peer->type & SIP_TYPE_PEER) {
         ast_data_add_str(data_peer, "type", "peer");
      } else if (peer->type & SIP_TYPE_USER) {
         ast_data_add_str(data_peer, "type", "user");
      }

      /* mailboxes */
      total_mailboxes = 0;
      AST_LIST_TRAVERSE(&peer->mailboxes, mailbox, entry) {
         if (!total_mailboxes) {
            data_peer_mailboxes = ast_data_add_node(data_peer, "mailboxes");
            if (!data_peer_mailboxes) {
               break;
            }
            total_mailboxes++;
         }

         data_peer_mailbox = ast_data_add_node(data_peer_mailboxes, "mailbox");
         if (!data_peer_mailbox) {
            continue;
         }
         ast_data_add_str(data_peer_mailbox, "mailbox", mailbox->mailbox);
         ast_data_add_str(data_peer_mailbox, "context", mailbox->context);
      }

      /* amaflags */
      enum_node = ast_data_add_node(data_peer, "amaflags");
      if (!enum_node) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }
      ast_data_add_int(enum_node, "value", peer->amaflags);
      ast_data_add_str(enum_node, "text", ast_cdr_flags2str(peer->amaflags));

      /* sip options */
      data_sip_options = ast_data_add_node(data_peer, "sipoptions");
      if (!data_sip_options) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }
      for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
         ast_data_add_bool(data_sip_options, sip_options[x].text, peer->sipoptions & sip_options[x].id);
      }

      /* callingpres */
      enum_node = ast_data_add_node(data_peer, "callingpres");
      if (!enum_node) {
         ao2_unlock(peer);
         ao2_ref(peer, -1);
         continue;
      }
      ast_data_add_int(enum_node, "value", peer->callingpres);
      ast_data_add_str(enum_node, "text", ast_describe_caller_presentation(peer->callingpres));

      /* codecs */
      ast_data_add_codecs(data_peer, "codecs", peer->caps);

      if (!ast_data_search_match(search, data_peer)) {
         ast_data_remove_node(data_root, data_peer);
      }

      ao2_unlock(peer);
      ao2_ref(peer, -1);
   }
   ao2_iterator_destroy(&i);

   return 0;
}

static int pidf_validate_presence

(

struct ast_xml_doc *

doc

)

[static]

Definition at line 27199 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), ast_xml_find_namespace(), ast_xml_free_attr(), ast_xml_get_attribute(), ast_xml_get_ns_href(), ast_xml_get_root(), ast_xml_node_get_children(), ast_xml_node_get_name(), ast_xml_node_get_next(), entity, FALSE, LOG_WARNING, pidf_validate_tuple(), and TRUE.

Referenced by sip_pidf_validate().

{
   struct ast_xml_node *presence_node = ast_xml_get_root(doc);
   struct ast_xml_node *child_nodes;
   struct ast_xml_node *node_iterator;
   struct ast_xml_ns *ns;
   const char *entity;
   const char *namespace;
   const char presence_namespace[] = "urn:ietf:params:xml:ns:pidf";

   if (!presence_node) {
      ast_log(LOG_WARNING, "Unable to retrieve root node of the XML document\n");
      return FALSE;
   }
   /* Okay, we managed to open the document! YAY! Now, let's start making sure it's all PIDF-ified
    * correctly.
    */
   if (strcmp(ast_xml_node_get_name(presence_node), "presence")) {
      ast_log(LOG_WARNING, "Root node of PIDF document is not 'presence'. Invalid\n");
      return FALSE;
   }

   /* The presence element must have an entity attribute and an xmlns attribute. Furthermore
    * the xmlns attribute must be "urn:ietf:params:xml:ns:pidf"
    */
   if (!(entity = ast_xml_get_attribute(presence_node, "entity"))) {
      ast_log(LOG_WARNING, "Presence element of PIDF document has no 'entity' attribute\n");
      return FALSE;
   }
   /* We're not interested in what the entity is, just that it exists */
   ast_xml_free_attr(entity);

   if (!(ns = ast_xml_find_namespace(doc, presence_node, NULL))) {
      ast_log(LOG_WARNING, "Couldn't find default namespace...\n");
      return FALSE;
   }

   namespace = ast_xml_get_ns_href(ns);
   if (ast_strlen_zero(namespace) || strcmp(namespace, presence_namespace)) {
      ast_log(LOG_WARNING, "PIDF document has invalid namespace value %s\n", namespace);
      return FALSE;
   }

   if (!(child_nodes = ast_xml_node_get_children(presence_node))) {
      ast_log(LOG_WARNING, "PIDF document has no elements as children of 'presence'. Invalid\n");
      return FALSE;
   }

   /* Check for tuple elements. RFC 3863 says that PIDF documents can have any number of
    * tuples, including 0. The big thing here is that if there are tuple elements present,
    * they have to have a single status element within.
    *
    * The RFC is worded such that tuples should appear as the first elements as children of
    * the presence element. However, we'll be accepting of documents which may place other elements
    * before the tuple(s).
    */
   for (node_iterator = child_nodes; node_iterator;
         node_iterator = ast_xml_node_get_next(node_iterator)) {
      if (strcmp(ast_xml_node_get_name(node_iterator), "tuple")) {
         /* Not a tuple. We don't give a rat's hind quarters */
         continue;
      }
      if (pidf_validate_tuple(node_iterator) == FALSE) {
         ast_log(LOG_WARNING, "Unable to validate tuple\n");
         return FALSE;
      }
   }

   return TRUE;
}

static int pidf_validate_tuple

(

struct ast_xml_node *

tuple_node

)

[static]

Definition at line 27159 of file chan_sip.c.

References ast_log(), ast_xml_free_attr(), ast_xml_get_attribute(), ast_xml_node_get_children(), ast_xml_node_get_name(), ast_xml_node_get_next(), FALSE, id, LOG_WARNING, and TRUE.

Referenced by pidf_validate_presence().

{
   const char *id;
   int status_found = FALSE;
   struct ast_xml_node *tuple_children;
   struct ast_xml_node *tuple_children_iterator;
   /* Tuples have to have an id attribute or they're invalid */
   if (!(id = ast_xml_get_attribute(tuple_node, "id"))) {
      ast_log(LOG_WARNING, "Tuple XML element has no attribute 'id'\n");
      return FALSE;
   }
   /* We don't care what it actually is, just that it's there */
   ast_xml_free_attr(id);
   /* This is a tuple. It must have a status element */
   if (!(tuple_children = ast_xml_node_get_children(tuple_node))) {
      /* The tuple has no children. It sucks */
      ast_log(LOG_WARNING, "Tuple XML element has no child elements\n");
      return FALSE;
   }
   for (tuple_children_iterator = tuple_children; tuple_children_iterator;
         tuple_children_iterator = ast_xml_node_get_next(tuple_children_iterator)) {
      /* Similar to the wording used regarding tuples, the status element should appear
       * first. However, we will once again relax things and accept the status at any
       * position. We will enforce that only a single status element can be present.
       */
      if (strcmp(ast_xml_node_get_name(tuple_children_iterator), "status")) {
         /* Not the status, we don't care */
         continue;
      }
      if (status_found == TRUE) {
         /* THERE CAN BE ONLY ONE!!! */
         ast_log(LOG_WARNING, "Multiple status elements found in tuple. Only one allowed\n");
         return FALSE;
      }
      status_found = TRUE;
   }
   return status_found;
}

unsigned int port_str2int	(	const char *	pt,
		unsigned int	standard
	)

converts ascii port to int representation. If no pt buffer is provided or the pt has errors when being converted to an int value, the port provided as the standard is used.

Definition at line 3608 of file chan_sip.c.

References ast_strlen_zero().

Referenced by build_peer(), reload_config(), and sip_parse_register_line().

{
   int port = standard;
   if (ast_strlen_zero(pt) || (sscanf(pt, "%30d", &port) != 1) || (port < 1) || (port > 65535)) {
      port = standard;
   }

   return port;
}

static void print_codec_to_cli	(	int	fd,
		struct ast_codec_pref *	pref
	)		[static]

Print codec list from preference to CLI/manager.

Definition at line 19714 of file chan_sip.c.

References ast_cli(), ast_codec_pref_index(), AST_CODEC_PREF_SIZE, ast_getformatname(), and ast_codec_pref::framing.

Referenced by _sip_show_peer(), sip_show_settings(), and sip_show_user().

{
   int x;
   struct ast_format codec;

   for(x = 0; x < AST_CODEC_PREF_SIZE; x++) {
      if (!(ast_codec_pref_index(pref, x, &codec))) {
         break;
      }
      ast_cli(fd, "%s", ast_getformatname(&codec));
      ast_cli(fd, ":%d", pref->framing[x]);
      if (x < 31 && ast_codec_pref_index(pref, x + 1, &codec))
         ast_cli(fd, ",");
   }
   if (!x)
      ast_cli(fd, "none");
}

static void print_group	(	int	fd,
		ast_group_t	group,
		int	crlf
	)		[static]

Print call group and pickup group.

Definition at line 19392 of file chan_sip.c.

References ast_cli(), and ast_print_group().

Referenced by _sip_show_peer(), and sip_show_user().

{
   char buf[256];
   ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_group(buf, sizeof(buf), group) );
}

static void print_named_groups	(	int	fd,
		struct ast_namedgroups *	groups,
		int	crlf
	)		[static]

Print named call groups and pickup groups.

Definition at line 19399 of file chan_sip.c.

References ast_cli(), ast_free, ast_print_namedgroups(), and ast_str_create().

Referenced by _sip_show_peer(), and sip_show_user().

{
   struct ast_str *buf = ast_str_create(1024);
   if (buf) {
      ast_cli(fd, crlf ? "%s\r\n" : "%s\n", ast_print_namedgroups(&buf, group) );
      ast_free(buf);
   }
}

static void proc_422_rsp	(	struct sip_pvt *	p,
		struct sip_request *	rsp
	)		[static]

Handle 422 response to INVITE with session-timer requested.

Session-Timers: An INVITE originated by Asterisk that asks for session-timers support from the UAS can result into a 422 response. This is how a UAS or an intermediary proxy server tells Asterisk that the session refresh interval offered by Asterisk is too low for them. The proc_422_rsp() function handles a 422 response. It extracts the Min-SE header that comes back in 422 and sends a new INVITE accordingly.

Definition at line 29406 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), LOG_WARNING, parse_minse(), sip_get_header(), and transmit_invite().

Referenced by handle_response_invite().

{
   int rtn;
   const char *p_hdrval;
   int minse;

   p_hdrval = sip_get_header(rsp, "Min-SE");
   if (ast_strlen_zero(p_hdrval)) {
      ast_log(LOG_WARNING, "422 response without a Min-SE header %s\n", p_hdrval);
      return;
   }
   rtn = parse_minse(p_hdrval, &minse);
   if (rtn != 0) {
      ast_log(LOG_WARNING, "Parsing of Min-SE header failed %s\n", p_hdrval);
      return;
   }
   p->stimer->st_cached_min_se = minse;
   if (p->stimer->st_interval < minse) {
      p->stimer->st_interval = minse;
   }
   transmit_invite(p, SIP_INVITE, 1, 2, NULL);
}

static int proc_session_timer

(

const void *

vp

)

[static]

Session-Timers: Process session refresh timeout event.

Definition at line 29259 of file chan_sip.c.

References ast_channel_name(), ast_channel_trylock, ast_channel_uniqueid(), ast_channel_unlock, ast_debug, ast_log(), AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_UP, EVENT_FLAG_CALL, FALSE, LOG_WARNING, manager_event, sip_pvt_lock, sip_pvt_unlock, stop_session_timer(), transmit_reinvite_with_sdp(), and TRUE.

Referenced by start_session_timer().

{
   struct sip_pvt *p = (struct sip_pvt *) vp;
   int res = 0;

   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in proc_session_timer - %s\n", p->callid);
      goto return_unref;
   }

   ast_debug(2, "Session timer expired: %d - %s\n", p->stimer->st_schedid, p->callid);

   if (!p->owner) {
      goto return_unref;
   }

   if ((p->stimer->st_active != TRUE) || (ast_channel_state(p->owner) != AST_STATE_UP)) {
      goto return_unref;
   }

   if (p->stimer->st_ref == SESSION_TIMER_REFRESHER_US) {
      res = 1;
      if (T38_ENABLED == p->t38.state) {
         transmit_reinvite_with_sdp(p, TRUE, TRUE);
      } else {
         transmit_reinvite_with_sdp(p, FALSE, TRUE);
      }
   } else {
      if (p->stimer->quit_flag) {
         goto return_unref;
      }
      ast_log(LOG_WARNING, "Session-Timer expired - %s\n", p->callid);
      sip_pvt_lock(p);
      while (p->owner && ast_channel_trylock(p->owner)) {
         sip_pvt_unlock(p);
         usleep(1);
         if (p->stimer && p->stimer->quit_flag) {
            goto return_unref;
         }
         sip_pvt_lock(p);
      }

      manager_event(EVENT_FLAG_CALL, "SessionTimeout", "Source: SIPSessionTimer\r\n"
            "Channel: %s\r\nUniqueid: %s\r\n", ast_channel_name(p->owner), ast_channel_uniqueid(p->owner));
      ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
      ast_channel_unlock(p->owner);
      sip_pvt_unlock(p);
   }

return_unref:
   if (!res) {
      /* An error occurred.  Stop session timer processing */
      if (p->stimer) {
         ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
         p->stimer->st_schedid = -1;
         stop_session_timer(p);
      }

      /* If we are not asking to be rescheduled, then we need to release our
       * reference to the dialog. */
      dialog_unref(p, "removing session timer ref");
   }

   return res;
}

static int process_crypto	(	struct sip_pvt *	p,
		struct ast_rtp_instance *	rtp,
		struct sip_srtp **	srtp,
		const char *	a
	)		[static]

Definition at line 33303 of file chan_sip.c.

References ast_debug, ast_log(), ast_rtp_instance_get_dtls(), ast_set_flag, ast_test_flag, FALSE, LOG_WARNING, sdp_crypto_process(), sdp_crypto_setup(), setup_srtp(), ast_rtp_engine_dtls::stop, and TRUE.

Referenced by process_sdp().

{
   struct ast_rtp_engine_dtls *dtls;

   /* If no RTP instance exists for this media stream don't bother processing the crypto line */
   if (!rtp) {
      ast_debug(3, "Received offer with crypto line for media stream that is not enabled\n");
      return FALSE;
   }

   if (strncasecmp(a, "crypto:", 7)) {
      return FALSE;
   }
   if (!*srtp) {
      if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         ast_log(LOG_WARNING, "Ignoring unexpected crypto attribute in SDP answer\n");
         return FALSE;
      }

      if (setup_srtp(srtp) < 0) {
         return FALSE;
      }
   }

   if (!(*srtp)->crypto && !((*srtp)->crypto = sdp_crypto_setup())) {
      return FALSE;
   }

   if (sdp_crypto_process((*srtp)->crypto, a, rtp, *srtp) < 0) {
      return FALSE;
   }

   ast_set_flag(*srtp, SRTP_CRYPTO_OFFER_OK);

   if ((dtls = ast_rtp_instance_get_dtls(rtp))) {
      dtls->stop(rtp);
      p->dtls_cfg.enabled = 0;
   }

   return TRUE;
}

static int process_sdp	(	struct sip_pvt *	p,
		struct sip_request *	req,
		int	t38action
	)		[static]

Process SIP SDP offer, select formats and activate media channels If offer is rejected, we will not change any properties of the call Return 0 on success, a negative value on errors. Must be called after find_sdp().

< RTP audio destination IP address

< RTP video destination IP address

< RTP text destination IP address

< UDPTL image destination IP address

< RTP audio destination port number

< RTP video destination port number

< RTP text destination port number

< UDPTL image destination port number

Definition at line 9908 of file chan_sip.c.

References ast_rtp_engine_dtls::active, ast_async_goto(), ast_calloc, ast_channel_caller(), ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_macrocontext(), ast_channel_name(), ast_channel_nativeformats(), ast_channel_readformat(), ast_channel_set_fd(), ast_channel_unlock, ast_channel_writeformat(), ast_clear_flag, ast_codec_choose(), AST_CONTROL_HOLD, AST_CONTROL_UNHOLD, ast_debug, ast_exists_extension(), ast_format_cap_alloc_nolock(), ast_format_cap_append(), ast_format_cap_copy(), ast_format_cap_destroy(), ast_format_cap_has_type(), ast_format_cap_is_empty(), ast_format_cap_iscompatible(), ast_format_cap_joint_append(), ast_format_cap_joint_copy(), ast_format_cap_set(), ast_format_set(), AST_FORMAT_T140RED, AST_FORMAT_TYPE_AUDIO, ast_getformatname_multiple(), AST_LIST_INSERT_TAIL, ast_log(), ast_malloc, ast_null_frame, ast_queue_control(), ast_queue_control_data(), ast_queue_frame(), ast_rtp_codecs_payload_formats(), ast_rtp_codecs_payloads_copy(), ast_rtp_codecs_payloads_destroy(), ast_rtp_codecs_payloads_initialize(), ast_rtp_codecs_payloads_set_m_type(), AST_RTP_DTMF, ast_rtp_instance_fd(), ast_rtp_instance_get_codecs(), ast_rtp_instance_get_dtls(), ast_rtp_instance_get_remote_address(), ast_rtp_instance_set_prop(), ast_rtp_instance_set_remote_address(), ast_rtp_instance_stop(), ast_rtp_lookup_mime_multiple2(), AST_RTP_PROPERTY_DTMF, AST_RTP_PROPERTY_DTMF_COMPENSATE, AST_RTP_PROPERTY_RTCP, ast_rtp_red_init(), ast_set_flag, ast_set_read_format(), ast_set_write_format(), ast_skip_blanks(), ast_sockaddr_isnull(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_str_alloca, ast_strlen_zero(), ast_test_flag, ast_udptl_get_far_max_datagram(), ast_udptl_set_error_correction_scheme(), ast_udptl_set_far_max_datagram(), ast_udptl_set_peer(), ast_udptl_stop(), ast_verb, ast_verbose(), change_hold_state(), change_t38_state(), debug, FALSE, get_sdp_iterate(), get_sdp_line(), has_media_stream(), ast_party_caller::id, initialize_udptl(), len(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, ast_party_id::number, offered_media_list_destroy(), pbx_builtin_setvar_helper(), process_crypto(), process_sdp_a_audio(), process_sdp_a_dtls(), process_sdp_a_ice(), process_sdp_a_image(), process_sdp_a_sendonly(), process_sdp_a_text(), process_sdp_a_video(), process_sdp_c(), process_sdp_o(), S_COR, S_OR, sip_debug_test_pvt(), sip_srtp_alloc(), sockaddr_is_null_or_any(), start_ice(), ast_party_number::str, text, TRUE, type, UDPTL_ERROR_CORRECTION_NONE, ast_party_number::valid, and value.

Referenced by handle_incoming(), handle_request_invite(), handle_response(), and handle_response_invite().

{
   int res = 0;

   /* Iterators for SDP parsing */
   int start = req->sdp_start;
   int next = start;
   int iterator = start;

   /* Temporary vars for SDP parsing */
   char type = '\0';
   const char *value = NULL;
   const char *m = NULL;           /* SDP media offer */
   const char *nextm = NULL;
   int len = -1;
   struct offered_media *offer;

   /* Host information */
   struct ast_sockaddr sessionsa;
   struct ast_sockaddr audiosa;
   struct ast_sockaddr videosa;
   struct ast_sockaddr textsa;
   struct ast_sockaddr imagesa;
   struct ast_sockaddr *sa = NULL;     /*!< RTP audio destination IP address */
   struct ast_sockaddr *vsa = NULL; /*!< RTP video destination IP address */
   struct ast_sockaddr *tsa = NULL; /*!< RTP text destination IP address */
   struct ast_sockaddr *isa = NULL; /*!< UDPTL image destination IP address */
   int portno = -1;        /*!< RTP audio destination port number */
   int vportno = -1;       /*!< RTP video destination port number */
   int tportno = -1;       /*!< RTP text destination port number */
   int udptlportno = -1;         /*!< UDPTL image destination port number */

   /* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */
   struct ast_format_cap *peercapability = ast_format_cap_alloc_nolock();
   struct ast_format_cap *vpeercapability = ast_format_cap_alloc_nolock();
   struct ast_format_cap *tpeercapability = ast_format_cap_alloc_nolock();

   int peernoncodeccapability = 0, vpeernoncodeccapability = 0, tpeernoncodeccapability = 0;

   struct ast_rtp_codecs newaudiortp = { 0, }, newvideortp = { 0, }, newtextrtp = { 0, };
   struct ast_format_cap *newjointcapability = ast_format_cap_alloc_nolock(); /* Negotiated capability */
   struct ast_format_cap *newpeercapability = ast_format_cap_alloc_nolock();
   int newnoncodeccapability;

   const char *codecs;
   int codec;

   /* SRTP */
   int secure_audio = FALSE;
   int secure_video = FALSE;

   /* Others */
   int sendonly = -1;
   int numberofports;
   int last_rtpmap_codec = 0;
   int red_data_pt[10];    /* For T.140 RED */
   int red_num_gen = 0;    /* For T.140 RED */
   char red_fmtp[100] = "empty"; /* For T.140 RED */
   int debug = sip_debug_test_pvt(p);

   /* START UNKNOWN */
   char buf[SIPBUFSIZE];
   struct ast_format tmp_fmt;
   /* END UNKNOWN */

   /* Initial check */
   if (!p->rtp) {
      ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
      res = -1;
      goto process_sdp_cleanup;
   }
   if (!peercapability || !vpeercapability || !tpeercapability || !newpeercapability || !newjointcapability) {
      res = -1;
      goto process_sdp_cleanup;
   }

   if (ast_rtp_codecs_payloads_initialize(&newaudiortp) || ast_rtp_codecs_payloads_initialize(&newvideortp) ||
       ast_rtp_codecs_payloads_initialize(&newtextrtp)) {
      res = -1;
      goto process_sdp_cleanup;
   }

   /* Update our last rtprx when we receive an SDP, too */
   p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */

   offered_media_list_destroy(p);

   /* Scan for the first media stream (m=) line to limit scanning of globals */
   nextm = get_sdp_iterate(&next, req, "m");
   if (ast_strlen_zero(nextm)) {
      ast_log(LOG_WARNING, "Insufficient information for SDP (m= not found)\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   /* Scan session level SDP parameters (lines before first media stream) */
   while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
      int processed = FALSE;
      switch (type) {
      case 'o':
         /* If we end up receiving SDP that doesn't actually modify the session we don't want to treat this as a fatal
          * error. We just want to ignore the SDP and let the rest of the packet be handled as normal.
          */
         if (!process_sdp_o(value, p)) {
            res = (p->session_modify == FALSE) ? 0 : -1;
            goto process_sdp_cleanup;
         }
         processed = TRUE;
         break;
      case 'c':
         if (process_sdp_c(value, &sessionsa)) {
            processed = TRUE;
            sa = &sessionsa;
            vsa = sa;
            tsa = sa;
            isa = sa;
         }
         break;
      case 'a':
         if (process_sdp_a_sendonly(value, &sendonly)) {
            processed = TRUE;
         }
         else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec))
            processed = TRUE;
         else if (process_sdp_a_image(value, p))
            processed = TRUE;

         if (process_sdp_a_ice(value, p, p->rtp)) {
            processed = TRUE;
         }
         if (process_sdp_a_ice(value, p, p->vrtp)) {
            processed = TRUE;
         }
         if (process_sdp_a_ice(value, p, p->trtp)) {
            processed = TRUE;
         }

         if (process_sdp_a_dtls(value, p, p->rtp)) {
            processed = TRUE;
         }
         if (process_sdp_a_dtls(value, p, p->vrtp)) {
            processed = TRUE;
         }
         if (process_sdp_a_dtls(value, p, p->trtp)) {
            processed = TRUE;
         }

         break;
      }

      ast_debug(3, "Processing session-level SDP %c=%s... %s\n", type, value, (processed == TRUE)? "OK." : "UNSUPPORTED OR FAILED.");
   }

   /* default: novideo and notext set */
   p->novideo = TRUE;
   p->notext = TRUE;

   /* Scan media stream (m=) specific parameters loop */
   while (!ast_strlen_zero(nextm)) {
      int audio = FALSE;
      int video = FALSE;
      int image = FALSE;
      int text = FALSE;
      int processed_crypto = FALSE;
      char protocol[18] = {0,};
      int x;
      struct ast_rtp_engine_dtls *dtls;

      numberofports = 0;
      len = -1;
      start = next;
      m = nextm;
      iterator = next;
      nextm = get_sdp_iterate(&next, req, "m");

      if (!(offer = ast_calloc(1, sizeof(*offer)))) {
         ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer list\n");
         res = -1;
         goto process_sdp_cleanup;
      }
      AST_LIST_INSERT_TAIL(&p->offered_media, offer, next);
      offer->type = SDP_UNKNOWN;

      /* Check for 'audio' media offer */
      if (strncmp(m, "audio ", 6) == 0) {
         if ((sscanf(m, "audio %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
             (sscanf(m, "audio %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
            codecs = m + len;
            /* produce zero-port m-line since it may be needed later
             * length is "m=audio 0 " + protocol + " " + codecs + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be exactly the right length */
            sprintf(offer->decline_m_line, "m=audio 0 %s %s\r\n", protocol, codecs);

            if (x == 0) {
               ast_log(LOG_WARNING, "Ignoring audio media offer because port number is zero\n");
               continue;
            }

            if (has_media_stream(p, SDP_AUDIO)) {
               ast_log(LOG_WARNING, "Declining non-primary audio stream: %s\n", m);
               continue;
            }

            /* Check number of ports offered for stream */
            if (numberofports > 1) {
               ast_log(LOG_WARNING, "%d ports offered for audio media, not supported by Asterisk. Will try anyway...\n", numberofports);
            }

            if ((!strcmp(protocol, "RTP/SAVPF") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               if (req->method != SIP_RESPONSE) {
                  ast_log(LOG_NOTICE, "Received SAVPF profle in audio offer but AVPF is not enabled, enabling: %s\n", m);
                  secure_audio = 1;
                  ast_set_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
               }
               else {

                  ast_log(LOG_WARNING, "Received SAVPF profle in audio answer but AVPF is not enabled: %s\n", m);
                  continue;
               }
            } else if ((!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVP")) && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               if (req->method != SIP_RESPONSE) {
                  ast_log(LOG_NOTICE, "Received SAVP profle in audio offer but AVPF is enabled, disabling: %s\n", m);
                  secure_audio = 1;
                  ast_clear_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
               }
               else {
                  ast_log(LOG_WARNING, "Received SAVP profile in audio offer but AVPF is enabled: %s\n", m);
                  continue;
               }
            } else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
               secure_audio = 1;

               if (p->srtp) {
                  ast_set_flag(p->srtp, SRTP_CRYPTO_OFFER_OK);
               }
            } else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
               secure_audio = 1;
            } else if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               if (req->method != SIP_RESPONSE) {
                  ast_log(LOG_NOTICE, "Received AVPF profile in audio offer but AVPF is not enabled, enabling: %s\n", m);
                  ast_set_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
               }
               else {
                  ast_log(LOG_WARNING, "Received AVP profile in audio answer but AVPF is enabled: %s\n", m);
                  continue;
               }
            } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               if (req->method != SIP_RESPONSE) {
                  ast_log(LOG_NOTICE, "Received AVP profile in audio answer but AVPF is enabled, disabling: %s\n", m);
                  ast_clear_flag(&p->flags[2], SIP_PAGE3_USE_AVPF);
               }
               else {
                  ast_log(LOG_WARNING, "Received AVP profile in audio answer but AVPF is enabled: %s\n", m);
                  continue;
               }
            } else if ((!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) &&
                  (!(dtls = ast_rtp_instance_get_dtls(p->rtp)) || !dtls->active(p->rtp))) {
               ast_log(LOG_WARNING, "Received UDP/TLS in audio offer but DTLS is not enabled: %s\n", m);
               continue;
            } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
               ast_log(LOG_WARNING, "Unknown RTP profile in audio offer: %s\n", m);
               continue;
            }

            audio = TRUE;
            offer->type = SDP_AUDIO;
            portno = x;

            /* Scan through the RTP payload types specified in a "m=" line: */
            for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
               if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
                  ast_log(LOG_WARNING, "Invalid syntax in RTP audio format list: %s\n", codecs);
                  res = -1;
                  goto process_sdp_cleanup;
               }
               if (debug) {
                  ast_verbose("Found RTP audio format %d\n", codec);
               }

               ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
            }
         } else {
            ast_log(LOG_WARNING, "Rejecting audio media offer due to invalid or unsupported syntax: %s\n", m);
            res = -1;
            goto process_sdp_cleanup;
         }
      }
      /* Check for 'video' media offer */
      else if (strncmp(m, "video ", 6) == 0) {
         if ((sscanf(m, "video %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
             (sscanf(m, "video %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
            codecs = m + len;
            /* produce zero-port m-line since it may be needed later
             * length is "m=video 0 " + protocol + " " + codecs + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be exactly the right length */
            sprintf(offer->decline_m_line, "m=video 0 %s %s\r\n", protocol, codecs);

            if (x == 0) {
               ast_log(LOG_WARNING, "Ignoring video stream offer because port number is zero\n");
               continue;
            }

            /* Check number of ports offered for stream */
            if (numberofports > 1) {
               ast_log(LOG_WARNING, "%d ports offered for video stream, not supported by Asterisk. Will try anyway...\n", numberofports);
            }

            if (has_media_stream(p, SDP_VIDEO)) {
               ast_log(LOG_WARNING, "Declining non-primary video stream: %s\n", m);
               continue;
            }

            if ((!strcmp(protocol, "RTP/SAVPF") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received SAVPF profle in video offer but AVPF is not enabled: %s\n", m);
               continue;
            } else if ((!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVP")) && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received SAVP profile in video offer but AVPF is enabled: %s\n", m);
               continue;
            } else if (!strcmp(protocol, "UDP/TLS/RTP/SAVP") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) {
               secure_video = 1;

               if (p->vsrtp || (p->vsrtp = sip_srtp_alloc())) {
                  ast_set_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK);
               }
            } else if (!strcmp(protocol, "RTP/SAVP") || !strcmp(protocol, "RTP/SAVPF")) {
               secure_video = 1;
            } else if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received AVPF profile in video offer but AVPF is not enabled: %s\n", m);
               continue;
            } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received AVP profile in video offer but AVPF is enabled: %s\n", m);
               continue;
            } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
               ast_log(LOG_WARNING, "Unknown RTP profile in video offer: %s\n", m);
               continue;
            }

            video = TRUE;
            p->novideo = FALSE;
            offer->type = SDP_VIDEO;
            vportno = x;

            /* Scan through the RTP payload types specified in a "m=" line: */
            for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
               if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
                  ast_log(LOG_WARNING, "Invalid syntax in RTP video format list: %s\n", codecs);
                  res = -1;
                  goto process_sdp_cleanup;
               }
               if (debug) {
                  ast_verbose("Found RTP video format %d\n", codec);
               }
               ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
            }
         } else {
            ast_log(LOG_WARNING, "Rejecting video media offer due to invalid or unsupported syntax: %s\n", m);
            res = -1;
            goto process_sdp_cleanup;
         }
      }
      /* Check for 'text' media offer */
      else if (strncmp(m, "text ", 5) == 0) {
         if ((sscanf(m, "text %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) ||
             (sscanf(m, "text %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
            codecs = m + len;
            /* produce zero-port m-line since it may be needed later
             * length is "m=text 0 " + protocol + " " + codecs + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(9 + strlen(protocol) + 1 + strlen(codecs) + 3))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be exactly the right length */
            sprintf(offer->decline_m_line, "m=text 0 %s %s\r\n", protocol, codecs);

            if (x == 0) {
               ast_log(LOG_WARNING, "Ignoring text stream offer because port number is zero\n");
               continue;
            }

            /* Check number of ports offered for stream */
            if (numberofports > 1) {
               ast_log(LOG_WARNING, "%d ports offered for text stream, not supported by Asterisk. Will try anyway...\n", numberofports);
            }

            if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received AVPF profile in text offer but AVPF is not enabled: %s\n", m);
               continue;
            } else if (!strcmp(protocol, "RTP/AVP") && ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
               ast_log(LOG_WARNING, "Received AVP profile in text offer but AVPF is enabled: %s\n", m);
               continue;
            } else if (strcmp(protocol, "RTP/AVP") && strcmp(protocol, "RTP/AVPF")) {
               ast_log(LOG_WARNING, "Unknown RTP profile in text offer: %s\n", m);
               continue;
            }

            if (has_media_stream(p, SDP_TEXT)) {
               ast_log(LOG_WARNING, "Declining non-primary text stream: %s\n", m);
               continue;
            }

            text = TRUE;
            p->notext = FALSE;
            offer->type = SDP_TEXT;
            tportno = x;

            /* Scan through the RTP payload types specified in a "m=" line: */
            for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
               if (sscanf(codecs, "%30u%n", &codec, &len) != 1) {
                  ast_log(LOG_WARNING, "Invalid syntax in RTP video format list: %s\n", codecs);
                  res = -1;
                  goto process_sdp_cleanup;
               }
               if (debug) {
                  ast_verbose("Found RTP text format %d\n", codec);
               }
               ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
            }
         } else {
            ast_log(LOG_WARNING, "Rejecting text stream offer due to invalid or unsupported syntax: %s\n", m);
            res = -1;
            goto process_sdp_cleanup;
         }
      }
      /* Check for 'image' media offer */
      else if (strncmp(m, "image ", 6) == 0) {
         if (((sscanf(m, "image %30u udptl t38%n", &x, &len) == 1 && len > 0) ||
              (sscanf(m, "image %30u UDPTL t38%n", &x, &len) == 1 && len > 0))) {
            /* produce zero-port m-line since it may be needed later
             * length is "m=image 0 udptl t38" + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(22))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be exactly the right length */
            strcpy(offer->decline_m_line, "m=image 0 udptl t38\r\n");

            if (x == 0) {
               ast_log(LOG_WARNING, "Ignoring image stream offer because port number is zero\n");
               continue;
            }

            if (initialize_udptl(p)) {
               ast_log(LOG_WARNING, "Failed to initialize UDPTL, declining image stream\n");
               continue;
            }

            if (has_media_stream(p, SDP_IMAGE)) {
               ast_log(LOG_WARNING, "Declining non-primary image stream: %s\n", m);
               continue;
            }

            image = TRUE;
            if (debug) {
               ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
            }

            offer->type = SDP_IMAGE;
            udptlportno = x;

            if (p->t38.state != T38_ENABLED) {
               memset(&p->t38.their_parms, 0, sizeof(p->t38.their_parms));

               /* default EC to none, the remote end should
                * respond with the EC they want to use */
               ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
            }
         } else if (sscanf(m, "image %30u %17s t38%n", &x, protocol, &len) == 2 && len > 0) {
            ast_log(LOG_WARNING, "Declining image stream due to unsupported transport: %s\n", m);
            /* produce zero-port m-line since this is guaranteed to be declined
             * length is "m=image 0 strlen(protocol) t38" + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(10 + strlen(protocol) + 7))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be exactly the right length */
            sprintf(offer->decline_m_line, "m=image 0 %s t38\r\n", protocol);
            continue;
         } else {
            ast_log(LOG_WARNING, "Rejecting image media offer due to invalid or unsupported syntax: %s\n", m);
            res = -1;
            goto process_sdp_cleanup;
         }
      } else {
         char type[20] = {0,};
         if ((sscanf(m, "%19s %30u/%30u %n", type, &x, &numberofports, &len) == 3 && len > 0) ||
              (sscanf(m, "%19s %30u %n", type, &x, &len) == 2 && len > 0)) {
            /* produce zero-port m-line since it may be needed later
             * length is "m=" + type + " 0 " + remainder + "\r\n\0" */
            if (!(offer->decline_m_line = ast_malloc(2 + strlen(type) + 3 + strlen(m + len) + 3))) {
               ast_log(LOG_WARNING, "Failed to allocate memory for SDP offer declination\n");
               res = -1;
               goto process_sdp_cleanup;
            }
            /* guaranteed to be long enough */
            sprintf(offer->decline_m_line, "m=%s 0 %s\r\n", type, m + len);
            continue;
         } else {
            ast_log(LOG_WARNING, "Unsupported top-level media type in offer: %s\n", m);
            res = -1;
            goto process_sdp_cleanup;
         }
      }

      /* Media stream specific parameters */
      while ((type = get_sdp_line(&iterator, next - 1, req, &value)) != '\0') {
         int processed = FALSE;

         switch (type) {
         case 'c':
            if (audio) {
               if (process_sdp_c(value, &audiosa)) {
                  processed = TRUE;
                  sa = &audiosa;
               }
            } else if (video) {
               if (process_sdp_c(value, &videosa)) {
                  processed = TRUE;
                  vsa = &videosa;
               }
            } else if (text) {
               if (process_sdp_c(value, &textsa)) {
                  processed = TRUE;
                  tsa = &textsa;
               }
            } else if (image) {
               if (process_sdp_c(value, &imagesa)) {
                  processed = TRUE;
                  isa = &imagesa;
               }
            }
            break;
         case 'a':
            /* Audio specific scanning */
            if (audio) {
               if (process_sdp_a_ice(value, p, p->rtp)) {
                  processed = TRUE;
               } else if (process_sdp_a_dtls(value, p, p->rtp)) {
                  processed = TRUE;
               } else if (process_sdp_a_sendonly(value, &sendonly)) {
                  processed = TRUE;
               } else if (!processed_crypto && process_crypto(p, p->rtp, &p->srtp, value)) {
                  processed_crypto = TRUE;
                  processed = TRUE;
               } else if (process_sdp_a_audio(value, p, &newaudiortp, &last_rtpmap_codec)) {
                  processed = TRUE;
               }
            }
            /* Video specific scanning */
            else if (video) {
               if (process_sdp_a_ice(value, p, p->vrtp)) {
                  processed = TRUE;
               } else if (process_sdp_a_dtls(value, p, p->vrtp)) {
                  processed = TRUE;
               } else if (!processed_crypto && process_crypto(p, p->vrtp, &p->vsrtp, value)) {
                  processed_crypto = TRUE;
                  processed = TRUE;
               } else if (process_sdp_a_video(value, p, &newvideortp, &last_rtpmap_codec)) {
                  processed = TRUE;
               }
            }
            /* Text (T.140) specific scanning */
            else if (text) {
               if (process_sdp_a_ice(value, p, p->trtp)) {
                  processed = TRUE;
               } else if (process_sdp_a_text(value, p, &newtextrtp, red_fmtp, &red_num_gen, red_data_pt, &last_rtpmap_codec)) {
                  processed = TRUE;
               } else if (!processed_crypto && process_crypto(p, p->trtp, &p->tsrtp, value)) {
                  processed_crypto = TRUE;
                  processed = TRUE;
               }
            }
            /* Image (T.38 FAX) specific scanning */
            else if (image) {
               if (process_sdp_a_image(value, p))
                  processed = TRUE;
            }
            break;
         }

         ast_debug(3, "Processing media-level (%s) SDP %c=%s... %s\n",
              (audio == TRUE)? "audio" : (video == TRUE)? "video" : (text == TRUE)? "text" : "image",
              type, value,
              (processed == TRUE)? "OK." : "UNSUPPORTED OR FAILED.");
      }

      /* Ensure crypto lines are provided where necessary */
      if (audio && secure_audio && !processed_crypto) {
         ast_log(LOG_WARNING, "Rejecting secure audio stream without encryption details: %s\n", m);
         return -1;
      } else if (video && secure_video && !processed_crypto) {
         ast_log(LOG_WARNING, "Rejecting secure video stream without encryption details: %s\n", m);
         return -1;
      }
   }

   /* Sanity checks */
   if (!sa && !vsa && !tsa && !isa) {
      ast_log(LOG_WARNING, "Insufficient information in SDP (c=)...\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if ((portno == -1) &&
       (vportno == -1) &&
       (tportno == -1) &&
       (udptlportno == -1)) {
      ast_log(LOG_WARNING, "Failing due to no acceptable offer found\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (secure_audio && !(p->srtp && (ast_test_flag(p->srtp, SRTP_CRYPTO_OFFER_OK)))) {
      ast_log(LOG_WARNING, "Can't provide secure audio requested in SDP offer\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (!secure_audio && p->srtp) {
      ast_log(LOG_WARNING, "We are requesting SRTP for audio, but they responded without it!\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (secure_video && !(p->vsrtp && (ast_test_flag(p->vsrtp, SRTP_CRYPTO_OFFER_OK)))) {
      ast_log(LOG_WARNING, "Can't provide secure video requested in SDP offer\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (!p->novideo && !secure_video && p->vsrtp) {
      ast_log(LOG_WARNING, "We are requesting SRTP for video, but they responded without it!\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (!(secure_audio || secure_video) && ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
      ast_log(LOG_WARNING, "Matched device setup to use SRTP, but request was not!\n");
      res = -1;
      goto process_sdp_cleanup;
   }

   if (udptlportno == -1) {
      change_t38_state(p, T38_DISABLED);
   }

   /* Now gather all of the codecs that we are asked for: */
   ast_rtp_codecs_payload_formats(&newaudiortp, peercapability, &peernoncodeccapability);
   ast_rtp_codecs_payload_formats(&newvideortp, vpeercapability, &vpeernoncodeccapability);
   ast_rtp_codecs_payload_formats(&newtextrtp, tpeercapability, &tpeernoncodeccapability);

   ast_format_cap_append(newpeercapability, peercapability);
   ast_format_cap_append(newpeercapability, vpeercapability);
   ast_format_cap_append(newpeercapability, tpeercapability);

   ast_format_cap_joint_copy(p->caps, newpeercapability, newjointcapability);
   if (ast_format_cap_is_empty(newjointcapability) && udptlportno == -1) {
      ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
      /* Do NOT Change current setting */
      res = -1;
      goto process_sdp_cleanup;
   }

   newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;

   if (debug) {
      /* shame on whoever coded this.... */
      char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];

      ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
             ast_getformatname_multiple(s1, SIPBUFSIZE, p->caps),
             ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
             ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
             ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
             ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
   }
   if (debug) {
      struct ast_str *s1 = ast_str_alloca(SIPBUFSIZE);
      struct ast_str *s2 = ast_str_alloca(SIPBUFSIZE);
      struct ast_str *s3 = ast_str_alloca(SIPBUFSIZE);

      ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
             ast_rtp_lookup_mime_multiple2(s1, NULL, p->noncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple2(s2, NULL, peernoncodeccapability, 0, 0),
             ast_rtp_lookup_mime_multiple2(s3, NULL, newnoncodeccapability, 0, 0));
   }

   if (portno != -1 || vportno != -1 || tportno != -1) {
      /* We are now ready to change the sip session and RTP structures with the offered codecs, since
         they are acceptable */
      ast_format_cap_copy(p->jointcaps, newjointcapability);                /* Our joint codec profile for this call */
      ast_format_cap_copy(p->peercaps, newpeercapability);                  /* The other side's capability in latest offer */
      p->jointnoncodeccapability = newnoncodeccapability;     /* DTMF capabilities */

      /* respond with single most preferred joint codec, limiting the other side's choice */
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) {
         ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt);
         ast_format_cap_set(p->jointcaps, &tmp_fmt);
      }
   }

   /* Setup audio address and port */
   if (p->rtp) {
      if (sa && portno > 0) {
         start_ice(p->rtp);
         ast_sockaddr_set_port(sa, portno);
         ast_rtp_instance_set_remote_address(p->rtp, sa);
         if (debug) {
            ast_verbose("Peer audio RTP is at port %s\n",
                   ast_sockaddr_stringify(sa));
         }

         ast_rtp_codecs_payloads_copy(&newaudiortp, ast_rtp_instance_get_codecs(p->rtp), p->rtp);
         /* Ensure RTCP is enabled since it may be inactive
            if we're coming back from a T.38 session */
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
         /* Ensure audio RTCP reads are enabled */
         if (p->owner) {
            ast_channel_set_fd(p->owner, 1, ast_rtp_instance_fd(p->rtp, 1));
         }

         if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
            ast_clear_flag(&p->flags[0], SIP_DTMF);
            if (newnoncodeccapability & AST_RTP_DTMF) {
               /* XXX Would it be reasonable to drop the DSP at this point? XXX */
               ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
               /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
               ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, 1);
               ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF_COMPENSATE, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
            } else {
               ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
            }
         }
      } else if (udptlportno > 0) {
         if (debug)
            ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.\n");
         /* Prevent audio RTCP reads */
         if (p->owner) {
            ast_channel_set_fd(p->owner, 1, -1);
         }
         /* Silence RTCP while audio RTP is inactive */
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
      } else {
         ast_rtp_instance_stop(p->rtp);
         if (debug)
            ast_verbose("Peer doesn't provide audio\n");
      }
   }

   /* Setup video address and port */
   if (p->vrtp) {
      if (vsa && vportno > 0) {
         start_ice(p->vrtp);
         ast_sockaddr_set_port(vsa, vportno);
         ast_rtp_instance_set_remote_address(p->vrtp, vsa);
         if (debug) {
            ast_verbose("Peer video RTP is at port %s\n",
                   ast_sockaddr_stringify(vsa));
         }
         ast_rtp_codecs_payloads_copy(&newvideortp, ast_rtp_instance_get_codecs(p->vrtp), p->vrtp);
      } else {
         ast_rtp_instance_stop(p->vrtp);
         if (debug)
            ast_verbose("Peer doesn't provide video\n");
      }
   }

   /* Setup text address and port */
   if (p->trtp) {
      if (tsa && tportno > 0) {
         start_ice(p->trtp);
         ast_sockaddr_set_port(tsa, tportno);
         ast_rtp_instance_set_remote_address(p->trtp, tsa);
         if (debug) {
            ast_verbose("Peer T.140 RTP is at port %s\n",
                   ast_sockaddr_stringify(tsa));
         }
         if (ast_format_cap_iscompatible(p->jointcaps, ast_format_set(&tmp_fmt, AST_FORMAT_T140RED, 0))) {
            p->red = 1;
            ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
         } else {
            p->red = 0;
         }
         ast_rtp_codecs_payloads_copy(&newtextrtp, ast_rtp_instance_get_codecs(p->trtp), p->trtp);
      } else {
         ast_rtp_instance_stop(p->trtp);
         if (debug)
            ast_verbose("Peer doesn't provide T.140\n");
      }
   }

   /* Setup image address and port */
   if (p->udptl) {
      if (isa && udptlportno > 0) {
         if (ast_test_flag(&p->flags[1], SIP_PAGE2_SYMMETRICRTP) && ast_test_flag(&p->flags[1], SIP_PAGE2_UDPTL_DESTINATION)) {
            ast_rtp_instance_get_remote_address(p->rtp, isa);
            if (!ast_sockaddr_isnull(isa) && debug) {
               ast_debug(1, "Peer T.38 UDPTL is set behind NAT and with destination, destination address now %s\n", ast_sockaddr_stringify(isa));
            }
         }
         ast_sockaddr_set_port(isa, udptlportno);
         ast_udptl_set_peer(p->udptl, isa);
         if (debug)
            ast_debug(1,"Peer T.38 UDPTL is at port %s\n", ast_sockaddr_stringify(isa));

         /* verify the far max ifp can be calculated. this requires far max datagram to be set. */
         if (!ast_udptl_get_far_max_datagram(p->udptl)) {
            /* setting to zero will force a default if none was provided by the SDP */
            ast_udptl_set_far_max_datagram(p->udptl, 0);
         }

         /* Remote party offers T38, we need to update state */
         if ((t38action == SDP_T38_ACCEPT) &&
             (p->t38.state == T38_LOCAL_REINVITE)) {
            change_t38_state(p, T38_ENABLED);
         } else if ((t38action == SDP_T38_INITIATE) &&
               p->owner && p->lastinvite) {
            change_t38_state(p, T38_PEER_REINVITE); /* T38 Offered in re-invite from remote party */
            /* If fax detection is enabled then send us off to the fax extension */
            if (ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_T38)) {
               ast_channel_lock(p->owner);
               if (strcmp(ast_channel_exten(p->owner), "fax")) {
                  const char *target_context = S_OR(ast_channel_macrocontext(p->owner), ast_channel_context(p->owner));
                  ast_channel_unlock(p->owner);
                  if (ast_exists_extension(p->owner, target_context, "fax", 1,
                     S_COR(ast_channel_caller(p->owner)->id.number.valid, ast_channel_caller(p->owner)->id.number.str, NULL))) {
                     ast_verb(2, "Redirecting '%s' to fax extension due to peer T.38 re-INVITE\n", ast_channel_name(p->owner));
                     pbx_builtin_setvar_helper(p->owner, "FAXEXTEN", ast_channel_exten(p->owner));
                     if (ast_async_goto(p->owner, target_context, "fax", 1)) {
                        ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(p->owner), target_context);
                     }
                  } else {
                     ast_log(LOG_NOTICE, "T.38 re-INVITE detected but no fax extension\n");
                  }
               } else {
                  ast_channel_unlock(p->owner);
               }
            }
         }
      } else {
         change_t38_state(p, T38_DISABLED);
         ast_udptl_stop(p->udptl);
         if (debug)
            ast_debug(1, "Peer doesn't provide T.38 UDPTL\n");
      }
   }

   if ((portno == -1) && (p->t38.state != T38_DISABLED) && (p->t38.state != T38_REJECTED)) {
      ast_debug(3, "Have T.38 but no audio, accepting offer anyway\n");
      res = 0;
      goto process_sdp_cleanup;
   }

   /* Ok, we're going with this offer */
   ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcaps));

   if (!p->owner) { /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
      res = 0;
      goto process_sdp_cleanup;
   }

   ast_debug(4, "We have an owner, now see if we need to change this call\n");
   if (ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO)) {
      if (debug) {
         char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
         ast_debug(1, "Setting native formats after processing SDP. peer joint formats %s, old nativeformats %s\n",
            ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcaps),
            ast_getformatname_multiple(s2, SIPBUFSIZE, ast_channel_nativeformats(p->owner)));
      }

      ast_codec_choose(&p->prefs, p->jointcaps, 1, &tmp_fmt);

      ast_format_cap_set(ast_channel_nativeformats(p->owner), &tmp_fmt);
      ast_format_cap_joint_append(p->caps, vpeercapability, ast_channel_nativeformats(p->owner));
      ast_format_cap_joint_append(p->caps, tpeercapability, ast_channel_nativeformats(p->owner));

      ast_set_read_format(p->owner, ast_channel_readformat(p->owner));
      ast_set_write_format(p->owner, ast_channel_writeformat(p->owner));
   }

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) && (!ast_sockaddr_isnull(sa) || !ast_sockaddr_isnull(vsa) || !ast_sockaddr_isnull(tsa) || !ast_sockaddr_isnull(isa)) && (!sendonly || sendonly == -1)) {
      ast_queue_control(p->owner, AST_CONTROL_UNHOLD);
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      change_hold_state(p, req, FALSE, sendonly);
   } else if ((sockaddr_is_null_or_any(sa) && sockaddr_is_null_or_any(vsa) && sockaddr_is_null_or_any(tsa) && sockaddr_is_null_or_any(isa)) || (sendonly && sendonly != -1)) {
      ast_queue_control_data(p->owner, AST_CONTROL_HOLD,
                   S_OR(p->mohsuggest, NULL),
                   !ast_strlen_zero(p->mohsuggest) ? strlen(p->mohsuggest) + 1 : 0);
      if (sendonly)
         ast_rtp_instance_stop(p->rtp);
      /* RTCP needs to go ahead, even if we're on hold!!! */
      /* Activate a re-invite */
      ast_queue_frame(p->owner, &ast_null_frame);
      change_hold_state(p, req, TRUE, sendonly);
   }

process_sdp_cleanup:
   if (res) {
      offered_media_list_destroy(p);
   }
   ast_rtp_codecs_payloads_destroy(&newtextrtp);
   ast_rtp_codecs_payloads_destroy(&newvideortp);
   ast_rtp_codecs_payloads_destroy(&newaudiortp);
   ast_format_cap_destroy(peercapability);
   ast_format_cap_destroy(vpeercapability);
   ast_format_cap_destroy(tpeercapability);
   ast_format_cap_destroy(newjointcapability);
   ast_format_cap_destroy(newpeercapability);
   return res;
}

static int process_sdp_a_audio	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newaudiortp,
		int *	last_rtpmap_codec
	)		[static]

Definition at line 11067 of file chan_sip.c.

References ast_codec_pref_setsize(), ast_debug, AST_FORMAT_G719, ast_format_sdp_parse(), AST_FORMAT_SIREN14, AST_FORMAT_SIREN7, ast_getformatname(), ast_log(), ast_rtp_codecs_get_payload_format(), ast_rtp_codecs_packetization_set(), ast_rtp_codecs_payload_lookup(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_payloads_unset(), ast_rtp_instance_get_codecs(), AST_RTP_MAX_PT, AST_RTP_OPT_G726_NONSTANDARD, ast_test_flag, ast_verbose(), ast_rtp_payload_type::asterisk_format, debug, FALSE, format, ast_rtp_payload_type::format, ast_format::id, LOG_WARNING, ast_rtp_codecs::pref, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   char fmtp_string[256];
   unsigned int sample_rate;
   int debug = sip_debug_test_pvt(p);

   if (!strncasecmp(a, "ptime", 5)) {
      char *tmp = strrchr(a, ':');
      long int framing = 0;
      if (tmp) {
         tmp++;
         framing = strtol(tmp, NULL, 10);
         if (framing == LONG_MIN || framing == LONG_MAX) {
            framing = 0;
            ast_debug(1, "Can't read framing from SDP: %s\n", a);
         }
      }
      if (framing && p->autoframing) {
         struct ast_codec_pref *pref = &ast_rtp_instance_get_codecs(p->rtp)->pref;
         int codec_n;
         for (codec_n = 0; codec_n < AST_RTP_MAX_PT; codec_n++) {
            struct ast_rtp_payload_type format = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(p->rtp), codec_n);
            if (!format.asterisk_format)  /* non-codec or not found */
               continue;
            ast_debug(1, "Setting framing for %s to %ld\n", ast_getformatname(&format.format), framing);
            ast_codec_pref_setsize(pref, &format.format, framing);
         }
         ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, pref);
      }
      found = TRUE;
   } else if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
             ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate))) {
            if (debug)
               ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
            //found_rtpmap_codecs[last_rtpmap_codec] = codec;
            (*last_rtpmap_codec)++;
            found = TRUE;
         } else {
            ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
            if (debug)
               ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   } else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
      struct ast_format *format;

      if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
         unsigned int bit_rate;

         if (!ast_format_sdp_parse(format, fmtp_string)) {
            found = TRUE;
         } else {
            ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
         }

         switch ((int) format->id) {
         case AST_FORMAT_SIREN7:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 32000) {
                  ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
            break;
         case AST_FORMAT_SIREN14:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 48000) {
                  ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
            break;
         case AST_FORMAT_G719:
            if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
               if (bit_rate != 64000) {
                  ast_log(LOG_WARNING, "Got G.719 offer at %d bps, but only 64000 bps supported; ignoring.\n", bit_rate);
                  ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
               } else {
                  found = TRUE;
               }
            }
            break;
         }
      }
   }

   return found;
}

static int process_sdp_a_dtls	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_instance *	instance
	)		[static]

Definition at line 11017 of file chan_sip.c.

References ast_log(), AST_RTP_DTLS_HASH_SHA1, AST_RTP_DTLS_SETUP_ACTIVE, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_HOLDCONN, AST_RTP_DTLS_SETUP_PASSIVE, ast_rtp_instance_get_dtls(), FALSE, LOG_WARNING, ast_rtp_engine_dtls::reset, ast_rtp_engine_dtls::set_fingerprint, ast_rtp_engine_dtls::set_setup, TRUE, and value.

Referenced by process_sdp().

{
   struct ast_rtp_engine_dtls *dtls;
   int found = FALSE;
   char value[256], hash[6];

   if (!instance || !p->dtls_cfg.enabled || !(dtls = ast_rtp_instance_get_dtls(instance))) {
      return found;
   }

   if (sscanf(a, "setup: %255s", value) == 1) {
      found = TRUE;

      if (!strcasecmp(value, "active")) {
         dtls->set_setup(instance, AST_RTP_DTLS_SETUP_ACTIVE);
      } else if (!strcasecmp(value, "passive")) {
         dtls->set_setup(instance, AST_RTP_DTLS_SETUP_PASSIVE);
      } else if (!strcasecmp(value, "actpass")) {
         dtls->set_setup(instance, AST_RTP_DTLS_SETUP_ACTPASS);
      } else if (!strcasecmp(value, "holdconn")) {
         dtls->set_setup(instance, AST_RTP_DTLS_SETUP_HOLDCONN);
      } else {
         ast_log(LOG_WARNING, "Unsupported setup attribute value '%s' received on dialog '%s'\n",
            value, p->callid);
      }
   } else if (sscanf(a, "connection: %255s", value) == 1) {
      found = TRUE;

      if (!strcasecmp(value, "new")) {
         dtls->reset(instance);
      } else if (!strcasecmp(value, "existing")) {
         /* Since they want to just use what already exists we go on as if nothing happened */
      } else {
         ast_log(LOG_WARNING, "Unsupported connection attribute value '%s' received on dialog '%s'\n",
            value, p->callid);
      }
   } else if (sscanf(a, "fingerprint: %5s %255s", hash, value) == 2) {
      found = TRUE;

      if (!strcasecmp(hash, "sha-1")) {
         dtls->set_fingerprint(instance, AST_RTP_DTLS_HASH_SHA1, value);
      } else {
         ast_log(LOG_WARNING, "Unsupported fingerprint hash type '%s' received on dialog '%s'\n",
            hash, p->callid);
      }
   }

   return found;
}

static int process_sdp_a_ice	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_instance *	instance
	)		[static]

Definition at line 10962 of file chan_sip.c.

References ast_rtp_engine_ice::add_remote_candidate, ast_rtp_engine_ice_candidate::address, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, ast_rtp_instance_get_ice(), ast_sockaddr_parse(), ast_sockaddr_set_port, ast_strlen_zero(), FALSE, ast_rtp_engine_ice_candidate::foundation, ast_rtp_engine_ice::ice_lite, ast_rtp_engine_ice_candidate::id, PARSE_PORT_FORBID, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_rtp_engine_ice::set_authentication, ast_rtp_engine_ice_candidate::transport, TRUE, and ast_rtp_engine_ice_candidate::type.

Referenced by process_sdp().

{
   struct ast_rtp_engine_ice *ice;
   int found = FALSE;
   char ufrag[256], pwd[256], foundation[32], transport[4], address[46], cand_type[6], relay_address[46] = "";
   struct ast_rtp_engine_ice_candidate candidate = { 0, };
   int port, relay_port = 0;

   if (!instance || !(ice = ast_rtp_instance_get_ice(instance))) {
      return found;
   }

   if (sscanf(a, "ice-ufrag: %255s", ufrag) == 1) {
      ice->set_authentication(instance, ufrag, NULL);
      found = TRUE;
   } else if (sscanf(a, "ice-pwd: %255s", pwd) == 1) {
      ice->set_authentication(instance, NULL, pwd);
      found = TRUE;
   } else if (sscanf(a, "candidate: %31s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, &candidate.priority,
           address, &port, cand_type, relay_address, &relay_port) >= 7) {
      candidate.foundation = foundation;
      candidate.transport = transport;

      ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID);
      ast_sockaddr_set_port(&candidate.address, port);

      if (!strcasecmp(cand_type, "host")) {
         candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
      } else if (!strcasecmp(cand_type, "srflx")) {
         candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
      } else if (!strcasecmp(cand_type, "relay")) {
         candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
      } else {
         return found;
      }

      if (!ast_strlen_zero(relay_address)) {
         ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID);
      }

      if (relay_port) {
         ast_sockaddr_set_port(&candidate.relay_address, relay_port);
      }

      ice->add_remote_candidate(instance, &candidate);

      found = TRUE;
   } else if (!strcasecmp(a, "ice-lite")) {
      ice->ice_lite(instance);
      found = TRUE;
   }

   return found;
}

static int process_sdp_a_image	(	const char *	a,
		struct sip_pvt *	p
	)		[static]

Definition at line 11263 of file chan_sip.c.

References ast_debug, AST_T38_RATE_12000, AST_T38_RATE_14400, AST_T38_RATE_2400, AST_T38_RATE_4800, AST_T38_RATE_7200, AST_T38_RATE_9600, AST_T38_RATE_MANAGEMENT_LOCAL_TCF, AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF, ast_udptl_set_error_correction_scheme(), ast_udptl_set_far_max_datagram(), FALSE, initialize_udptl(), TRUE, UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, and UDPTL_ERROR_CORRECTION_REDUNDANCY.

Referenced by process_sdp().

{
   int found = FALSE;
   char s[256];
   unsigned int x;
   char *attrib = ast_strdupa(a);
   char *pos;

   if (initialize_udptl(p)) {
      return found;
   }

   /* Due to a typo in an IANA registration of one of the T.38 attributes,
    * RFC5347 section 2.5.2 recommends that all T.38 attributes be parsed in
    * a case insensitive manner. Hence, the importance of proof reading (and
    * code reviews).
    */
   for (pos = attrib; *pos; ++pos) {
      *pos = tolower(*pos);
   }

   if ((sscanf(attrib, "t38faxmaxbuffer:%30u", &x) == 1)) {
      ast_debug(3, "MaxBufferSize:%d\n", x);
      found = TRUE;
   } else if ((sscanf(attrib, "t38maxbitrate:%30u", &x) == 1) || (sscanf(attrib, "t38faxmaxrate:%30u", &x) == 1)) {
      ast_debug(3, "T38MaxBitRate: %d\n", x);
      switch (x) {
      case 14400:
         p->t38.their_parms.rate = AST_T38_RATE_14400;
         break;
      case 12000:
         p->t38.their_parms.rate = AST_T38_RATE_12000;
         break;
      case 9600:
         p->t38.their_parms.rate = AST_T38_RATE_9600;
         break;
      case 7200:
         p->t38.their_parms.rate = AST_T38_RATE_7200;
         break;
      case 4800:
         p->t38.their_parms.rate = AST_T38_RATE_4800;
         break;
      case 2400:
         p->t38.their_parms.rate = AST_T38_RATE_2400;
         break;
      }
      found = TRUE;
   } else if ((sscanf(attrib, "t38faxversion:%30u", &x) == 1)) {
      ast_debug(3, "FaxVersion: %u\n", x);
      p->t38.their_parms.version = x;
      found = TRUE;
   } else if ((sscanf(attrib, "t38faxmaxdatagram:%30u", &x) == 1) || (sscanf(attrib, "t38maxdatagram:%30u", &x) == 1)) {
      /* override the supplied value if the configuration requests it */
      if (((signed int) p->t38_maxdatagram >= 0) && ((unsigned int) p->t38_maxdatagram > x)) {
         ast_debug(1, "Overriding T38FaxMaxDatagram '%d' with '%d'\n", x, p->t38_maxdatagram);
         x = p->t38_maxdatagram;
      }
      ast_debug(3, "FaxMaxDatagram: %u\n", x);
      ast_udptl_set_far_max_datagram(p->udptl, x);
      found = TRUE;
   } else if ((strncmp(attrib, "t38faxfillbitremoval", 20) == 0)) {
      if (sscanf(attrib, "t38faxfillbitremoval:%30u", &x) == 1) {
         ast_debug(3, "FillBitRemoval: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.fill_bit_removal = TRUE;
         }
      } else {
         ast_debug(3, "FillBitRemoval\n");
         p->t38.their_parms.fill_bit_removal = TRUE;
      }
      found = TRUE;
   } else if ((strncmp(attrib, "t38faxtranscodingmmr", 20) == 0)) {
      if (sscanf(attrib, "t38faxtranscodingmmr:%30u", &x) == 1) {
         ast_debug(3, "Transcoding MMR: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.transcoding_mmr = TRUE;
         }
      } else {
         ast_debug(3, "Transcoding MMR\n");
         p->t38.their_parms.transcoding_mmr = TRUE;
      }
      found = TRUE;
   } else if ((strncmp(attrib, "t38faxtranscodingjbig", 21) == 0)) {
      if (sscanf(attrib, "t38faxtranscodingjbig:%30u", &x) == 1) {
         ast_debug(3, "Transcoding JBIG: %d\n", x);
         if (x == 1) {
            p->t38.their_parms.transcoding_jbig = TRUE;
         }
      } else {
         ast_debug(3, "Transcoding JBIG\n");
         p->t38.their_parms.transcoding_jbig = TRUE;
      }
      found = TRUE;
   } else if ((sscanf(attrib, "t38faxratemanagement:%255s", s) == 1)) {
      ast_debug(3, "RateManagement: %s\n", s);
      if (!strcasecmp(s, "localTCF"))
         p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_LOCAL_TCF;
      else if (!strcasecmp(s, "transferredTCF"))
         p->t38.their_parms.rate_management = AST_T38_RATE_MANAGEMENT_TRANSFERRED_TCF;
      found = TRUE;
   } else if ((sscanf(attrib, "t38faxudpec:%255s", s) == 1)) {
      ast_debug(3, "UDP EC: %s\n", s);
      if (!strcasecmp(s, "t38UDPRedundancy")) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
      } else if (!strcasecmp(s, "t38UDPFEC")) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
      } else {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
      }
      found = TRUE;
   }

   return found;
}

static int process_sdp_a_sendonly	(	const char *	a,
		int *	sendonly
	)		[static]

Definition at line 10942 of file chan_sip.c.

References FALSE, and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;

   if (!strcasecmp(a, "sendonly")) {
      if (*sendonly == -1)
         *sendonly = 1;
      found = TRUE;
   } else if (!strcasecmp(a, "inactive")) {
      if (*sendonly == -1)
         *sendonly = 2;
      found = TRUE;
   }  else if (!strcasecmp(a, "sendrecv")) {
      if (*sendonly == -1)
         *sendonly = 0;
      found = TRUE;
   }
   return found;
}

static int process_sdp_a_text	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newtextrtp,
		char *	red_fmtp,
		int *	red_num_gen,
		int *	red_data_pt,
		int *	last_rtpmap_codec
	)		[static]

Definition at line 11214 of file chan_sip.c.

References AST_RED_MAX_GENERATION, ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_verbose(), debug, FALSE, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   unsigned int sample_rate;
   char *red_cp;
   int debug = sip_debug_test_pvt(p);

   if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         if (!strncasecmp(mimeSubtype, "T140", 4)) { /* Text */
            if (p->trtp) {
               /* ast_verbose("Adding t140 mimeSubtype to textrtp struct\n"); */
               ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
               found = TRUE;
            }
         } else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
            if (p->trtp) {
               ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
               sprintf(red_fmtp, "fmtp:%d ", codec);
               if (debug)
                  ast_verbose("RED submimetype has payload type: %d\n", codec);
               found = TRUE;
            }
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
      /* count numbers of generations in fmtp */
      red_cp = &red_fmtp[strlen(red_fmtp)];
      strncpy(red_fmtp, a, 100);

      sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
      red_cp = strtok(red_cp, "/");
      while (red_cp && (*red_num_gen)++ < AST_RED_MAX_GENERATION) {
         sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
         red_cp = strtok(NULL, "/");
      }
      red_cp = red_fmtp;
      found = TRUE;
   }

   return found;
}

static int process_sdp_a_video	(	const char *	a,
		struct sip_pvt *	p,
		struct ast_rtp_codecs *	newvideortp,
		int *	last_rtpmap_codec
	)		[static]

Definition at line 11169 of file chan_sip.c.

References ast_format_sdp_parse(), ast_rtp_codecs_get_payload_format(), ast_rtp_codecs_payloads_set_rtpmap_type_rate(), ast_rtp_codecs_payloads_unset(), ast_verbose(), debug, FALSE, format, sip_debug_test_pvt(), and TRUE.

Referenced by process_sdp().

{
   int found = FALSE;
   int codec;
   char mimeSubtype[128];
   unsigned int sample_rate;
   int debug = sip_debug_test_pvt(p);
   char fmtp_string[256];

   if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
      /* We have a rtpmap to handle */
      if (*last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
         /* Note: should really look at the '#chans' params too */
         if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) {
            if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
               if (debug)
                  ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
               //found_rtpmap_codecs[last_rtpmap_codec] = codec;
               (*last_rtpmap_codec)++;
               found = TRUE;
            } else {
               ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
               if (debug)
                  ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
            }
         }
      } else {
         if (debug)
            ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
      }
   } else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
      struct ast_format *format;

      if ((format = ast_rtp_codecs_get_payload_format(newvideortp, codec))) {
         if (!ast_format_sdp_parse(format, fmtp_string)) {
            found = TRUE;
         } else {
            ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
         }
      }
   }

   return found;
}

static int process_sdp_c	(	const char *	c,
		struct ast_sockaddr *	addr
	)		[static]

Definition at line 10915 of file chan_sip.c.

References ast_log(), ast_sockaddr_resolve_first_af(), FALSE, LOG_WARNING, and TRUE.

Referenced by process_sdp().

{
   char proto[4], host[258];
   int af;

   /* Check for Media-description-level-address */
   if (sscanf(c, "IN %3s %255s", proto, host) == 2) {
      if (!strcmp("IP4", proto)) {
         af = AF_INET;
      } else if (!strcmp("IP6", proto)) {
         af = AF_INET6;
      } else {
         ast_log(LOG_WARNING, "Unknown protocol '%s'.\n", proto);
         return FALSE;
      }
      if (ast_sockaddr_resolve_first_af(addr, host, 0, af)) {
         ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in c= line, '%s'\n", c);
         return FALSE;
      }
      return TRUE;
   } else {
      ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
      return FALSE;
   }
   return FALSE;
}

static int process_sdp_o	(	const char *	o,
		struct sip_pvt *	p
	)		[static]

Definition at line 10836 of file chan_sip.c.

References ast_debug, ast_log(), ast_strlen_zero(), ast_test_flag, FALSE, LOG_WARNING, and TRUE.

Referenced by process_sdp().

{
   char *o_copy;
   char *token;
   int64_t rua_version;

   /* Store the SDP version number of remote UA. This will allow us to
   distinguish between session modifications and session refreshes. If
   the remote UA does not send an incremented SDP version number in a
   subsequent RE-INVITE then that means its not changing media session.
   The RE-INVITE may have been sent to update connected party, remote
   target or to refresh the session (Session-Timers).  Asterisk must not
   change media session and increment its own version number in answer
   SDP in this case. */

   p->session_modify = TRUE;

   if (ast_strlen_zero(o)) {
      ast_log(LOG_WARNING, "SDP syntax error. SDP without an o= line\n");
      return FALSE;
   }

   o_copy = ast_strdupa(o);
   token = strsep(&o_copy, " ");  /* Skip username   */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line username\n");
      return FALSE;
   }
   token = strsep(&o_copy, " ");  /* Skip session-id */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line session-id\n");
      return FALSE;
   }
   token = strsep(&o_copy, " ");  /* Version         */
   if (!o_copy) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line\n");
      return FALSE;
   }
   if (!sscanf(token, "%30" SCNd64, &rua_version)) {
      ast_log(LOG_WARNING, "SDP syntax error in o= line version\n");
      return FALSE;
   }

   /* we need to check the SDP version number the other end sent us;
    * our rules for deciding what to accept are a bit complex.
    *
    * 1) if 'ignoresdpversion' has been set for this dialog, then
    *    we will just accept whatever they sent and assume it is
    *    a modification of the session, even if it is not
    * 2) otherwise, if this is the first SDP we've seen from them
    *    we accept it
    * 3) otherwise, if the new SDP version number is higher than the
    *    old one, we accept it
    * 4) otherwise, if this SDP is in response to us requesting a switch
    *    to T.38, we accept the SDP, but also generate a warning message
    *    that this peer should have the 'ignoresdpversion' option set,
    *    because it is not following the SDP offer/answer RFC; if we did
    *    not request a switch to T.38, then we stop parsing the SDP, as it
    *    has not changed from the previous version
    */

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_IGNORESDPVERSION) ||
       (p->sessionversion_remote < 0) ||
       (p->sessionversion_remote < rua_version)) {
      p->sessionversion_remote = rua_version;
   } else {
      if (p->t38.state == T38_LOCAL_REINVITE) {
         p->sessionversion_remote = rua_version;
         ast_log(LOG_WARNING, "Call %s responded to our T.38 reinvite without changing SDP version; 'ignoresdpversion' should be set for this peer.\n", p->callid);
      } else {
         p->session_modify = FALSE;
         ast_debug(2, "Call %s responded to our reinvite without changing SDP version; ignoring SDP.\n", p->callid);
         return FALSE;
      }
   }

   return TRUE;
}

static int process_via	(	struct sip_pvt *	p,
		const struct sip_request *	req
	)		[static]

Process the Via header according to RFC 3261 section 18.2.2.

Parameters:

p	a sip_pvt structure that will be modified according to the received header
req	a sip request with a Via header to process

This function will update the destination of the response according to the Via header in the request and RFC 3261 section 18.2.2. We do not have a transport layer so we ignore certain values like the 'received' param (we set the destination address to the addres the request came from in the respprep() function).

Return values:

-1	error
0	success

Definition at line 8802 of file chan_sip.c.

References ast_log(), ast_sockaddr_is_ipv4_multicast(), ast_sockaddr_resolve_first_transport(), ast_sockaddr_set_port, free_via(), LOG_ERROR, LOG_WARNING, PARSE_PORT_FORBID, parse_via(), and sip_get_header().

Referenced by respprep().

{
   struct sip_via *via = parse_via(sip_get_header(req, "Via"));

   if (!via) {
      ast_log(LOG_ERROR, "error processing via header\n");
      return -1;
   }

   if (via->maddr) {
      if (ast_sockaddr_resolve_first_transport(&p->sa, via->maddr, PARSE_PORT_FORBID, p->socket.type)) {
         ast_log(LOG_WARNING, "Can't find address for maddr '%s'\n", via->maddr);
         ast_log(LOG_ERROR, "error processing via header\n");
         free_via(via);
         return -1;
      }

      if (ast_sockaddr_is_ipv4_multicast(&p->sa)) {
         setsockopt(sipsock, IPPROTO_IP, IP_MULTICAST_TTL, &via->ttl, sizeof(via->ttl));
      }
   }

   ast_sockaddr_set_port(&p->sa, via->port ? via->port : STANDARD_SIP_PORT);

   free_via(via);
   return 0;
}

static struct sip_proxy* proxy_from_config	(	const char *	proxy,
		int	sipconf_lineno,
		struct sip_proxy *	dest
	)		[static, read]

Parse proxy string and return an ao2_alloc'd proxy. If dest is non-NULL, no allocation is performed and dest is used instead. On error NULL is returned.

Definition at line 3561 of file chan_sip.c.

References ao2_alloc, ao2_ref, ast_copy_string(), ast_log(), ast_skip_blanks(), ast_strlen_zero(), FALSE, LOG_WARNING, name, proxy_update(), and sip_parse_host().

Referenced by build_peer(), reload_config(), and sip_request_call().

{
   char *mutable_proxy, *sep, *name;
   int allocated = 0;

   if (!dest) {
      dest = ao2_alloc(sizeof(struct sip_proxy), NULL);
      if (!dest) {
         ast_log(LOG_WARNING, "Unable to allocate config storage for proxy\n");
         return NULL;
      }
      allocated = 1;
   }

   /* Format is: [transport://]name[:port][,force] */
   mutable_proxy = ast_skip_blanks(ast_strdupa(proxy));
   sep = strchr(mutable_proxy, ',');
   if (sep) {
      *sep++ = '\0';
      dest->force = !strncasecmp(ast_skip_blanks(sep), "force", 5);
   } else {
      dest->force = FALSE;
   }

   sip_parse_host(mutable_proxy, sipconf_lineno, &name, &dest->port, &dest->transport);

   /* Check that there is a name at all */
   if (ast_strlen_zero(name)) {
      if (allocated) {
         ao2_ref(dest, -1);
      } else {
         dest->name[0] = '\0';
      }
      return NULL;
   }
   ast_copy_string(dest->name, name, sizeof(dest->name));

   /* Resolve host immediately */
   proxy_update(dest);

   return dest;
}

static int proxy_update

(

struct sip_proxy *

proxy

)

[static]

Resolve DNS srv name or host name in a sip_proxy structure

Definition at line 3537 of file chan_sip.c.

References ast_get_ip_or_srv(), ast_log(), ast_sockaddr_parse(), ast_sockaddr_set_port, FALSE, get_address_family_filter(), LOG_WARNING, sip_cfg, and TRUE.

Referenced by proxy_from_config().

{
   /* if it's actually an IP address and not a name,
           there's no need for a managed lookup */
   if (!ast_sockaddr_parse(&proxy->ip, proxy->name, 0)) {
      /* Ok, not an IP address, then let's check if it's a domain or host */
      /* XXX Todo - if we have proxy port, don't do SRV */
      proxy->ip.ss.ss_family = get_address_family_filter(SIP_TRANSPORT_UDP); /* Filter address family */
      if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
            ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
            return FALSE;
      }

   }

   ast_sockaddr_set_port(&proxy->ip, proxy->port);

   proxy->last_dnsupdate = time(NULL);
   return TRUE;
}

static int publish_expire

(

const void *

data

)

[static]

Definition at line 1079 of file chan_sip.c.

References ao2_ref, ao2_unlink, ast_assert, event_state_compositor::compositor, and get_esc().

Referenced by create_esc_entry(), handle_sip_publish_modify(), and handle_sip_publish_refresh().

{
   struct sip_esc_entry *esc_entry = (struct sip_esc_entry *) data;
   struct event_state_compositor *esc = get_esc(esc_entry->event);

   ast_assert(esc != NULL);

   ao2_unlink(esc->compositor, esc_entry);
   ao2_ref(esc_entry, -1);
   return 0;
}

static void pvt_set_needdestroy	(	struct sip_pvt *	pvt,
		const char *	reason
	)		[inline, static]

Definition at line 3499 of file chan_sip.c.

References ao2_t_link, and append_history.

Referenced by __sip_autodestruct(), forked_invite_init(), handle_incoming(), handle_request_publish(), handle_request_refer(), handle_request_subscribe(), handle_response(), handle_response_invite(), handle_response_message(), handle_response_notify(), handle_response_peerpoke(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), retrans_pkt(), sip_hangup(), and sip_reg_timeout().

{
   if (pvt->final_destruction_scheduled) {
      return; /* This is already scheduled for final destruction, let the scheduler take care of it. */
   }
   append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
   if (!pvt->needdestroy) {
      pvt->needdestroy = 1;
      ao2_t_link(dialogs_needdestroy, pvt, "link pvt into dialogs_needdestroy container");
   }
}

static int read_raw_content_length

(

const char *

message

)

[static]

Get the content length from an unparsed SIP message.

Parameters:

message

The unparsed SIP message headers

Returns:

The value of the Content-Length header or -1 if message is invalid

Definition at line 2841 of file chan_sip.c.

References ast_free, ast_str_buffer(), ast_str_create(), ast_str_set(), lws2sws(), and sip_cfg.

Referenced by check_message_integrity().

{
   char *content_length_str;
   int content_length = -1;

   struct ast_str *msg_copy;
   char *msg;

   /* Using a ast_str because lws2sws takes one of those */
   if (!(msg_copy = ast_str_create(strlen(message) + 1))) {
      return -1;
   }
   ast_str_set(&msg_copy, 0, "%s", message);

   if (sip_cfg.pedanticsipchecking) {
      lws2sws(msg_copy);
   }

   msg = ast_str_buffer(msg_copy);

   /* Let's find a Content-Length header */
   if ((content_length_str = strcasestr(msg, "\nContent-Length:"))) {
      content_length_str += sizeof("\nContent-Length:") - 1;
   } else if ((content_length_str = strcasestr(msg, "\nl:"))) {
      content_length_str += sizeof("\nl:") - 1;
   } else {
      /* RFC 3261 18.3
       * "In the case of stream-oriented transports such as TCP, the Content-
       *  Length header field indicates the size of the body.  The Content-
       *  Length header field MUST be used with stream oriented transports."
       */
      goto done;
   }

   /* Double-check that this is a complete header */
   if (!strchr(content_length_str, '\n')) {
      goto done;
   }

   if (sscanf(content_length_str, "%30d", &content_length) != 1) {
      content_length = -1;
   }

done:
   ast_free(msg_copy);
   return content_length;
}

static struct sip_peer * realtime_peer	(	const char *	newpeername,
		struct ast_sockaddr *	addr,
		char *	callbackexten,
		int	devstate_only,
		int	which_objects
	)		[static, read]

realtime_peer: Get peer from realtime storage Checks the "sippeers" realtime family from extconfig.conf Checks the "sipregs" realtime family from extconfig.conf if it's configured. This returns a pointer to a peer and because we use build_peer, we can rest assured that the refcount is bumped.

Note:

This is never called with both newpeername and addr at the same time. If you do, be prepared to get a peer with a different name than newpeername.

Definition at line 5552 of file chan_sip.c.

References ao2_t_link, ast_check_realtime(), ast_copy_flags, ast_copy_string(), ast_debug, AST_SCHED_REPLACE_UNREF, ast_sockaddr_isnull(), ast_sockaddr_stringify_addr(), ast_test_flag, ast_variables_destroy(), build_peer(), cleanup(), expire_register(), ipaddr, ast_variable::name, ast_variable::next, realtime_peer_by_addr(), realtime_peer_by_name(), sip_cfg, sip_ref_peer(), sip_unref_peer(), TRUE, ast_variable::value, and var.

Referenced by sip_find_peer_full().

{
   struct sip_peer *peer = NULL;
   struct ast_variable *var = NULL;
   struct ast_variable *varregs = NULL;
   char ipaddr[INET6_ADDRSTRLEN];
   int realtimeregs = ast_check_realtime("sipregs");

   if (addr) {
      ast_copy_string(ipaddr, ast_sockaddr_stringify_addr(addr), sizeof(ipaddr));
   } else {
      ipaddr[0] = '\0';
   }

   if (newpeername && realtime_peer_by_name(&newpeername, addr, ipaddr, &var, realtimeregs ? &varregs : NULL)) {
      ;
   } else if (addr && realtime_peer_by_addr(&newpeername, addr, ipaddr, callbackexten, &var, realtimeregs ? &varregs : NULL)) {
      ;
   } else {
      return NULL;
   }

   /* If we're looking for users, don't return peers (although this check
    * should probably be done in realtime_peer_by_* instead...) */
   if (which_objects == FINDUSERS) {
      struct ast_variable *tmp;
      for (tmp = var; tmp; tmp = tmp->next) {
         if (!strcasecmp(tmp->name, "type") && (!strcasecmp(tmp->value, "peer"))) {
            goto cleanup;
         }
      }
   }

   /* Peer found in realtime, now build it in memory */
   peer = build_peer(newpeername, var, varregs, TRUE, devstate_only);
   if (!peer) {
      goto cleanup;
   }

   /* Previous versions of Asterisk did not require the type field to be
    * set for real time peers.  This statement preserves that behavior. */
   if  (peer->type == 0) {
      if (which_objects == FINDUSERS) {
         peer->type = SIP_TYPE_USER;
      } else if (which_objects == FINDPEERS) {
         peer->type = SIP_TYPE_PEER;
      } else {
         peer->type = SIP_TYPE_PEER | SIP_TYPE_USER;
      }
   }

   ast_debug(3, "-REALTIME- loading peer from database to memory. Name: %s. Peer objects: %d\n", peer->name, rpeerobjs);

   if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && !devstate_only) {
      /* Cache peer */
      ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_RTAUTOCLEAR|SIP_PAGE2_RTCACHEFRIENDS);
      if (ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR)) {
         AST_SCHED_REPLACE_UNREF(peer->expire, sched, sip_cfg.rtautoclear * 1000, expire_register, peer,
               sip_unref_peer(_data, "remove registration ref"),
               sip_unref_peer(peer, "remove registration ref"),
               sip_ref_peer(peer, "add registration ref"));
      }
      ao2_t_link(peers, peer, "link peer into peers table");
      if (!ast_sockaddr_isnull(&peer->addr)) {
         ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
      }
   }
   peer->is_realtime = 1;

cleanup:
   ast_variables_destroy(var);
   ast_variables_destroy(varregs);
   return peer;
}

static int realtime_peer_by_addr	(	const char **	name,
		struct ast_sockaddr *	addr,
		const char *	ipaddr,
		const char *	callbackexten,
		struct ast_variable **	var,
		struct ast_variable **	varregs
	)		[static]

Definition at line 5448 of file chan_sip.c.

References ast_copy_string(), ast_load_realtime(), ast_log(), ast_sockaddr_stringify_port(), ast_strlen_zero(), ast_variables_destroy(), get_insecure_variable_from_sippeers(), get_insecure_variable_from_sipregs(), get_name_from_variable(), LOG_WARNING, realtime_peer_get_sippeer_helper(), and SENTINEL.

Referenced by realtime_peer().

{
   char portstring[6]; /* up to 5 digits plus null terminator */
   ast_copy_string(portstring, ast_sockaddr_stringify_port(addr), sizeof(portstring));

   /* We're not finding this peer by this name anymore. Reset it. */
   *name = NULL;

   /* First check for fixed IP hosts with matching callbackextensions, if specified */
   if (!ast_strlen_zero(callbackexten) && (*var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, "callbackextension", callbackexten, SENTINEL))) {
      ;
   /* Check for fixed IP hosts */
   } else if ((*var = ast_load_realtime("sippeers", "host", ipaddr, "port", portstring, SENTINEL))) {
      ;
   /* Check for registered hosts (in sipregs) */
   } else if (varregs && (*varregs = ast_load_realtime("sipregs", "ipaddr", ipaddr, "port", portstring, SENTINEL)) &&
         (*var = realtime_peer_get_sippeer_helper(name, varregs))) {
      ;
   /* Check for registered hosts (in sippeers) */
   } else if (!varregs && (*var = ast_load_realtime("sippeers", "ipaddr", ipaddr, "port", portstring, SENTINEL))) {
      ;
   /* We couldn't match on ipaddress and port, so we need to check if port is insecure */
   } else if ((*var = get_insecure_variable_from_sippeers("host", ipaddr))) {
      ;
   /* Same as above, but try the IP address field (in sipregs)
    * Observe that it fetches the name/var at the same time, without the
    * realtime_peer_get_sippeer_helper. Also note that it is quite inefficient.
    * Avoid sipregs if possible. */
   } else if (varregs && (*varregs = get_insecure_variable_from_sipregs("ipaddr", ipaddr, var))) {
      ;
   /* Same as above, but try the IP address field (in sippeers) */
   } else if (!varregs && (*var = get_insecure_variable_from_sippeers("ipaddr", ipaddr))) {
      ;
   }

   /* Nothing found? */
   if (!*var) {
      return 0;
   }

   /* Check peer name. It must not be empty. There may exist a
    * different match that does have a name, but it's too late for
    * that now. */
   if (!*name && !(*name = get_name_from_variable(*var))) {
      ast_log(LOG_WARNING, "Found peer for IP %s but it has no name\n", ipaddr);
      ast_variables_destroy(*var);
      *var = NULL;
      if (varregs && *varregs) {
         ast_variables_destroy(*varregs);
         *varregs = NULL;
      }
      return 0;
   }

   /* Make sure varregs is populated if var is. The inverse,
    * ensuring that var is set when varregs is, is taken
    * care of by realtime_peer_get_sippeer_helper(). */
   if (varregs && !*varregs) {
      *varregs = ast_load_realtime("sipregs", "name", *name, SENTINEL);
   }
   return 1;
}

static int realtime_peer_by_name	(	const char *const *	name,
		struct ast_sockaddr *	addr,
		const char *	ipaddr,
		struct ast_variable **	var,
		struct ast_variable **	varregs
	)		[static]

Note:

If this one loaded something, then we need to ensure that the host field matched. The only reason why we can't have this as a criteria is because we only have the IP address and the host field might be set as a name (and the reverse PTR might not match).

Definition at line 5376 of file chan_sip.c.

References ast_free, ast_load_realtime(), ast_sockaddr_cmp(), ast_sockaddr_resolve(), ast_variables_destroy(), get_address_family_filter(), ast_variable::name, ast_variable::next, PARSE_PORT_FORBID, SENTINEL, and ast_variable::value.

Referenced by realtime_peer().

{
   /* Peer by name and host=dynamic */
   if ((*var = ast_load_realtime("sippeers", "name", *name, "host", "dynamic", SENTINEL))) {
      ;
   /* Peer by name and host=IP */
   } else if (addr && !(*var = ast_load_realtime("sippeers", "name", *name, "host", ipaddr, SENTINEL))) {
      ;
   /* Peer by name and host=HOSTNAME */
   } else if ((*var = ast_load_realtime("sippeers", "name", *name, SENTINEL))) {
      /*!\note
       * If this one loaded something, then we need to ensure that the host
       * field matched.  The only reason why we can't have this as a criteria
       * is because we only have the IP address and the host field might be
       * set as a name (and the reverse PTR might not match).
       */
      if (addr) {
         struct ast_variable *tmp;
         for (tmp = *var; tmp; tmp = tmp->next) {
            if (!strcasecmp(tmp->name, "host")) {
               struct ast_sockaddr *addrs = NULL;

               if (ast_sockaddr_resolve(&addrs,
                         tmp->value,
                         PARSE_PORT_FORBID,
                         get_address_family_filter(SIP_TRANSPORT_UDP)) <= 0 ||
                         ast_sockaddr_cmp(&addrs[0], addr)) {
                  /* No match */
                  ast_variables_destroy(*var);
                  *var = NULL;
               }
               ast_free(addrs);
               break;
            }
         }
      }
   }

   /* Did we find anything? */
   if (*var) {
      if (varregs) {
         *varregs = ast_load_realtime("sipregs", "name", *name, SENTINEL);
      }
      return 1;
   }
   return 0;
}

static struct ast_variable* realtime_peer_get_sippeer_helper	(	const char **	name,
		struct ast_variable **	varregs
	)		[static, read]

Definition at line 5430 of file chan_sip.c.

References ast_load_realtime(), ast_log(), ast_variables_destroy(), get_name_from_variable(), LOG_WARNING, name, SENTINEL, and var.

Referenced by realtime_peer_by_addr().

                                                                                                               {
   struct ast_variable *var = NULL;
   const char *old_name = *name;
   *name = get_name_from_variable(*varregs);
   if (!*name || !(var = ast_load_realtime("sippeers", "name", *name, SENTINEL))) {
      if (!*name) {
         ast_log(LOG_WARNING, "Found sipreg but it has no name\n");
      }
      ast_variables_destroy(*varregs);
      *varregs = NULL;
      *name = old_name;
   }
   return var;
}

static void realtime_update_peer	(	const char *	peername,
		struct ast_sockaddr *	addr,
		const char *	username,
		const char *	fullcontact,
		const char *	useragent,
		int	expirey,
		unsigned short	deprecated_username,
		int	lastms
	)		[static]

Update peer object in realtime storage If the Asterisk system name is set in asterisk.conf, we will use that name and store that in the "regserver" field in the sippeers table to facilitate multi-server setups.

Definition at line 5077 of file chan_sip.c.

References ast_check_realtime(), ast_config_AST_SYSTEM_NAME, ast_copy_string(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), ast_strlen_zero(), ast_update_realtime(), ipaddr, SENTINEL, and sip_cfg.

Referenced by update_peer().

{
   char port[10];
   char ipaddr[INET6_ADDRSTRLEN];
   char regseconds[20];
   char *tablename = NULL;
   char str_lastms[20];

   const char *sysname = ast_config_AST_SYSTEM_NAME;
   char *syslabel = NULL;

   time_t nowtime = time(NULL) + expirey;
   const char *fc = fullcontact ? "fullcontact" : NULL;

   int realtimeregs = ast_check_realtime("sipregs");

   tablename = realtimeregs ? "sipregs" : "sippeers";

   snprintf(str_lastms, sizeof(str_lastms), "%d", lastms);
   snprintf(regseconds, sizeof(regseconds), "%d", (int)nowtime);  /* Expiration time */
   ast_copy_string(ipaddr, ast_sockaddr_isnull(addr) ? "" : ast_sockaddr_stringify_addr(addr), sizeof(ipaddr));
   ast_copy_string(port, ast_sockaddr_port(addr) ? ast_sockaddr_stringify_port(addr) : "", sizeof(port));

   if (ast_strlen_zero(sysname)) /* No system name, disable this */
      sysname = NULL;
   else if (sip_cfg.rtsave_sysname)
      syslabel = "regserver";

   /* XXX IMPORTANT: Anytime you add a new parameter to be updated, you
         *  must also add it to contrib/scripts/asterisk.ldap-schema,
         *  contrib/scripts/asterisk.ldif,
         *  and to configs/res_ldap.conf.sample as described in
         *  bugs 15156 and 15895
         */
   if (fc) {
      ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
         "port", port, "regseconds", regseconds,
         deprecated_username ? "username" : "defaultuser", defaultuser,
         "useragent", useragent, "lastms", str_lastms,
         fc, fullcontact, syslabel, sysname, SENTINEL); /* note fc and syslabel _can_ be NULL */
   } else {
      ast_update_realtime(tablename, "name", peername, "ipaddr", ipaddr,
         "port", port, "regseconds", regseconds,
         "useragent", useragent, "lastms", str_lastms,
         deprecated_username ? "username" : "defaultuser", defaultuser,
         syslabel, sysname, SENTINEL); /* note syslabel _can_ be NULL */
   }
}

static void receive_message	(	struct sip_pvt *	p,
		struct sip_request *	req,
		struct ast_sockaddr *	addr,
		const char *	e
	)		[static]

Receive SIP MESSAGE method messages.

Note:

We only handle messages within current calls currently Reference: RFC 3428

Definition at line 18565 of file chan_sip.c.

References ast_copy_string(), ast_debug, AST_FRAME_TEXT, ast_log(), ast_msg_alloc(), ast_msg_destroy(), ast_msg_queue(), ast_msg_set_body(), ast_msg_set_context(), ast_msg_set_exten(), ast_msg_set_from(), ast_msg_set_to(), ast_msg_set_var(), ast_queue_frame(), ast_sockaddr_stringify(), ast_string_field_set, ast_strlen_zero(), ast_verbose(), check_user(), context, ast_msg::context, copy_request(), ast_frame::data, ast_frame::datalen, ast_msg::exten, f, ast_frame::frametype, get_calleridname(), get_content(), get_destination(), get_in_brackets(), ast_frame_subclass::integer, len(), LOG_NOTICE, LOG_WARNING, ast_frame::offset, ast_frame::ptr, set_message_vars_from_req(), set_pvt_allowed_methods(), sip_cfg, sip_debug_test_pvt(), sip_find_peer(), sip_get_header(), sip_scheddestroy(), sip_unref_peer(), ast_frame::subclass, transmit_response(), and TRUE.

Referenced by handle_request_message().

{
   char *buf;
   size_t len;
   struct ast_frame f;
   const char *content_type = sip_get_header(req, "Content-Type");
   struct ast_msg *msg;
   int res;
   char *from;
   char *to;
   char from_name[50];
   char stripped[SIPBUFSIZE];

   if (strncmp(content_type, "text/plain", strlen("text/plain"))) { /* No text/plain attachment */
      transmit_response(p, "415 Unsupported Media Type", req); /* Good enough, or? */
      if (!p->owner) {
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
      return;
   }

   if (!(buf = get_content(req))) {
      ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
      transmit_response(p, "500 Internal Server Error", req);
      if (!p->owner) {
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      }
      return;
   }

   /* Strip trailing line feeds from message body. (get_content may add
    * a trailing linefeed and we don't need any at the end) */
   len = strlen(buf);
   while (len > 0) {
      if (buf[--len] != '\n') {
         ++len;
         break;
      }
   }
   buf[len] = '\0';

   if (p->owner) {
      if (sip_debug_test_pvt(p)) {
         ast_verbose("SIP Text message received: '%s'\n", buf);
      }
      memset(&f, 0, sizeof(f));
      f.frametype = AST_FRAME_TEXT;
      f.subclass.integer = 0;
      f.offset = 0;
      f.data.ptr = buf;
      f.datalen = strlen(buf) + 1;
      ast_queue_frame(p->owner, &f);
      transmit_response(p, "202 Accepted", req); /* We respond 202 accepted, since we relay the message */
      return;
   }

   /*
    * At this point MESSAGE is outside of a call.
    *
    * NOTE: p->owner is NULL so no additional check is needed after
    * this point.
    */

   if (!sip_cfg.accept_outofcall_message) {
      /* Message outside of a call, we do not support that */
      ast_debug(1, "MESSAGE outside of a call administratively disabled.\n");
      transmit_response(p, "405 Method Not Allowed", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   copy_request(&p->initreq, req);

   if (sip_cfg.auth_message_requests) {
      int res;

      set_pvt_allowed_methods(p, req);
      res = check_user(p, req, SIP_MESSAGE, e, XMIT_UNRELIABLE, addr);
      if (res == AUTH_CHALLENGE_SENT) {
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }
      if (res < 0) { /* Something failed in authentication */
         ast_log(LOG_NOTICE, "Failed to authenticate device %s\n", sip_get_header(req, "From"));
         transmit_response(p, "403 Forbidden", req);
         sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
         return;
      }
      /* Auth was successful.  Proceed. */
   } else {
      struct sip_peer *peer;

      /*
       * MESSAGE outside of a call, not authenticating it.
       * Check to see if we match a peer anyway so that we can direct
       * it to the right context.
       */

      peer = sip_find_peer(NULL, &p->recv, TRUE, FINDPEERS, 0, p->socket.type);
      if (peer) {
         /* Only if no auth is required. */
         if (ast_strlen_zero(peer->secret) && ast_strlen_zero(peer->md5secret)) {
            ast_string_field_set(p, context, peer->context);
         }
         if (!ast_strlen_zero(peer->messagecontext)) {
            ast_string_field_set(p, messagecontext, peer->messagecontext);
         }
         ast_string_field_set(p, peername, peer->name);
         peer = sip_unref_peer(peer, "from sip_find_peer() in receive_message");
      }
   }

   /* Override the context with the message context _BEFORE_
    * getting the destination.  This way we can guarantee the correct
    * extension is used in the message context when it is present. */
   if (!ast_strlen_zero(p->messagecontext)) {
      ast_string_field_set(p, context, p->messagecontext);
   } else if (!ast_strlen_zero(sip_cfg.messagecontext)) {
      ast_string_field_set(p, context, sip_cfg.messagecontext);
   }

   switch (get_destination(p, NULL, NULL)) {
   case SIP_GET_DEST_REFUSED:
      /* Okay to send 403 since this is after auth processing */
      transmit_response(p, "403 Forbidden", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   case SIP_GET_DEST_INVALID_URI:
      transmit_response(p, "416 Unsupported URI Scheme", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   case SIP_GET_DEST_EXTEN_NOT_FOUND:
   case SIP_GET_DEST_EXTEN_MATCHMORE:
      transmit_response(p, "404 Not Found", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   case SIP_GET_DEST_EXTEN_FOUND:
      break;
   }

   if (!(msg = ast_msg_alloc())) {
      transmit_response(p, "500 Internal Server Error", req);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   to = ast_strdupa(REQ_OFFSET_TO_STR(req, rlpart2));
   from = ast_strdupa(sip_get_header(req, "From"));

   res = ast_msg_set_to(msg, "%s", to);

   /* Build "display" <uri> for from string. */
   from = (char *) get_calleridname(from, from_name, sizeof(from_name));
   from = get_in_brackets(from);
   if (from_name[0]) {
      res |= ast_msg_set_from(msg, "\"%s\" <%s>", from_name, from);
   } else {
      res |= ast_msg_set_from(msg, "<%s>", from);
   }

   res |= ast_msg_set_body(msg, "%s", buf);
   res |= ast_msg_set_context(msg, "%s", p->context);

   res |= ast_msg_set_var(msg, "SIP_RECVADDR", ast_sockaddr_stringify(&p->recv));
   if (!ast_strlen_zero(p->peername)) {
      res |= ast_msg_set_var(msg, "SIP_PEERNAME", p->peername);
   }

   ast_copy_string(stripped, sip_get_header(req, "Contact"), sizeof(stripped));
   res |= ast_msg_set_var(msg, "SIP_FULLCONTACT", get_in_brackets(stripped));

   res |= ast_msg_set_exten(msg, "%s", p->exten);
   res |= set_message_vars_from_req(msg, req);

   if (res) {
      ast_msg_destroy(msg);
      transmit_response(p, "500 Internal Server Error", req);
   } else {
      ast_msg_queue(msg);
      transmit_response(p, "202 Accepted", req);
   }

   sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
}

static void ref_proxy	(	struct sip_pvt *	pvt,
		struct sip_proxy *	proxy
	)		[static]

maintain proper refcounts for a sip_pvt's outboundproxy

This function sets pvt's outboundproxy pointer to the one referenced by the proxy parameter. Because proxy may be a refcounted object, and because pvt's old outboundproxy may also be a refcounted object, we need to maintain the proper refcounts.

Parameters:

pvt	The sip_pvt for which we wish to set the outboundproxy
proxy	The sip_proxy which we will point pvt towards.

Returns:

Returns void

Definition at line 3370 of file chan_sip.c.

References ao2_ref, and sip_cfg.

Referenced by __sip_ack(), __sip_subscribe_mwi_do(), create_addr(), and create_addr_from_peer().

{
   struct sip_proxy *old_obproxy = pvt->outboundproxy;
   /* The sip_cfg.outboundproxy is statically allocated, and so
    * we don't ever need to adjust refcounts for it
    */
   if (proxy && proxy != &sip_cfg.outboundproxy) {
      ao2_ref(proxy, +1);
   }
   pvt->outboundproxy = proxy;
   if (old_obproxy && old_obproxy != &sip_cfg.outboundproxy) {
      ao2_ref(old_obproxy, -1);
   }
}

static const char * referstatus2str

(

enum referstatus

rstatus

)

[static]

Convert transfer status to string.

Definition at line 3494 of file chan_sip.c.

References map_x_s().

Referenced by show_channels_cb().

{
   return map_x_s(referstatusstrings, rstatus, "");
}

static void reg_source_db

(

struct sip_peer *

peer

)

[static]

Get registration details from Asterisk DB.

Definition at line 15828 of file chan_sip.c.

References args, AST_APP_ARG, ast_db_get(), ast_debug, AST_DECLARE_APP_ARGS, AST_NONSTANDARD_RAW_ARGS, ast_random(), AST_SCHED_REPLACE_UNREF, ast_sockaddr_copy(), ast_sockaddr_parse(), ast_sockaddr_stringify_host(), ast_string_field_set, expire_register(), register_peer_exten(), sip_cfg, sip_poke_peer_s(), sip_ref_peer(), sip_unref_peer(), and TRUE.

Referenced by build_peer(), and temp_peer().

{
   char data[256];
   struct ast_sockaddr sa;
   int expire;
   char full_addr[128];
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(addr);
      AST_APP_ARG(port);
      AST_APP_ARG(expiry_str);
      AST_APP_ARG(username);
      AST_APP_ARG(contact);
   );

   /* If read-only RT backend, then refresh from local DB cache */
   if (peer->rt_fromcontact && sip_cfg.peer_rtupdate) {
      return;
   }
   if (ast_db_get("SIP/Registry", peer->name, data, sizeof(data))) {
      return;
   }

   AST_NONSTANDARD_RAW_ARGS(args, data, ':');

   snprintf(full_addr, sizeof(full_addr), "%s:%s", args.addr, args.port);

   if (!ast_sockaddr_parse(&sa, full_addr, 0)) {
      return;
   }

   if (args.expiry_str) {
      expire = atoi(args.expiry_str);
   } else {
      return;
   }

   if (args.username) {
      ast_string_field_set(peer, username, args.username);
   }
   if (args.contact) {
      ast_string_field_set(peer, fullcontact, args.contact);
   }

   ast_debug(2, "SIP Seeding peer from astdb: '%s' at %s@%s for %d\n",
       peer->name, peer->username, ast_sockaddr_stringify_host(&sa), expire);

   ast_sockaddr_copy(&peer->addr, &sa);
   if (peer->maxms) {
      /* Don't poke peer immediately, just schedule it within qualifyfreq */
      AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
            ast_random() % ((peer->qualifyfreq) ? peer->qualifyfreq : global_qualifyfreq) + 1,
            sip_poke_peer_s, peer,
            sip_unref_peer(_data, "removing poke peer ref"),
            sip_unref_peer(peer, "removing poke peer ref"),
            sip_ref_peer(peer, "adding poke peer ref"));
   }
   AST_SCHED_REPLACE_UNREF(peer->expire, sched, (expire + 10) * 1000, expire_register, peer,
         sip_unref_peer(_data, "remove registration ref"),
         sip_unref_peer(peer, "remove registration ref"),
         sip_ref_peer(peer, "add registration ref"));
   register_peer_exten(peer, TRUE);
}

static void register_peer_exten	(	struct sip_peer *	peer,
		int	onoff
	)		[static]

Automatically add peer extension to dial plan.

Definition at line 5127 of file chan_sip.c.

References ast_add_extension(), ast_context_find(), ast_context_remove_extension(), ast_copy_string(), ast_exists_extension(), ast_free_ptr(), ast_log(), ast_strdup, ast_strlen_zero(), context, E_MATCH, ext, LOG_WARNING, pbx_find_extension(), S_OR, sip_cfg, and pbx_find_info::stacklen.

Referenced by expire_register(), handle_response_peerpoke(), parse_register_contact(), reg_source_db(), sip_destroy_peer(), and sip_poke_noanswer().

{
   char multi[256];
   char *stringp, *ext, *context;
   struct pbx_find_info q = { .stacklen = 0 };

   /* XXX note that sip_cfg.regcontext is both a global 'enable' flag and
    * the name of the global regexten context, if not specified
    * individually.
    */
   if (ast_strlen_zero(sip_cfg.regcontext))
      return;

   ast_copy_string(multi, S_OR(peer->regexten, peer->name), sizeof(multi));
   stringp = multi;
   while ((ext = strsep(&stringp, "&"))) {
      if ((context = strchr(ext, '@'))) {
         *context++ = '\0';   /* split ext@context */
         if (!ast_context_find(context)) {
            ast_log(LOG_WARNING, "Context %s must exist in regcontext= in sip.conf!\n", context);
            continue;
         }
      } else {
         context = sip_cfg.regcontext;
      }
      if (onoff) {
         if (!ast_exists_extension(NULL, context, ext, 1, NULL)) {
            ast_add_extension(context, 1, ext, 1, NULL, NULL, "Noop",
                ast_strdup(peer->name), ast_free_ptr, "SIP");
         }
      } else if (pbx_find_extension(NULL, NULL, &q, context, ext, 1, NULL, "", E_MATCH)) {
         ast_context_remove_extension(context, ext, 1, NULL);
      }
   }
}

static int register_realtime_peers_with_callbackextens

(

void

)

[static]

Definition at line 5511 of file chan_sip.c.

References ast_category_browse(), ast_category_root(), ast_check_realtime(), ast_config_destroy(), ast_load_realtime_multientry(), ast_log(), build_peer(), FALSE, LOG_NOTICE, SENTINEL, sip_unref_peer(), TRUE, and var.

Referenced by reload_config().

{
   struct ast_config *cfg;
   char *cat = NULL;

   if (!(ast_check_realtime("sippeers"))) {
      return 0;
   }

   /* This is hacky. We want name to be the cat, so it is the first property */
   if (!(cfg = ast_load_realtime_multientry("sippeers", "name LIKE", "%", "callbackextension LIKE", "%", SENTINEL))) {
      return -1;
   }

   while ((cat = ast_category_browse(cfg, cat))) {
      struct sip_peer *peer;
      struct ast_variable *var = ast_category_root(cfg, cat);

      if (!(peer = build_peer(cat, var, NULL, TRUE, FALSE))) {
         continue;
      }
      ast_log(LOG_NOTICE, "Created realtime peer '%s' for registration\n", peer->name);

      peer->is_realtime = 1;
      sip_unref_peer(peer, "register_realtime_peers: Done registering releasing");
   }

   ast_config_destroy(cfg);

   return 0;
}

static enum check_auth_result register_verify	(	struct sip_pvt *	p,
		struct ast_sockaddr *	addr,
		struct sip_request *	req,
		const char *	uri
	)		[static]

Verify registration of user.

• Registration is done in several steps, first a REGISTER without auth to get a challenge (nonce) then a second one with auth
• Registration requests are only matched with peers that are marked as "dynamic"

Definition at line 16898 of file chan_sip.c.

References ao2_lock, ao2_t_link, ao2_unlock, ast_apply_acl(), ast_copy_flags, ast_copy_string(), AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), AST_LIST_EMPTY, ast_log(), ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_sockaddr_stringify_port(), ast_string_field_set, ast_strlen_zero(), bogus_peer, build_contact(), check_auth(), check_request_transport, check_sip_domain(), EVENT_FLAG_SYSTEM, exten, extract_host_from_hostport(), FALSE, get_in_brackets(), LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, name, parse_register_contact(), parse_uri_legacy_check(), remove_uri_parameters(), set_peer_nat(), sip_cancel_destroy(), sip_cfg, sip_find_peer(), sip_get_header(), SIP_PEDANTIC_DECODE, sip_pvt_lock, sip_pvt_unlock, sip_ref_peer(), sip_send_mwi_to_peer(), sip_unref_peer(), temp_peer(), terminate_uri(), transmit_fake_auth_response(), transmit_response(), transmit_response_with_date(), TRUE, update_peer(), and update_peer_lastmsgssent().

Referenced by handle_request_register().

{
   enum check_auth_result res = AUTH_NOT_FOUND;
   struct sip_peer *peer;
   char tmp[256];
   char *c, *name, *unused_password, *domain;
   char *uri2 = ast_strdupa(uri);
   int send_mwi = 0;

   terminate_uri(uri2);

   ast_copy_string(tmp, sip_get_header(req, "To"), sizeof(tmp));

   c = get_in_brackets(tmp);
   c = remove_uri_parameters(c);

   if (parse_uri_legacy_check(c, "sip:,sips:", &name, &unused_password, &domain, NULL)) {
      ast_log(LOG_NOTICE, "Invalid to address: '%s' from %s (missing sip:) trying to use anyway...\n", c, ast_sockaddr_stringify_addr(addr));
      return -1;
   }

   SIP_PEDANTIC_DECODE(name);
   SIP_PEDANTIC_DECODE(domain);

   extract_host_from_hostport(&domain);

   if (ast_strlen_zero(domain)) {
      /* <sip:name@[EMPTY]>, never good */
      transmit_response(p, "404 Not found", &p->initreq);
      return AUTH_UNKNOWN_DOMAIN;
   }

   if (ast_strlen_zero(name)) {
      /* <sip:[EMPTY][@]hostport>, unsure whether valid for
       * registration. RFC 3261, 10.2 states:
       * "The To header field and the Request-URI field typically
       * differ, as the former contains a user name."
       * But, Asterisk has always treated the domain-only uri as a
       * username: we allow admins to create accounts described by
       * domain name. */
      name = domain;
   }

   /* This here differs from 1.4 and 1.6: the domain matching ACLs were
    * skipped if it was a domain-only URI (used as username). Here we treat
    * <sip:hostport> as <sip:host@hostport> and won't forget to test the
    * domain ACLs against host. */
   if (!AST_LIST_EMPTY(&domain_list)) {
      if (!check_sip_domain(domain, NULL, 0)) {
         if (sip_cfg.alwaysauthreject) {
            transmit_fake_auth_response(p, &p->initreq, XMIT_UNRELIABLE);
         } else {
            transmit_response(p, "404 Not found (unknown domain)", &p->initreq);
         }
         return AUTH_UNKNOWN_DOMAIN;
      }
   }

   ast_string_field_set(p, exten, name);
   build_contact(p);
   if (req->ignore) {
      /* Expires is a special case, where we only want to load the peer if this isn't a deregistration attempt */
      const char *expires = sip_get_header(req, "Expires");
      int expire = atoi(expires);

      if (ast_strlen_zero(expires)) { /* No expires header; look in Contact */
         if ((expires = strcasestr(sip_get_header(req, "Contact"), ";expires="))) {
            expire = atoi(expires + 9);
         }
      }
      if (!ast_strlen_zero(expires) && expire == 0) {
         transmit_response_with_date(p, "200 OK", req);
         return 0;
      }
   }
   peer = sip_find_peer(name, NULL, TRUE, FINDPEERS, FALSE, 0);

   /* If we don't want username disclosure, use the bogus_peer when a user
    * is not found. */
   if (!peer && sip_cfg.alwaysauthreject && sip_cfg.autocreatepeer == AUTOPEERS_DISABLED) {
      peer = bogus_peer;
      sip_ref_peer(peer, "register_verify: ref the bogus_peer");
   }

   if (!(peer && ast_apply_acl(peer->acl, addr, "SIP Peer ACL: "))) {
      /* Peer fails ACL check */
      if (peer) {
         sip_unref_peer(peer, "register_verify: sip_unref_peer: from sip_find_peer operation");
         peer = NULL;
         res = AUTH_ACL_FAILED;
      } else {
         res = AUTH_NOT_FOUND;
      }
   }

   if (peer) {
      ao2_lock(peer);
      if (!peer->host_dynamic) {
         ast_log(LOG_ERROR, "Peer '%s' is trying to register, but not configured as host=dynamic\n", peer->name);
         res = AUTH_PEER_NOT_DYNAMIC;
      } else {

         set_peer_nat(p, peer);

         ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_NAT_FORCE_RPORT);

         if (!(res = check_auth(p, req, peer->name, peer->secret, peer->md5secret, SIP_REGISTER, uri2, XMIT_UNRELIABLE))) {
            if (sip_cancel_destroy(p))
               ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");

            if (check_request_transport(peer, req)) {
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
               transmit_response_with_date(p, "403 Forbidden", req);
               res = AUTH_BAD_TRANSPORT;
            } else {

               /* We have a successful registration attempt with proper authentication,
               now, update the peer */
               switch (parse_register_contact(p, peer, req)) {
               case PARSE_REGISTER_DENIED:
                  ast_log(LOG_WARNING, "Registration denied because of contact ACL\n");
                  transmit_response_with_date(p, "603 Denied", req);
                  res = 0;
                  break;
               case PARSE_REGISTER_FAILED:
                  ast_log(LOG_WARNING, "Failed to parse contact info\n");
                  transmit_response_with_date(p, "400 Bad Request", req);
                  res = 0;
                  break;
               case PARSE_REGISTER_QUERY:
                  ast_string_field_set(p, fullcontact, peer->fullcontact);
                  transmit_response_with_date(p, "200 OK", req);
                  res = 0;
                  send_mwi = 1;
                  break;
               case PARSE_REGISTER_UPDATE:
                  ast_string_field_set(p, fullcontact, peer->fullcontact);
                  /* If expiry is 0, peer has been unregistered already */
                  if (p->expiry != 0) {
                     update_peer(peer, p->expiry);
                  }
                  /* Say OK and ask subsystem to retransmit msg counter */
                  transmit_response_with_date(p, "200 OK", req);
                  send_mwi = 1;
                  res = 0;
                  break;
               }
            }

         }
      }
      ao2_unlock(peer);
   }
   if (!peer && sip_cfg.autocreatepeer != AUTOPEERS_DISABLED) {
      /* Create peer if we have autocreate mode enabled */
      peer = temp_peer(name);
      if (peer) {
         ao2_t_link(peers, peer, "link peer into peer table");
         if (!ast_sockaddr_isnull(&peer->addr)) {
            ao2_t_link(peers_by_ip, peer, "link peer into peers-by-ip table");
         }
         ao2_lock(peer);
         if (sip_cancel_destroy(p))
            ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
         switch (parse_register_contact(p, peer, req)) {
         case PARSE_REGISTER_DENIED:
            ast_log(LOG_WARNING, "Registration denied because of contact ACL\n");
            transmit_response_with_date(p, "403 Forbidden", req);
            res = 0;
            break;
         case PARSE_REGISTER_FAILED:
            ast_log(LOG_WARNING, "Failed to parse contact info\n");
            transmit_response_with_date(p, "400 Bad Request", req);
            res = 0;
            break;
         case PARSE_REGISTER_QUERY:
            ast_string_field_set(p, fullcontact, peer->fullcontact);
            transmit_response_with_date(p, "200 OK", req);
            send_mwi = 1;
            res = 0;
            break;
         case PARSE_REGISTER_UPDATE:
            ast_string_field_set(p, fullcontact, peer->fullcontact);
            /* Say OK and ask subsystem to retransmit msg counter */
            transmit_response_with_date(p, "200 OK", req);
            manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\n", peer->name, ast_sockaddr_stringify(addr));
            send_mwi = 1;
            res = 0;
            break;
         }
         ao2_unlock(peer);
      }
   }
   if (!res) {
      if (send_mwi) {
         sip_pvt_unlock(p);
         sip_send_mwi_to_peer(peer, 0);
         sip_pvt_lock(p);
      } else {
         update_peer_lastmsgssent(peer, -1, 0);
      }
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
   }
   if (res < 0) {
      switch (res) {
      case AUTH_SECRET_FAILED:
         /* Wrong password in authentication. Go away, don't try again until you fixed it */
         transmit_response(p, "403 Forbidden", &p->initreq);
         if (global_authfailureevents) {
            const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
            const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
            manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
                     "ChannelType: SIP\r\n"
                     "Peer: SIP/%s\r\n"
                     "PeerStatus: Rejected\r\n"
                     "Cause: AUTH_SECRET_FAILED\r\n"
                     "Address: %s\r\n"
                     "Port: %s\r\n",
                     name, peer_addr, peer_port);
         }
         break;
      case AUTH_USERNAME_MISMATCH:
         /* Username and digest username does not match.
            Asterisk uses the From: username for authentication. We need the
            devices to use the same authentication user name until we support
            proper authentication by digest auth name */
      case AUTH_NOT_FOUND:
      case AUTH_PEER_NOT_DYNAMIC:
      case AUTH_ACL_FAILED:
         if (sip_cfg.alwaysauthreject) {
            transmit_fake_auth_response(p, &p->initreq, XMIT_UNRELIABLE);
            if (global_authfailureevents) {
               const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
               const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
               manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
                        "ChannelType: SIP\r\n"
                        "Peer: SIP/%s\r\n"
                        "PeerStatus: Rejected\r\n"
                        "Cause: %s\r\n"
                        "Address: %s\r\n"
                        "Port: %s\r\n",
                        name,
                        res == AUTH_PEER_NOT_DYNAMIC ? "AUTH_PEER_NOT_DYNAMIC" : "URI_NOT_FOUND",
                        peer_addr, peer_port);
            }
         } else {
            /* URI not found */
            if (res == AUTH_PEER_NOT_DYNAMIC) {
               transmit_response(p, "403 Forbidden", &p->initreq);
               if (global_authfailureevents) {
                  const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
                  const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
                  manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
                     "ChannelType: SIP\r\n"
                     "Peer: SIP/%s\r\n"
                     "PeerStatus: Rejected\r\n"
                     "Cause: AUTH_PEER_NOT_DYNAMIC\r\n"
                     "Address: %s\r\n"
                     "Port: %s\r\n",
                     name, peer_addr, peer_port);
               }
            } else {
               transmit_response(p, "404 Not found", &p->initreq);
               if (global_authfailureevents) {
                  const char *peer_addr = ast_strdupa(ast_sockaddr_stringify_addr(addr));
                  const char *peer_port = ast_strdupa(ast_sockaddr_stringify_port(addr));
                  manager_event(EVENT_FLAG_SYSTEM, "PeerStatus",
                     "ChannelType: SIP\r\n"
                     "Peer: SIP/%s\r\n"
                     "PeerStatus: Rejected\r\n"
                     "Cause: %s\r\n"
                     "Address: %s\r\n"
                     "Port: %s\r\n",
                     name,
                     (res == AUTH_USERNAME_MISMATCH) ? "AUTH_USERNAME_MISMATCH" : "URI_NOT_FOUND",
                     peer_addr, peer_port);
               }
            }
         }
         break;
      case AUTH_BAD_TRANSPORT:
      default:
         break;
      }
   }
   if (peer) {
      sip_unref_peer(peer, "register_verify: sip_unref_peer: tossing stack peer pointer at end of func");
   }

   return res;
}

static struct sip_registry* registry_addref	(	struct sip_registry *	reg,
		char *	tag
	)		[static, read]

Add object reference to SIP registry.

Definition at line 3476 of file chan_sip.c.

References ast_debug, and ASTOBJ_REF.

Referenced by handle_response_register(), sip_send_all_registers(), and transmit_register().

{
   ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
   return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}

void * registry_unref	(	struct sip_registry *	reg,
		char *	tag
	)		[static]

Definition at line 3468 of file chan_sip.c.

References ast_debug, ASTOBJ_UNREF, and sip_registry_destroy().

Referenced by __sip_destroy(), cleanup_all_regs(), dialog_unlink_all(), handle_response_register(), sip_reg_timeout(), sip_register(), sip_registry_destroy(), sip_reregister(), sip_send_all_registers(), and transmit_register().

{
   ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
   ASTOBJ_UNREF(reg, sip_registry_destroy);
   return NULL;
}

static const char * regstate2str

(

enum sipregistrystate

regstate

)

[static]

Convert registration state status to string.

Definition at line 15079 of file chan_sip.c.

References map_x_s().

Referenced by handle_response_register(), manager_show_registry(), sip_reg_timeout(), and sip_show_registry().

{
   return map_x_s(regstatestrings, regstate, "Unknown");
}

static int reinvite_timeout

(

const void *

data

)

[static]

Definition at line 7008 of file chan_sip.c.

References ao2_unlock, ast_channel_unlock, ast_channel_unref, check_pendings(), and sip_pvt_lock_full().

Referenced by sip_hangup().

{
   struct sip_pvt *dialog = (struct sip_pvt *) data;
   struct ast_channel *owner = sip_pvt_lock_full(dialog);
   dialog->reinviteid = -1;
   check_pendings(dialog);
   if (owner) {
      ast_channel_unlock(owner);
      ast_channel_unref(owner);
   }
   ao2_unlock(dialog);
   dialog_unref(dialog, "unref for reinvite timeout");
   return 0;
}

static int reload

(

void

)

[static]

Part of Asterisk module interface.

Definition at line 33421 of file chan_sip.c.

References sip_reload().

{
   if (sip_reload(0, 0, NULL)) {
      return 0;
   }
   return 1;
}

static int reload_config

(

enum channelreloadreason

reason

)

[static]

Re-read SIP.conf config file.

Note:

This function reloads all config data, except for active peers (with registrations). They will only change configuration data at restart, not at reload. SIP debug and recordhistory state will not change

< Don't force proxy usage, use route: headers

< Keep track of hold status for a peer

< Match auth username if available instead of From: Default off.

< Default DTMF setting: RFC2833

< Allow re-invites

< Default to nat=auto_force_rport

Definition at line 31422 of file chan_sip.c.

References ast_tcptls_session_args::accept_fd, acl_change_event_subscribe(), add_realm_authentication(), add_sip_domain(), ao2_t_callback, ao2_t_link, ao2_t_ref, ast_append_acl(), ast_append_ha(), ast_bind(), ast_category_browse(), AST_CERTFILE, ast_clear_flag, ast_config_AST_SYSTEM_NAME, ast_config_destroy(), ast_config_load, ast_context_find_or_create(), ast_copy_flags, ast_copy_string(), ast_debug, ast_enable_packet_fragmentation(), ast_false(), ast_find_ourip(), AST_FLAGS_ALL, ast_free, ast_free_acl_list(), ast_free_ha(), ast_get_indication_zone(), ast_get_version(), ast_jb_read_conf(), AST_LIST_EMPTY, ast_log(), AST_MAX_CONTEXT, ast_mutex_lock, ast_mutex_unlock, ast_parse_allow_disallow(), ast_parse_arg(), ast_set2_flag, ast_set_flag, ast_set_qos(), ast_skip_blanks(), ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_parse(), ast_sockaddr_port, ast_sockaddr_resolve_first(), ast_sockaddr_set_port, ast_sockaddr_setnull(), ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_ssl_setup(), ast_str2cos(), ast_str2tos(), ast_strdup, ast_strip(), ast_strlen_zero(), ast_tcptls_server_start(), ast_test_flag, ast_tls_read_conf(), ast_tone_zone_unref(), ast_true(), ast_unload_realtime(), ast_variable_browse(), ast_variable_retrieve(), ast_verb, ASTOBJ_CONTAINER_DESTROYALL, authl_lock, bindaddr, build_peer(), ast_tls_config::cafile, ast_tls_config::capath, ast_tls_config::certfile, CHANNEL_ACL_RELOAD, CHANNEL_MODULE_LOAD, channelreloadreason2txt(), ast_tls_config::cipher, cleanup_all_regs(), cleanup_stale_contexts(), clear_sip_domains(), CONFIG_FLAG_FILEUNCHANGED, CONFIG_STATUS_FILEINVALID, CONFIG_STATUS_FILEUNCHANGED, context, DEFAULT_ALLOWGUEST, DEFAULT_CONTEXT, DEFAULT_MAXMS, DEFAULT_PARKINGLOT, default_prefs, DEFAULT_REALM, default_transports, display_nat_warning(), ast_tls_config::enabled, errno, EVENT_FLAG_SYSTEM, externaddr, FALSE, gen, global_jbconf, global_t1min, handle_common_options(), handle_t38_options(), internip, ast_variable::lineno, ast_tcptls_session_args::local_address, LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, mark_parsed_methods(), max_expiry, media_address, min_expiry, ast_variable::name, netlock, network_change_event_subscribe(), network_change_event_unsubscribe(), ast_variable::next, OBJ_NODATA, PARSE_ADDR, PARSE_DEFAULT, PARSE_IN_RANGE, PARSE_INT32, peer_markall_autopeers_func(), peer_markall_func(), port_str2int(), proxy_from_config(), ast_tls_config::pvtfile, register_realtime_peers_with_callbackextens(), regl, S_OR, secret, sip_cfg, sip_register(), sip_registry_destroy(), sip_set_default_format_capabilities(), sip_subscribe_mwi(), sip_unref_peer(), str2stmode(), str2strefresherparam(), ast_tcptls_session_args::tls_cfg, TRUE, and ast_variable::value.

Referenced by load_module(), and sip_do_reload().

{
   struct ast_config *cfg, *ucfg;
   struct ast_variable *v;
   struct sip_peer *peer;
   char *cat, *stringp, *context, *oldregcontext;
   char newcontexts[AST_MAX_CONTEXT], oldcontexts[AST_MAX_CONTEXT];
   struct ast_flags mask[3] = {{0}};
   struct ast_flags setflags[3] = {{0}};
   struct ast_flags config_flags = { (reason == CHANNEL_MODULE_LOAD || reason == CHANNEL_ACL_RELOAD) ? 0 : ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) ? 0 : CONFIG_FLAG_FILEUNCHANGED };
   int auto_sip_domains = FALSE;
   struct ast_sockaddr old_bindaddr = bindaddr;
   int registry_count = 0, peer_count = 0, timerb_set = 0, timert1_set = 0;
   int subscribe_network_change = 1;
   time_t run_start, run_end;
   int bindport = 0;
   int acl_change_subscription_needed = 0;
   int min_subexpiry_set = 0, max_subexpiry_set = 0;

   run_start = time(0);
   ast_unload_realtime("sipregs");
   ast_unload_realtime("sippeers");
   cfg = ast_config_load(config, config_flags);

   /* We *must* have a config file otherwise stop immediately */
   if (!cfg) {
      ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
      return -1;
   } else if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
      ucfg = ast_config_load("users.conf", config_flags);
      if (ucfg == CONFIG_STATUS_FILEUNCHANGED) {
         return 1;
      } else if (ucfg == CONFIG_STATUS_FILEINVALID) {
         ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
         return 1;
      }
      /* Must reread both files, because one changed */
      ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
      if ((cfg = ast_config_load(config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
         ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
         ast_config_destroy(ucfg);
         return 1;
      }
      if (!cfg) {
         /* should have been able to reload here */
         ast_log(LOG_NOTICE, "Unable to load config %s\n", config);
         return -1;
      }
   } else if (cfg == CONFIG_STATUS_FILEINVALID) {
      ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed\n", config);
      return 1;
   } else {
      ast_clear_flag(&config_flags, CONFIG_FLAG_FILEUNCHANGED);
      if ((ucfg = ast_config_load("users.conf", config_flags)) == CONFIG_STATUS_FILEINVALID) {
         ast_log(LOG_ERROR, "Contents of users.conf are invalid and cannot be parsed\n");
         ast_config_destroy(cfg);
         return 1;
      }
   }

   sip_cfg.contact_acl = ast_free_acl_list(sip_cfg.contact_acl);

   default_tls_cfg.enabled = FALSE;    /* Default: Disable TLS */

   if (reason != CHANNEL_MODULE_LOAD) {
      ast_debug(4, "--------------- SIP reload started\n");

      clear_sip_domains();
      ast_mutex_lock(&authl_lock);
      if (authl) {
         ao2_t_ref(authl, -1, "Removing old global authentication");
         authl = NULL;
      }
      ast_mutex_unlock(&authl_lock);

      cleanup_all_regs();

      /* Then, actually destroy users and registry */
      ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
      ast_debug(4, "--------------- Done destroying registry list\n");
      ao2_t_callback(peers, OBJ_NODATA, peer_markall_func, NULL, "callback to mark all peers");
   }

   /* Reset certificate handling for TLS sessions */
   if (reason != CHANNEL_MODULE_LOAD) {
      ast_free(default_tls_cfg.certfile);
      ast_free(default_tls_cfg.pvtfile);
      ast_free(default_tls_cfg.cipher);
      ast_free(default_tls_cfg.cafile);
      ast_free(default_tls_cfg.capath);
   }
   default_tls_cfg.certfile = ast_strdup(AST_CERTFILE); /*XXX Not sure if this is useful */
   default_tls_cfg.pvtfile = ast_strdup("");
   default_tls_cfg.cipher = ast_strdup("");
   default_tls_cfg.cafile = ast_strdup("");
   default_tls_cfg.capath = ast_strdup("");

   /* Initialize copy of current sip_cfg.regcontext for later use in removing stale contexts */
   ast_copy_string(oldcontexts, sip_cfg.regcontext, sizeof(oldcontexts));
   oldregcontext = oldcontexts;

   /* Clear all flags before setting default values */
   /* Preserve debugging settings for console */
   sipdebug &= sip_debug_console;
   ast_clear_flag(&global_flags[0], AST_FLAGS_ALL);
   ast_clear_flag(&global_flags[1], AST_FLAGS_ALL);
   ast_clear_flag(&global_flags[2], AST_FLAGS_ALL);

   /* Reset IP addresses  */
   ast_sockaddr_parse(&bindaddr, "0.0.0.0:0", 0);
   memset(&internip, 0, sizeof(internip));

   /* Free memory for local network address mask */
   ast_free_ha(localaddr);
   memset(&localaddr, 0, sizeof(localaddr));
   memset(&externaddr, 0, sizeof(externaddr));
   memset(&media_address, 0, sizeof(media_address));
   memset(&default_prefs, 0 , sizeof(default_prefs));
   memset(&sip_cfg.outboundproxy, 0, sizeof(struct sip_proxy));
   sip_cfg.outboundproxy.force = FALSE;      /*!< Don't force proxy usage, use route: headers */
   default_transports = SIP_TRANSPORT_UDP;
   default_primary_transport = SIP_TRANSPORT_UDP;
   ourport_tcp = STANDARD_SIP_PORT;
   ourport_tls = STANDARD_TLS_PORT;
   externtcpport = STANDARD_SIP_PORT;
   externtlsport = STANDARD_TLS_PORT;
   sip_cfg.srvlookup = DEFAULT_SRVLOOKUP;
   global_tos_sip = DEFAULT_TOS_SIP;
   global_tos_audio = DEFAULT_TOS_AUDIO;
   global_tos_video = DEFAULT_TOS_VIDEO;
   global_tos_text = DEFAULT_TOS_TEXT;
   global_cos_sip = DEFAULT_COS_SIP;
   global_cos_audio = DEFAULT_COS_AUDIO;
   global_cos_video = DEFAULT_COS_VIDEO;
   global_cos_text = DEFAULT_COS_TEXT;

   externhost[0] = '\0';         /* External host name (for behind NAT DynDNS support) */
   externexpire = 0;       /* Expiration for DNS re-issuing */
   externrefresh = 10;

   /* Reset channel settings to default before re-configuring */
   sip_cfg.allow_external_domains = DEFAULT_ALLOW_EXT_DOM;           /* Allow external invites */
   sip_cfg.regcontext[0] = '\0';
   sip_set_default_format_capabilities(sip_cfg.caps);
   sip_cfg.regextenonqualify = DEFAULT_REGEXTENONQUALIFY;
   sip_cfg.legacy_useroption_parsing = DEFAULT_LEGACY_USEROPTION_PARSING;
   sip_cfg.send_diversion = DEFAULT_SEND_DIVERSION;
   sip_cfg.notifyringing = DEFAULT_NOTIFYRINGING;
   sip_cfg.notifycid = DEFAULT_NOTIFYCID;
   sip_cfg.notifyhold = FALSE;      /*!< Keep track of hold status for a peer */
   sip_cfg.directrtpsetup = FALSE;     /* Experimental feature, disabled by default */
   sip_cfg.alwaysauthreject = DEFAULT_ALWAYSAUTHREJECT;
   sip_cfg.auth_options_requests = DEFAULT_AUTH_OPTIONS;
   sip_cfg.auth_message_requests = DEFAULT_AUTH_MESSAGE;
   sip_cfg.messagecontext[0] = '\0';
   sip_cfg.accept_outofcall_message = DEFAULT_ACCEPT_OUTOFCALL_MESSAGE;
   sip_cfg.allowsubscribe = FALSE;
   sip_cfg.disallowed_methods = SIP_UNKNOWN;
   sip_cfg.contact_acl = NULL;      /* Reset the contact ACL */
   snprintf(global_useragent, sizeof(global_useragent), "%s %s", DEFAULT_USERAGENT, ast_get_version());
   snprintf(global_sdpsession, sizeof(global_sdpsession), "%s %s", DEFAULT_SDPSESSION, ast_get_version());
   snprintf(global_sdpowner, sizeof(global_sdpowner), "%s", DEFAULT_SDPOWNER);
   global_prematuremediafilter = TRUE;
   ast_copy_string(default_notifymime, DEFAULT_NOTIFYMIME, sizeof(default_notifymime));
   ast_copy_string(sip_cfg.realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM), sizeof(sip_cfg.realm));
   sip_cfg.domainsasrealm = DEFAULT_DOMAINSASREALM;
   ast_copy_string(default_callerid, DEFAULT_CALLERID, sizeof(default_callerid));
   ast_copy_string(default_mwi_from, DEFAULT_MWI_FROM, sizeof(default_mwi_from));
   sip_cfg.compactheaders = DEFAULT_COMPACTHEADERS;
   global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
   global_regattempts_max = 0;
   global_reg_retry_403 = 0;
   sip_cfg.pedanticsipchecking = DEFAULT_PEDANTIC;
   sip_cfg.autocreatepeer = DEFAULT_AUTOCREATEPEER;
   global_autoframing = 0;
   sip_cfg.allowguest = DEFAULT_ALLOWGUEST;
   global_callcounter = DEFAULT_CALLCOUNTER;
   global_match_auth_username = FALSE;    /*!< Match auth username if available instead of From: Default off. */
   global_rtptimeout = 0;
   global_rtpholdtimeout = 0;
   global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
   sip_cfg.allowtransfer = TRANSFER_OPENFORALL; /* Merrily accept all transfers by default */
   sip_cfg.rtautoclear = 120;
   ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE);   /* Default for all devices: TRUE */
   ast_set_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP_YES); /* Default for all devices: Yes */
   sip_cfg.peer_rtupdate = TRUE;
   global_dynamic_exclude_static = 0;  /* Exclude static peers */
   sip_cfg.tcp_enabled = FALSE;

   /* Session-Timers */
   global_st_mode = SESSION_TIMER_MODE_ACCEPT;
   global_st_refresher = SESSION_TIMER_REFRESHER_PARAM_UAS;
   global_min_se  = DEFAULT_MIN_SE;
   global_max_se  = DEFAULT_MAX_SE;

   /* Peer poking settings */
   global_qualify_gap = DEFAULT_QUALIFY_GAP;
   global_qualify_peers = DEFAULT_QUALIFY_PEERS;

   /* Initialize some reasonable defaults at SIP reload (used both for channel and as default for devices */
   ast_copy_string(sip_cfg.default_context, DEFAULT_CONTEXT, sizeof(sip_cfg.default_context));
   ast_copy_string(sip_cfg.default_record_on_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_on_feature));
   ast_copy_string(sip_cfg.default_record_off_feature, DEFAULT_RECORD_FEATURE, sizeof(sip_cfg.default_record_off_feature));
   sip_cfg.default_subscribecontext[0] = '\0';
   sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
   default_language[0] = '\0';
   default_fromdomain[0] = '\0';
   default_fromdomainport = 0;
   default_qualify = DEFAULT_QUALIFY;
   default_keepalive = DEFAULT_KEEPALIVE;
   default_zone[0] = '\0';
   default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
   ast_copy_string(default_mohinterpret, DEFAULT_MOHINTERPRET, sizeof(default_mohinterpret));
   ast_copy_string(default_mohsuggest, DEFAULT_MOHSUGGEST, sizeof(default_mohsuggest));
   ast_copy_string(default_vmexten, DEFAULT_VMEXTEN, sizeof(default_vmexten));
   ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833);    /*!< Default DTMF setting: RFC2833 */
   ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA);    /*!< Allow re-invites */
   ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */
   ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
   ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot));

   /* Debugging settings, always default to off */
   dumphistory = FALSE;
   recordhistory = FALSE;
   sipdebug &= ~sip_debug_config;

   /* Misc settings for the channel */
   global_relaxdtmf = FALSE;
   sip_cfg.callevents = DEFAULT_CALLEVENTS;
   global_authfailureevents = FALSE;
   global_t1 = DEFAULT_TIMER_T1;
   global_timer_b = 64 * DEFAULT_TIMER_T1;
   global_t1min = DEFAULT_T1MIN;
   global_qualifyfreq = DEFAULT_QUALIFYFREQ;
   global_t38_maxdatagram = -1;
   global_shrinkcallerid = 1;
   global_refer_addheaders = TRUE;
   authlimit = DEFAULT_AUTHLIMIT;
   authtimeout = DEFAULT_AUTHTIMEOUT;
   global_store_sip_cause = DEFAULT_STORE_SIP_CAUSE;
   min_expiry = DEFAULT_MIN_EXPIRY;
   max_expiry = DEFAULT_MAX_EXPIRY;
   default_expiry = DEFAULT_DEFAULT_EXPIRY;

   sip_cfg.matchexternaddrlocally = DEFAULT_MATCHEXTERNADDRLOCALLY;

   /* Copy the default jb config over global_jbconf */
   memcpy(&global_jbconf, &default_jbconf, sizeof(struct ast_jb_conf));

   ast_clear_flag(&global_flags[1], SIP_PAGE2_FAX_DETECT);
   ast_clear_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_VIDEOSUPPORT_ALWAYS);
   ast_clear_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT);
   ast_clear_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION);

   /* Read the [general] config section of sip.conf (or from realtime config) */
   for (v = ast_variable_browse(cfg, "general"); v; v = v->next) {
      if (handle_common_options(&setflags[0], &mask[0], v)) {
         continue;
      }
      if (handle_t38_options(&setflags[0], &mask[0], v, &global_t38_maxdatagram)) {
         continue;
      }
      /* handle jb conf */
      if (!ast_jb_read_conf(&global_jbconf, v->name, v->value)) {
         continue;
      }

      /* handle tls conf, don't allow setting of tlsverifyclient as it isn't supported by chan_sip */
      if (!strcasecmp(v->name, "tlsverifyclient")) {
         ast_log(LOG_WARNING, "Ignoring unsupported option 'tlsverifyclient'\n");
         continue;
      } else if (!ast_tls_read_conf(&default_tls_cfg, &sip_tls_desc, v->name, v->value)) {
         continue;
      }

      if (!strcasecmp(v->name, "context")) {
         ast_copy_string(sip_cfg.default_context, v->value, sizeof(sip_cfg.default_context));
      } else if (!strcasecmp(v->name, "recordonfeature")) {
         ast_copy_string(sip_cfg.default_record_on_feature, v->value, sizeof(sip_cfg.default_record_on_feature));
      } else if (!strcasecmp(v->name, "recordofffeature")) {
         ast_copy_string(sip_cfg.default_record_off_feature, v->value, sizeof(sip_cfg.default_record_off_feature));
      } else if (!strcasecmp(v->name, "subscribecontext")) {
         ast_copy_string(sip_cfg.default_subscribecontext, v->value, sizeof(sip_cfg.default_subscribecontext));
      } else if (!strcasecmp(v->name, "callcounter")) {
         global_callcounter = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "allowguest")) {
         sip_cfg.allowguest = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "realm")) {
         ast_copy_string(sip_cfg.realm, v->value, sizeof(sip_cfg.realm));
      } else if (!strcasecmp(v->name, "domainsasrealm")) {
         sip_cfg.domainsasrealm = ast_true(v->value);
      } else if (!strcasecmp(v->name, "useragent")) {
         ast_copy_string(global_useragent, v->value, sizeof(global_useragent));
         ast_debug(1, "Setting SIP channel User-Agent Name to %s\n", global_useragent);
      } else if (!strcasecmp(v->name, "sdpsession")) {
         ast_copy_string(global_sdpsession, v->value, sizeof(global_sdpsession));
      } else if (!strcasecmp(v->name, "sdpowner")) {
         /* Field cannot contain spaces */
         if (!strstr(v->value, " ")) {
            ast_copy_string(global_sdpowner, v->value, sizeof(global_sdpowner));
         } else {
            ast_log(LOG_WARNING, "'%s' must not contain spaces at line %d.  Using default.\n", v->value, v->lineno);
         }
      } else if (!strcasecmp(v->name, "allowtransfer")) {
         sip_cfg.allowtransfer = ast_true(v->value) ? TRANSFER_OPENFORALL : TRANSFER_CLOSED;
      } else if (!strcasecmp(v->name, "rtcachefriends")) {
         ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_RTCACHEFRIENDS);
      } else if (!strcasecmp(v->name, "rtsavesysname")) {
         sip_cfg.rtsave_sysname = ast_true(v->value);
      } else if (!strcasecmp(v->name, "rtupdate")) {
         sip_cfg.peer_rtupdate = ast_true(v->value);
      } else if (!strcasecmp(v->name, "ignoreregexpire")) {
         sip_cfg.ignore_regexpire = ast_true(v->value);
      } else if (!strcasecmp(v->name, "timert1")) {
         /* Defaults to 500ms, but RFC 3261 states that it is recommended
          * for the value to be set higher, though a lower value is only
          * allowed on private networks unconnected to the Internet. */
         global_t1 = atoi(v->value);
      } else if (!strcasecmp(v->name, "timerb")) {
         int tmp = atoi(v->value);
         if (tmp < 500) {
            global_timer_b = global_t1 * 64;
            ast_log(LOG_WARNING, "Invalid value for timerb ('%s').  Setting to default ('%d').\n", v->value, global_timer_b);
         }
         timerb_set = 1;
      } else if (!strcasecmp(v->name, "t1min")) {
         global_t1min = atoi(v->value);
      } else if (!strcasecmp(v->name, "transport")) {
         char *val = ast_strdupa(v->value);
         char *trans;

         default_transports = default_primary_transport = 0;
         while ((trans = strsep(&val, ","))) {
            trans = ast_skip_blanks(trans);

            if (!strncasecmp(trans, "udp", 3)) {
               default_transports |= SIP_TRANSPORT_UDP;
            } else if (!strncasecmp(trans, "tcp", 3)) {
               default_transports |= SIP_TRANSPORT_TCP;
            } else if (!strncasecmp(trans, "tls", 3)) {
               default_transports |= SIP_TRANSPORT_TLS;
            } else if (!strncasecmp(trans, "wss", 3)) {
               default_transports |= SIP_TRANSPORT_WSS;
            } else if (!strncasecmp(trans, "ws", 2)) {
               default_transports |= SIP_TRANSPORT_WS;
            } else {
               ast_log(LOG_NOTICE, "'%s' is not a valid transport type. if no other is specified, udp will be used.\n", trans);
            }
            if (default_primary_transport == 0) {
               default_primary_transport = default_transports;
            }
         }
      } else if (!strcasecmp(v->name, "tcpenable")) {
         if (!ast_false(v->value)) {
            ast_debug(2, "Enabling TCP socket for listening\n");
            sip_cfg.tcp_enabled = TRUE;
         }
      } else if (!strcasecmp(v->name, "tcpbindaddr")) {
         if (ast_parse_arg(v->value, PARSE_ADDR,
                 &sip_tcp_desc.local_address)) {
            ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
               v->name, v->value, v->lineno, config);
         }
         ast_debug(2, "Setting TCP socket address to %s\n",
              ast_sockaddr_stringify(&sip_tcp_desc.local_address));
      } else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) {
         global_dynamic_exclude_static = ast_true(v->value);
      } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny") || !strcasecmp(v->name, "contactacl")) {
         int ha_error = 0;
         ast_append_acl(v->name + 7, v->value, &sip_cfg.contact_acl, &ha_error, &acl_change_subscription_needed);
         if (ha_error) {
            ast_log(LOG_ERROR, "Bad ACL entry in configuration line %d : %s\n", v->lineno, v->value);
         }
      } else if (!strcasecmp(v->name, "rtautoclear")) {
         int i = atoi(v->value);
         if (i > 0) {
            sip_cfg.rtautoclear = i;
         } else {
            i = 0;
         }
         ast_set2_flag(&global_flags[1], i || ast_true(v->value), SIP_PAGE2_RTAUTOCLEAR);
      } else if (!strcasecmp(v->name, "usereqphone")) {
         ast_set2_flag(&global_flags[0], ast_true(v->value), SIP_USEREQPHONE);
      } else if (!strcasecmp(v->name, "prematuremedia")) {
         global_prematuremediafilter = ast_true(v->value);
      } else if (!strcasecmp(v->name, "relaxdtmf")) {
         global_relaxdtmf = ast_true(v->value);
      } else if (!strcasecmp(v->name, "vmexten")) {
         ast_copy_string(default_vmexten, v->value, sizeof(default_vmexten));
      } else if (!strcasecmp(v->name, "rtptimeout")) {
         if ((sscanf(v->value, "%30d", &global_rtptimeout) != 1) || (global_rtptimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtptimeout = 0;
         }
      } else if (!strcasecmp(v->name, "rtpholdtimeout")) {
         if ((sscanf(v->value, "%30d", &global_rtpholdtimeout) != 1) || (global_rtpholdtimeout < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP hold time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtpholdtimeout = 0;
         }
      } else if (!strcasecmp(v->name, "rtpkeepalive")) {
         if ((sscanf(v->value, "%30d", &global_rtpkeepalive) != 1) || (global_rtpkeepalive < 0)) {
            ast_log(LOG_WARNING, "'%s' is not a valid RTP keepalive time at line %d.  Using default.\n", v->value, v->lineno);
            global_rtpkeepalive = DEFAULT_RTPKEEPALIVE;
         }
      } else if (!strcasecmp(v->name, "compactheaders")) {
         sip_cfg.compactheaders = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifymimetype")) {
         ast_copy_string(default_notifymime, v->value, sizeof(default_notifymime));
      } else if (!strcasecmp(v->name, "directrtpsetup")) {
         sip_cfg.directrtpsetup = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifyringing")) {
         sip_cfg.notifyringing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifyhold")) {
         sip_cfg.notifyhold = ast_true(v->value);
      } else if (!strcasecmp(v->name, "notifycid")) {
         if (!strcasecmp(v->value, "ignore-context")) {
            sip_cfg.notifycid = IGNORE_CONTEXT;
         } else {
            sip_cfg.notifycid = ast_true(v->value) ? ENABLED : DISABLED;
         }
      } else if (!strcasecmp(v->name, "alwaysauthreject")) {
         sip_cfg.alwaysauthreject = ast_true(v->value);
      } else if (!strcasecmp(v->name, "auth_options_requests")) {
         if (ast_true(v->value)) {
            sip_cfg.auth_options_requests = 1;
         }
      } else if (!strcasecmp(v->name, "auth_message_requests")) {
         sip_cfg.auth_message_requests = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "accept_outofcall_message")) {
         sip_cfg.accept_outofcall_message = ast_true(v->value) ? 1 : 0;
      } else if (!strcasecmp(v->name, "outofcall_message_context")) {
         ast_copy_string(sip_cfg.messagecontext, v->value, sizeof(sip_cfg.messagecontext));
      } else if (!strcasecmp(v->name, "mohinterpret")) {
         ast_copy_string(default_mohinterpret, v->value, sizeof(default_mohinterpret));
      } else if (!strcasecmp(v->name, "mohsuggest")) {
         ast_copy_string(default_mohsuggest, v->value, sizeof(default_mohsuggest));
      } else if (!strcasecmp(v->name, "tonezone")) {
         struct ast_tone_zone *new_zone;
         if (!(new_zone = ast_get_indication_zone(v->value))) {
            ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone in [general] at line %d. Check indications.conf for available country codes.\n", v->value, v->lineno);
         } else {
            ast_tone_zone_unref(new_zone);
            ast_copy_string(default_zone, v->value, sizeof(default_zone));
         }
      } else if (!strcasecmp(v->name, "language")) {
         ast_copy_string(default_language, v->value, sizeof(default_language));
      } else if (!strcasecmp(v->name, "regcontext")) {
         ast_copy_string(newcontexts, v->value, sizeof(newcontexts));
         stringp = newcontexts;
         /* Let's remove any contexts that are no longer defined in regcontext */
         cleanup_stale_contexts(stringp, oldregcontext);
         /* Create contexts if they don't exist already */
         while ((context = strsep(&stringp, "&"))) {
            ast_copy_string(used_context, context, sizeof(used_context));
            ast_context_find_or_create(NULL, NULL, context, "SIP");
         }
         ast_copy_string(sip_cfg.regcontext, v->value, sizeof(sip_cfg.regcontext));
      } else if (!strcasecmp(v->name, "regextenonqualify")) {
         sip_cfg.regextenonqualify = ast_true(v->value);
      } else if (!strcasecmp(v->name, "legacy_useroption_parsing")) {
         sip_cfg.legacy_useroption_parsing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "send_diversion")) {
         sip_cfg.send_diversion = ast_true(v->value);
      } else if (!strcasecmp(v->name, "callerid")) {
         ast_copy_string(default_callerid, v->value, sizeof(default_callerid));
      } else if (!strcasecmp(v->name, "mwi_from")) {
         ast_copy_string(default_mwi_from, v->value, sizeof(default_mwi_from));
      } else if (!strcasecmp(v->name, "fromdomain")) {
         char *fromdomainport;
         ast_copy_string(default_fromdomain, v->value, sizeof(default_fromdomain));
         if ((fromdomainport = strchr(default_fromdomain, ':'))) {
            *fromdomainport++ = '\0';
            if (!(default_fromdomainport = port_str2int(fromdomainport, 0))) {
               ast_log(LOG_NOTICE, "'%s' is not a valid port number for fromdomain.\n",fromdomainport);
            }
         } else {
            default_fromdomainport = STANDARD_SIP_PORT;
         }
      } else if (!strcasecmp(v->name, "outboundproxy")) {
         struct sip_proxy *proxy;
         if (ast_strlen_zero(v->value)) {
            ast_log(LOG_WARNING, "no value given for outbound proxy on line %d of sip.conf\n", v->lineno);
            continue;
         }
         proxy = proxy_from_config(v->value, v->lineno, &sip_cfg.outboundproxy);
         if (!proxy) {
            ast_log(LOG_WARNING, "failure parsing the outbound proxy on line %d of sip.conf.\n", v->lineno);
            continue;
         }
      } else if (!strcasecmp(v->name, "autocreatepeer")) {
         if (!strcasecmp(v->value, "persist")) {
            sip_cfg.autocreatepeer = AUTOPEERS_PERSIST;
         } else {
            sip_cfg.autocreatepeer = ast_true(v->value) ? AUTOPEERS_VOLATILE : AUTOPEERS_DISABLED;
         }
      } else if (!strcasecmp(v->name, "match_auth_username")) {
         global_match_auth_username = ast_true(v->value);
      } else if (!strcasecmp(v->name, "srvlookup")) {
         sip_cfg.srvlookup = ast_true(v->value);
      } else if (!strcasecmp(v->name, "pedantic")) {
         sip_cfg.pedanticsipchecking = ast_true(v->value);
      } else if (!strcasecmp(v->name, "maxexpirey") || !strcasecmp(v->name, "maxexpiry")) {
         max_expiry = atoi(v->value);
         if (max_expiry < 1) {
            max_expiry = DEFAULT_MAX_EXPIRY;
         }
      } else if (!strcasecmp(v->name, "minexpirey") || !strcasecmp(v->name, "minexpiry")) {
         min_expiry = atoi(v->value);
         if (min_expiry < 1) {
            min_expiry = DEFAULT_MIN_EXPIRY;
         }
      } else if (!strcasecmp(v->name, "defaultexpiry") || !strcasecmp(v->name, "defaultexpirey")) {
         default_expiry = atoi(v->value);
         if (default_expiry < 1) {
            default_expiry = DEFAULT_DEFAULT_EXPIRY;
         }
      } else if (!strcasecmp(v->name, "submaxexpirey") || !strcasecmp(v->name, "submaxexpiry")) {
         max_subexpiry = atoi(v->value);
         if (max_subexpiry < 1) {
            max_subexpiry = DEFAULT_MAX_EXPIRY;
         }
         max_subexpiry_set = 1;
      } else if (!strcasecmp(v->name, "subminexpirey") || !strcasecmp(v->name, "subminexpiry")) {
         min_subexpiry = atoi(v->value);
         if (min_subexpiry < 1) {
            min_subexpiry = DEFAULT_MIN_EXPIRY;
         }
         min_subexpiry_set = 1;
      } else if (!strcasecmp(v->name, "mwiexpiry") || !strcasecmp(v->name, "mwiexpirey")) {
         mwi_expiry = atoi(v->value);
         if (mwi_expiry < 1) {
            mwi_expiry = DEFAULT_MWI_EXPIRY;
         }
      } else if (!strcasecmp(v->name, "tcpauthtimeout")) {
         if (ast_parse_arg(v->value, PARSE_INT32|PARSE_DEFAULT|PARSE_IN_RANGE,
                 &authtimeout, DEFAULT_AUTHTIMEOUT, 1, INT_MAX)) {
            ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
               v->name, v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "tcpauthlimit")) {
         if (ast_parse_arg(v->value, PARSE_INT32|PARSE_DEFAULT|PARSE_IN_RANGE,
                 &authlimit, DEFAULT_AUTHLIMIT, 1, INT_MAX)) {
            ast_log(LOG_WARNING, "Invalid %s '%s' at line %d of %s\n",
               v->name, v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "sipdebug")) {
         if (ast_true(v->value))
            sipdebug |= sip_debug_config;
      } else if (!strcasecmp(v->name, "dumphistory")) {
         dumphistory = ast_true(v->value);
      } else if (!strcasecmp(v->name, "recordhistory")) {
         recordhistory = ast_true(v->value);
      } else if (!strcasecmp(v->name, "registertimeout")) {
         global_reg_timeout = atoi(v->value);
         if (global_reg_timeout < 1) {
            global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;
         }
      } else if (!strcasecmp(v->name, "registerattempts")) {
         global_regattempts_max = atoi(v->value);
      } else if (!strcasecmp(v->name, "register_retry_403")) {
         global_reg_retry_403 = ast_true(v->value);
      } else if (!strcasecmp(v->name, "bindaddr") || !strcasecmp(v->name, "udpbindaddr")) {
         if (ast_parse_arg(v->value, PARSE_ADDR, &bindaddr)) {
            ast_log(LOG_WARNING, "Invalid address: %s\n", v->value);
         }
      } else if (!strcasecmp(v->name, "localnet")) {
         struct ast_ha *na;
         int ha_error = 0;

         if (!(na = ast_append_ha("d", v->value, localaddr, &ha_error))) {
            ast_log(LOG_WARNING, "Invalid localnet value: %s\n", v->value);
         } else {
            localaddr = na;
         }
         if (ha_error) {
            ast_log(LOG_ERROR, "Bad localnet configuration value line %d : %s\n", v->lineno, v->value);
         }
      } else if (!strcasecmp(v->name, "media_address")) {
         if (ast_parse_arg(v->value, PARSE_ADDR, &media_address))
            ast_log(LOG_WARNING, "Invalid address for media_address keyword: %s\n", v->value);
      } else if (!strcasecmp(v->name, "externaddr") || !strcasecmp(v->name, "externip")) {
         if (ast_parse_arg(v->value, PARSE_ADDR, &externaddr)) {
            ast_log(LOG_WARNING,
               "Invalid address for externaddr keyword: %s\n",
               v->value);
         }
         externexpire = 0;
      } else if (!strcasecmp(v->name, "externhost")) {
         ast_copy_string(externhost, v->value, sizeof(externhost));
         if (ast_sockaddr_resolve_first(&externaddr, externhost, 0)) {
            ast_log(LOG_WARNING, "Invalid address for externhost keyword: %s\n", externhost);
         }
         externexpire = time(NULL);
      } else if (!strcasecmp(v->name, "externrefresh")) {
         if (sscanf(v->value, "%30d", &externrefresh) != 1) {
            ast_log(LOG_WARNING, "Invalid externrefresh value '%s', must be an integer >0 at line %d\n", v->value, v->lineno);
            externrefresh = 10;
         }
      } else if (!strcasecmp(v->name, "externtcpport")) {
         if (!(externtcpport = port_str2int(v->value, 0))) {
            ast_log(LOG_WARNING, "Invalid externtcpport value, must be a positive integer between 1 and 65535 at line %d\n", v->lineno);
            externtcpport = 0;
         }
      } else if (!strcasecmp(v->name, "externtlsport")) {
         if (!(externtlsport = port_str2int(v->value, STANDARD_TLS_PORT))) {
            ast_log(LOG_WARNING, "Invalid externtlsport value, must be a positive integer between 1 and 65535 at line %d\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "allow")) {
         int error =  ast_parse_allow_disallow(&default_prefs, sip_cfg.caps, v->value, TRUE);
         if (error) {
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
         }
      } else if (!strcasecmp(v->name, "disallow")) {
         int error =  ast_parse_allow_disallow(&default_prefs, sip_cfg.caps, v->value, FALSE);
         if (error) {
            ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
         }
      } else if (!strcasecmp(v->name, "preferred_codec_only")) {
         ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
      } else if (!strcasecmp(v->name, "autoframing")) {
         global_autoframing = ast_true(v->value);
      } else if (!strcasecmp(v->name, "allowexternaldomains")) {
         sip_cfg.allow_external_domains = ast_true(v->value);
      } else if (!strcasecmp(v->name, "autodomain")) {
         auto_sip_domains = ast_true(v->value);
      } else if (!strcasecmp(v->name, "domain")) {
         char *domain = ast_strdupa(v->value);
         char *cntx = strchr(domain, ',');

         if (cntx) {
            *cntx++ = '\0';
         }

         if (ast_strlen_zero(cntx)) {
            ast_debug(1, "No context specified at line %d for domain '%s'\n", v->lineno, domain);
         }
         if (ast_strlen_zero(domain)) {
            ast_log(LOG_WARNING, "Empty domain specified at line %d\n", v->lineno);
         } else {
            add_sip_domain(ast_strip(domain), SIP_DOMAIN_CONFIG, cntx ? ast_strip(cntx) : "");
         }
      } else if (!strcasecmp(v->name, "register")) {
         if (sip_register(v->value, v->lineno) == 0) {
            registry_count++;
         }
      } else if (!strcasecmp(v->name, "mwi")) {
         sip_subscribe_mwi(v->value, v->lineno);
      } else if (!strcasecmp(v->name, "tos_sip")) {
         if (ast_str2tos(v->value, &global_tos_sip)) {
            ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "tos_audio")) {
         if (ast_str2tos(v->value, &global_tos_audio)) {
            ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "tos_video")) {
         if (ast_str2tos(v->value, &global_tos_video)) {
            ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "tos_text")) {
         if (ast_str2tos(v->value, &global_tos_text)) {
            ast_log(LOG_WARNING, "Invalid tos_text value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "cos_sip")) {
         if (ast_str2cos(v->value, &global_cos_sip)) {
            ast_log(LOG_WARNING, "Invalid cos_sip value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "cos_audio")) {
         if (ast_str2cos(v->value, &global_cos_audio)) {
            ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "cos_video")) {
         if (ast_str2cos(v->value, &global_cos_video)) {
            ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "cos_text")) {
         if (ast_str2cos(v->value, &global_cos_text)) {
            ast_log(LOG_WARNING, "Invalid cos_text value at line %d, refer to QoS documentation\n", v->lineno);
         }
      } else if (!strcasecmp(v->name, "bindport")) {
         if (sscanf(v->value, "%5d", &bindport) != 1) {
            ast_log(LOG_WARNING, "Invalid port number '%s' at line %d of %s\n", v->value, v->lineno, config);
         }
      } else if (!strcasecmp(v->name, "qualify")) {
         if (!strcasecmp(v->value, "no")) {
            default_qualify = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            default_qualify = DEFAULT_MAXMS;
         } else if (sscanf(v->value, "%30d", &default_qualify) != 1) {
            ast_log(LOG_WARNING, "Qualification default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
            default_qualify = 0;
         }
      } else if (!strcasecmp(v->name, "keepalive")) {
         if (!strcasecmp(v->value, "no")) {
            default_keepalive = 0;
         } else if (!strcasecmp(v->value, "yes")) {
            default_keepalive = DEFAULT_KEEPALIVE_INTERVAL;
         } else if (sscanf(v->value, "%30d", &default_keepalive) != 1) {
            ast_log(LOG_WARNING, "Keep alive default should be 'yes', 'no', or a number of milliseconds at line %d of sip.conf\n", v->lineno);
            default_keepalive = 0;
         }
      } else if (!strcasecmp(v->name, "qualifyfreq")) {
         int i;
         if (sscanf(v->value, "%30d", &i) == 1) {
            global_qualifyfreq = i * 1000;
         } else {
            ast_log(LOG_WARNING, "Invalid qualifyfreq number '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_qualifyfreq = DEFAULT_QUALIFYFREQ;
         }
      } else if (!strcasecmp(v->name, "callevents")) {
         sip_cfg.callevents = ast_true(v->value);
      } else if (!strcasecmp(v->name, "authfailureevents")) {
         global_authfailureevents = ast_true(v->value);
      } else if (!strcasecmp(v->name, "maxcallbitrate")) {
         default_maxcallbitrate = atoi(v->value);
         if (default_maxcallbitrate < 0) {
            default_maxcallbitrate = DEFAULT_MAX_CALL_BITRATE;
         }
      } else if (!strcasecmp(v->name, "matchexternaddrlocally") || !strcasecmp(v->name, "matchexterniplocally")) {
         sip_cfg.matchexternaddrlocally = ast_true(v->value);
      } else if (!strcasecmp(v->name, "session-timers")) {
         int i = (int) str2stmode(v->value);
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-timers '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_st_mode = SESSION_TIMER_MODE_ACCEPT;
         } else {
            global_st_mode = i;
         }
      } else if (!strcasecmp(v->name, "session-expires")) {
         if (sscanf(v->value, "%30d", &global_max_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-expires '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_max_se = DEFAULT_MAX_SE;
         }
      } else if (!strcasecmp(v->name, "session-minse")) {
         if (sscanf(v->value, "%30d", &global_min_se) != 1) {
            ast_log(LOG_WARNING, "Invalid session-minse '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_min_se = DEFAULT_MIN_SE;
         }
         if (global_min_se < DEFAULT_MIN_SE) {
            ast_log(LOG_WARNING, "session-minse '%s' at line %d of %s is not allowed to be < %d secs\n", v->value, v->lineno, config, DEFAULT_MIN_SE);
            global_min_se = DEFAULT_MIN_SE;
         }
      } else if (!strcasecmp(v->name, "session-refresher")) {
         int i = (int) str2strefresherparam(v->value);
         if (i < 0) {
            ast_log(LOG_WARNING, "Invalid session-refresher '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_st_refresher = SESSION_TIMER_REFRESHER_PARAM_UAS;
         } else {
            global_st_refresher = i;
         }
      } else if (!strcasecmp(v->name, "storesipcause")) {
         global_store_sip_cause = ast_true(v->value);
         if (global_store_sip_cause) {
            ast_log(LOG_WARNING, "Usage of SIP_CAUSE is deprecated.  Please use HANGUPCAUSE instead.\n");
         }
      } else if (!strcasecmp(v->name, "qualifygap")) {
         if (sscanf(v->value, "%30d", &global_qualify_gap) != 1) {
            ast_log(LOG_WARNING, "Invalid qualifygap '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_qualify_gap = DEFAULT_QUALIFY_GAP;
         }
      } else if (!strcasecmp(v->name, "qualifypeers")) {
         if (sscanf(v->value, "%30d", &global_qualify_peers) != 1) {
            ast_log(LOG_WARNING, "Invalid pokepeers '%s' at line %d of %s\n", v->value, v->lineno, config);
            global_qualify_peers = DEFAULT_QUALIFY_PEERS;
         }
      } else if (!strcasecmp(v->name, "disallowed_methods")) {
         char *disallow = ast_strdupa(v->value);
         mark_parsed_methods(&sip_cfg.disallowed_methods, disallow);
      } else if (!strcasecmp(v->name, "shrinkcallerid")) {
         if (ast_true(v->value)) {
            global_shrinkcallerid = 1;
         } else if (ast_false(v->value)) {
            global_shrinkcallerid = 0;
         } else {
            ast_log(LOG_WARNING, "shrinkcallerid value %s is not valid at line %d.\n", v->value, v->lineno);
         }
      } else if (!strcasecmp(v->name, "use_q850_reason")) {
         ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_Q850_REASON);
      } else if (!strcasecmp(v->name, "maxforwards")) {
         if (sscanf(v->value, "%30d", &sip_cfg.default_max_forwards) != 1
            || sip_cfg.default_max_forwards < 1 || 255 < sip_cfg.default_max_forwards) {
            ast_log(LOG_WARNING, "'%s' is not a valid maxforwards value at line %d.  Using default.\n", v->value, v->lineno);
            sip_cfg.default_max_forwards = DEFAULT_MAX_FORWARDS;
         }
      } else if (!strcasecmp(v->name, "subscribe_network_change_event")) {
         if (ast_true(v->value)) {
            subscribe_network_change = 1;
         } else if (ast_false(v->value)) {
            subscribe_network_change = 0;
         } else {
            ast_log(LOG_WARNING, "subscribe_network_change_event value %s is not valid at line %d.\n", v->value, v->lineno);
         }
      } else if (!strcasecmp(v->name, "snom_aoc_enabled")) {
         ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_SNOM_AOC);
      } else if (!strcasecmp(v->name, "icesupport")) {
         ast_set2_flag(&global_flags[2], ast_true(v->value), SIP_PAGE3_ICE_SUPPORT);
      } else if (!strcasecmp(v->name, "parkinglot")) {
         ast_copy_string(default_parkinglot, v->value, sizeof(default_parkinglot));
      } else if (!strcasecmp(v->name, "refer_addheaders")) {
         global_refer_addheaders = ast_true(v->value);
      }
   }

   /* Override global defaults if setting found in general section */
   ast_copy_flags(&global_flags[0], &setflags[0], mask[0].flags);
   ast_copy_flags(&global_flags[1], &setflags[1], mask[1].flags);
   ast_copy_flags(&global_flags[2], &setflags[2], mask[2].flags);

   /* For backwards compatibility the corresponding registration timer value is used if subscription timer value isn't set by configuration */
   if (!min_subexpiry_set) {
      min_subexpiry = min_expiry;
   }
   if (!max_subexpiry_set) {
      max_subexpiry = max_expiry;
   }

   if (reason != CHANNEL_MODULE_LOAD && sip_cfg.autocreatepeer != AUTOPEERS_PERSIST) {
      ao2_t_callback(peers, OBJ_NODATA, peer_markall_autopeers_func, NULL, "callback to mark autopeers for destruction");
   }

   if (subscribe_network_change) {
      network_change_event_subscribe();
   } else {
      network_change_event_unsubscribe();
   }

   if (global_t1 < global_t1min) {
      ast_log(LOG_WARNING, "'t1min' (%d) cannot be greater than 't1timer' (%d).  Resetting 't1timer' to the value of 't1min'\n", global_t1min, global_t1);
      global_t1 = global_t1min;
   }

   if (global_timer_b < global_t1 * 64) {
      if (timerb_set && timert1_set) {
         ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", global_timer_b, global_t1);
      } else if (timerb_set) {
         if ((global_t1 = global_timer_b / 64) < global_t1min) {
            ast_log(LOG_WARNING, "Timer B has been set lower than recommended (%d < 64 * timert1=%d). (RFC 3261, 17.1.1.2)\n", global_timer_b, global_t1);
            global_t1 = global_t1min;
            global_timer_b = global_t1 * 64;
         }
      } else {
         global_timer_b = global_t1 * 64;
      }
   }
   if (!sip_cfg.allow_external_domains && AST_LIST_EMPTY(&domain_list)) {
      ast_log(LOG_WARNING, "To disallow external domains, you need to configure local SIP domains.\n");
      sip_cfg.allow_external_domains = 1;
   }
   /* If not or badly configured, set default transports */
   if (!sip_cfg.tcp_enabled && (default_transports & SIP_TRANSPORT_TCP)) {
      ast_log(LOG_WARNING, "Cannot use 'tcp' transport with tcpenable=no. Removing from available transports.\n");
      default_primary_transport &= ~SIP_TRANSPORT_TCP;
      default_transports &= ~SIP_TRANSPORT_TCP;
   }
   if (!default_tls_cfg.enabled && (default_transports & SIP_TRANSPORT_TLS)) {
      ast_log(LOG_WARNING, "Cannot use 'tls' transport with tlsenable=no. Removing from available transports.\n");
      default_primary_transport &= ~SIP_TRANSPORT_TLS;
      default_transports &= ~SIP_TRANSPORT_TLS;
   }
   if (!default_transports) {
      ast_log(LOG_WARNING, "No valid transports available, falling back to 'udp'.\n");
      default_transports = default_primary_transport = SIP_TRANSPORT_UDP;
   } else if (!default_primary_transport) {
      ast_log(LOG_WARNING, "No valid default transport. Selecting 'udp' as default.\n");
      default_primary_transport = SIP_TRANSPORT_UDP;
   }

   /* Build list of authentication to various SIP realms, i.e. service providers */
   for (v = ast_variable_browse(cfg, "authentication"); v ; v = v->next) {
      /* Format for authentication is auth = username:password@realm */
      if (!strcasecmp(v->name, "auth")) {
         add_realm_authentication(&authl, v->value, v->lineno);
      }
   }

   if (bindport) {
      if (ast_sockaddr_port(&bindaddr)) {
         ast_log(LOG_WARNING, "bindport is also specified in bindaddr. "
            "Using %d.\n", bindport);
      }
      ast_sockaddr_set_port(&bindaddr, bindport);
   }

   if (!ast_sockaddr_port(&bindaddr)) {
      ast_sockaddr_set_port(&bindaddr, STANDARD_SIP_PORT);
   }

   /* Set UDP address and open socket */
   ast_sockaddr_copy(&internip, &bindaddr);
   if (ast_find_ourip(&internip, &bindaddr, 0)) {
      ast_log(LOG_WARNING, "Unable to get own IP address, SIP disabled\n");
      ast_config_destroy(cfg);
      return 0;
   }

   ast_mutex_lock(&netlock);
   if ((sipsock > -1) && (ast_sockaddr_cmp(&old_bindaddr, &bindaddr))) {
      close(sipsock);
      sipsock = -1;
   }
   if (sipsock < 0) {
      sipsock = socket(ast_sockaddr_is_ipv6(&bindaddr) ?
             AF_INET6 : AF_INET, SOCK_DGRAM, 0);
      if (sipsock < 0) {
         ast_log(LOG_WARNING, "Unable to create SIP socket: %s\n", strerror(errno));
         ast_config_destroy(cfg);
         ast_mutex_unlock(&netlock);
         return -1;
      } else {
         /* Allow SIP clients on the same host to access us: */
         const int reuseFlag = 1;

         setsockopt(sipsock, SOL_SOCKET, SO_REUSEADDR,
               (const char*)&reuseFlag,
               sizeof reuseFlag);

         ast_enable_packet_fragmentation(sipsock);

         if (ast_bind(sipsock, &bindaddr) < 0) {
            ast_log(LOG_WARNING, "Failed to bind to %s: %s\n",
               ast_sockaddr_stringify(&bindaddr), strerror(errno));
            close(sipsock);
            sipsock = -1;
         } else {
            ast_verb(2, "SIP Listening on %s\n", ast_sockaddr_stringify(&bindaddr));
            ast_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
         }
      }
   } else {
      ast_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
   }
   ast_mutex_unlock(&netlock);

   /* Start TCP server */
   if (sip_cfg.tcp_enabled) {
      if (ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {
         ast_sockaddr_copy(&sip_tcp_desc.local_address, &bindaddr);
      }
      if (!ast_sockaddr_port(&sip_tcp_desc.local_address)) {
         ast_sockaddr_set_port(&sip_tcp_desc.local_address, STANDARD_SIP_PORT);
      }
   } else {
      ast_sockaddr_setnull(&sip_tcp_desc.local_address);
   }
   ast_tcptls_server_start(&sip_tcp_desc);
   if (sip_cfg.tcp_enabled && sip_tcp_desc.accept_fd == -1) {
      /* TCP server start failed. Tell the admin */
      ast_log(LOG_ERROR, "SIP TCP Server start failed. Not listening on TCP socket.\n");
   } else {
      ast_debug(2, "SIP TCP server started\n");
   }

   /* Start TLS server if needed */
   memcpy(sip_tls_desc.tls_cfg, &default_tls_cfg, sizeof(default_tls_cfg));

   if (ast_ssl_setup(sip_tls_desc.tls_cfg)) {
      if (ast_sockaddr_isnull(&sip_tls_desc.local_address)) {
         ast_sockaddr_copy(&sip_tls_desc.local_address, &bindaddr);
         ast_sockaddr_set_port(&sip_tls_desc.local_address,
                     STANDARD_TLS_PORT);
      }
      if (!ast_sockaddr_port(&sip_tls_desc.local_address)) {
         ast_sockaddr_set_port(&sip_tls_desc.local_address,
                     STANDARD_TLS_PORT);
      }
      ast_tcptls_server_start(&sip_tls_desc);
      if (default_tls_cfg.enabled && sip_tls_desc.accept_fd == -1) {
         ast_log(LOG_ERROR, "TLS Server start failed. Not listening on TLS socket.\n");
         sip_tls_desc.tls_cfg = NULL;
      }
   } else if (sip_tls_desc.tls_cfg->enabled) {
      sip_tls_desc.tls_cfg = NULL;
      ast_log(LOG_WARNING, "SIP TLS server did not load because of errors.\n");
   }

   if (ucfg) {
      struct ast_variable *gen;
      int genhassip, genregistersip;
      const char *hassip, *registersip;
      
      genhassip = ast_true(ast_variable_retrieve(ucfg, "general", "hassip"));
      genregistersip = ast_true(ast_variable_retrieve(ucfg, "general", "registersip"));
      gen = ast_variable_browse(ucfg, "general");
      cat = ast_category_browse(ucfg, NULL);
      while (cat) {
         if (strcasecmp(cat, "general")) {
            hassip = ast_variable_retrieve(ucfg, cat, "hassip");
            registersip = ast_variable_retrieve(ucfg, cat, "registersip");
            if (ast_true(hassip) || (!hassip && genhassip)) {
               peer = build_peer(cat, gen, ast_variable_browse(ucfg, cat), 0, 0);
               if (peer) {
                  /* user.conf entries are always of type friend */
                  peer->type = SIP_TYPE_USER | SIP_TYPE_PEER;
                  ao2_t_link(peers, peer, "link peer into peer table");
                  if ((peer->type & SIP_TYPE_PEER) && !ast_sockaddr_isnull(&peer->addr)) {
                     ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
                  }
                  
                  sip_unref_peer(peer, "sip_unref_peer: from reload_config");
                  peer_count++;
               }
            }
            if (ast_true(registersip) || (!registersip && genregistersip)) {
               char tmp[256];
               const char *host = ast_variable_retrieve(ucfg, cat, "host");
               const char *username = ast_variable_retrieve(ucfg, cat, "username");
               const char *secret = ast_variable_retrieve(ucfg, cat, "secret");
               const char *contact = ast_variable_retrieve(ucfg, cat, "contact");
               const char *authuser = ast_variable_retrieve(ucfg, cat, "authuser");
               if (!host) {
                  host = ast_variable_retrieve(ucfg, "general", "host");
               }
               if (!username) {
                  username = ast_variable_retrieve(ucfg, "general", "username");
               }
               if (!secret) {
                  secret = ast_variable_retrieve(ucfg, "general", "secret");
               }
               if (!contact) {
                  contact = "s";
               }
               if (!ast_strlen_zero(username) && !ast_strlen_zero(host)) {
                  if (!ast_strlen_zero(secret)) {
                     if (!ast_strlen_zero(authuser)) {
                        snprintf(tmp, sizeof(tmp), "%s?%s:%s:%s@%s/%s", cat, username, secret, authuser, host, contact);
                     } else {
                        snprintf(tmp, sizeof(tmp), "%s?%s:%s@%s/%s", cat, username, secret, host, contact);
                     }
                  } else if (!ast_strlen_zero(authuser)) {
                     snprintf(tmp, sizeof(tmp), "%s?%s::%s@%s/%s", cat, username, authuser, host, contact);
                  } else {
                     snprintf(tmp, sizeof(tmp), "%s?%s@%s/%s", cat, username, host, contact);
                  }
                  if (sip_register(tmp, 0) == 0) {
                     registry_count++;
                  }
               }
            }
         }
         cat = ast_category_browse(ucfg, cat);
      }
      ast_config_destroy(ucfg);
   }

   /* Load peers, users and friends */
   cat = NULL;
   while ( (cat = ast_category_browse(cfg, cat)) ) {
      const char *utype;
      if (!strcasecmp(cat, "general") || !strcasecmp(cat, "authentication"))
         continue;
      utype = ast_variable_retrieve(cfg, cat, "type");
      if (!utype) {
         ast_log(LOG_WARNING, "Section '%s' lacks type\n", cat);
         continue;
      } else {
         if (!strcasecmp(utype, "user")) {
            ;
         } else if (!strcasecmp(utype, "friend")) {
            ;
         } else if (!strcasecmp(utype, "peer")) {
            ;
         } else {
            ast_log(LOG_WARNING, "Unknown type '%s' for '%s' in %s\n", utype, cat, "sip.conf");
            continue;
         }
         peer = build_peer(cat, ast_variable_browse(cfg, cat), NULL, 0, 0);
         if (peer) {
            display_nat_warning(cat, reason, &peer->flags[0]);
            ao2_t_link(peers, peer, "link peer into peers table");
            if ((peer->type & SIP_TYPE_PEER) && !ast_sockaddr_isnull(&peer->addr)) {
               ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
            }
            sip_unref_peer(peer, "unref the result of the build_peer call. Now, the links from the tables are the only ones left.");
            peer_count++;
         }
      }
   }

   /* Add default domains - host name, IP address and IP:port
    * Only do this if user added any sip domain with "localdomains"
    * In order to *not* break backwards compatibility
    *    Some phones address us at IP only, some with additional port number
    */
   if (auto_sip_domains) {
      char temp[MAXHOSTNAMELEN];

      /* First our default IP address */
      if (!ast_sockaddr_isnull(&bindaddr) && !ast_sockaddr_is_any(&bindaddr)) {
         add_sip_domain(ast_sockaddr_stringify_addr(&bindaddr),
                   SIP_DOMAIN_AUTO, NULL);
      } else if (!ast_sockaddr_isnull(&internip) && !ast_sockaddr_is_any(&internip)) {
      /* Our internal IP address, if configured */
         add_sip_domain(ast_sockaddr_stringify_addr(&internip),
                   SIP_DOMAIN_AUTO, NULL);
      } else {
         ast_log(LOG_NOTICE, "Can't add wildcard IP address to domain list, please add IP address to domain manually.\n");
      }

      /* If TCP is running on a different IP than UDP, then add it too */
      if (!ast_sockaddr_isnull(&sip_tcp_desc.local_address) &&
          !ast_sockaddr_cmp(&bindaddr, &sip_tcp_desc.local_address)) {
         add_sip_domain(ast_sockaddr_stringify_addr(&sip_tcp_desc.local_address),
                   SIP_DOMAIN_AUTO, NULL);
      }

      /* If TLS is running on a different IP than UDP and TCP, then add that too */
      if (!ast_sockaddr_isnull(&sip_tls_desc.local_address) &&
          !ast_sockaddr_cmp(&bindaddr, &sip_tls_desc.local_address) &&
          !ast_sockaddr_cmp(&sip_tcp_desc.local_address,
                  &sip_tls_desc.local_address)) {
         add_sip_domain(ast_sockaddr_stringify_addr(&sip_tcp_desc.local_address),
                   SIP_DOMAIN_AUTO, NULL);
      }

      /* Our extern IP address, if configured */
      if (!ast_sockaddr_isnull(&externaddr)) {
         add_sip_domain(ast_sockaddr_stringify_addr(&externaddr),
                   SIP_DOMAIN_AUTO, NULL);
      }

      /* Extern host name (NAT traversal support) */
      if (!ast_strlen_zero(externhost)) {
         add_sip_domain(externhost, SIP_DOMAIN_AUTO, NULL);
      }
      
      /* Our host name */
      if (!gethostname(temp, sizeof(temp))) {
         add_sip_domain(temp, SIP_DOMAIN_AUTO, NULL);
      }
   }

   /* Release configuration from memory */
   ast_config_destroy(cfg);

   register_realtime_peers_with_callbackextens();

   /* Load the list of manual NOTIFY types to support */
   if (notify_types) {
      ast_config_destroy(notify_types);
   }
   if ((notify_types = ast_config_load(notify_config, config_flags)) == CONFIG_STATUS_FILEINVALID) {
      ast_log(LOG_ERROR, "Contents of %s are invalid and cannot be parsed.\n", notify_config);
      notify_types = NULL;
   }

   /* Done, tell the manager */
   manager_event(EVENT_FLAG_SYSTEM, "ChannelReload", "ChannelType: SIP\r\nReloadReason: %s\r\nRegistry_Count: %d\r\nPeer_Count: %d\r\n", channelreloadreason2txt(reason), registry_count, peer_count);
   run_end = time(0);
   ast_debug(4, "SIP reload_config done...Runtime= %d sec\n", (int)(run_end-run_start));

   /* If an ACL change subscription is needed and doesn't exist, we need one. */
   if (acl_change_subscription_needed) {
      acl_change_event_subscribe();
   }

   return 0;
}

static char * remove_uri_parameters

(

char *

uri

)

[static]

Definition at line 13763 of file chan_sip.c.

Referenced by extract_uri(), parse_moved_contact(), register_verify(), reqprep(), and transmit_state_notify().

{
   char *atsign;
   atsign = strchr(uri, '@'); /* First, locate the at sign */
   if (!atsign) {
      atsign = uri;  /* Ok hostname only, let's stick with the rest */
   }
   atsign = strchr(atsign, ';'); /* Locate semi colon */
   if (atsign)
      *atsign = '\0';   /* Kill at the semi colon */
   return uri;
}

static int reply_digest	(	struct sip_pvt *	p,
		struct sip_request *	req,
		char *	header,
		int	sipmethod,
		char *	digest,
		int	digest_len
	)		[static]

reply to authentication for outbound registrations

Returns:

Returns -1 if we have no auth

Note:

This is used for register= servers in sip.conf, SIP proxies we register with for receiving calls from.

Definition at line 21843 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_skip_blanks(), ast_string_field_ptr_set, ast_string_field_set, ast_strlen_zero(), build_reply_digest(), LOG_WARNING, and sip_get_header().

Referenced by do_message_auth(), do_proxy_auth(), and do_register_auth().

{
   char tmp[512];
   char *c;
   char oldnonce[256];

   /* table of recognised keywords, and places where they should be copied */
   const struct x {
      const char *key;
      const ast_string_field *field;
   } *i, keys[] = {
      { "realm=", &p->realm },
      { "nonce=", &p->nonce },
      { "opaque=", &p->opaque },
      { "qop=", &p->qop },
      { "domain=", &p->domain },
      { NULL, 0 },
   };

   ast_copy_string(tmp, sip_get_header(req, header), sizeof(tmp));
   if (ast_strlen_zero(tmp))
      return -1;
   if (strncasecmp(tmp, "Digest ", strlen("Digest "))) {
      ast_log(LOG_WARNING, "missing Digest.\n");
      return -1;
   }
   c = tmp + strlen("Digest ");
   ast_copy_string(oldnonce, p->nonce, sizeof(oldnonce));
   while (c && *(c = ast_skip_blanks(c))) {  /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         char *src, *separator;
         if (strncasecmp(c, i->key, strlen(i->key)) != 0)
            continue;
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') {
            src = ++c;
            separator = "\"";
         } else {
            src = c;
            separator = ",";
         }
         strsep(&c, separator); /* clear separator and move ptr */
         ast_string_field_ptr_set(p, i->field, src);
         break;
      }
      if (i->key == NULL) /* not found, try ',' */
         strsep(&c, ",");
   }
   /* Reset nonce count */
   if (strcmp(p->nonce, oldnonce))
      p->noncecount = 0;

   /* Save auth data for following registrations */
   if (p->registry) {
      struct sip_registry *r = p->registry;

      if (strcmp(r->nonce, p->nonce)) {
         ast_string_field_set(r, realm, p->realm);
         ast_string_field_set(r, nonce, p->nonce);
         ast_string_field_set(r, authdomain, p->domain);
         ast_string_field_set(r, opaque, p->opaque);
         ast_string_field_set(r, qop, p->qop);
         r->noncecount = 0;
      }
   }
   return build_reply_digest(p, sipmethod, digest, digest_len);
}

static int reqprep	(	struct sip_request *	req,
		struct sip_pvt *	p,
		int	sipmethod,
		uint32_t	seqno,
		int	newbranch
	)		[static]

Initialize a SIP request message (not the initial one in a dialog)

< Strict routing flag

Definition at line 11888 of file chan_sip.c.

References add_header(), add_max_forwards(), add_route(), ast_copy_string(), ast_debug, ast_random(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_via(), copy_header(), FALSE, get_in_brackets(), init_req(), remove_uri_parameters(), set_destination(), sip_get_header(), sip_methods, st_get_se(), cfsip_methods::text, text, TRUE, and url.

Referenced by sipinfo_send(), transmit_cc_notify(), transmit_info_with_aoc(), transmit_info_with_digit(), transmit_info_with_vidupdate(), transmit_invite(), transmit_message(), transmit_notify_with_sipfrag(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), transmit_state_notify(), and update_connectedline().

{
   struct sip_request *orig = &p->initreq;
   char stripped[80];
   char tmp[80];
   char newto[256];
   const char *c;
   const char *ot, *of;
   int is_strict = FALSE;     /*!< Strict routing flag */
   int is_outbound = ast_test_flag(&p->flags[0], SIP_OUTGOING);   /* Session direction */

   snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);

   if (!seqno) {
      p->ocseq++;
      seqno = p->ocseq;
   }

   /* A CANCEL must have the same branch as the INVITE that it is canceling. */
   if (sipmethod == SIP_CANCEL) {
      p->branch = p->invite_branch;
      build_via(p);
   } else if (newbranch && (sipmethod == SIP_INVITE)) {
      p->branch ^= ast_random();
      p->invite_branch = p->branch;
      build_via(p);
   } else if (newbranch) {
      p->branch ^= ast_random();
      build_via(p);
   }

   /* Check for strict or loose router */
   if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop, ";lr") == NULL) {
      is_strict = TRUE;
      if (sipdebug)
         ast_debug(1, "Strict routing enforced for session %s\n", p->callid);
   }

   if (sipmethod == SIP_CANCEL)
      c = REQ_OFFSET_TO_STR(&p->initreq, rlpart2); /* Use original URI */
   else if (sipmethod == SIP_ACK) {
      /* Use URI from Contact: in 200 OK (if INVITE)
      (we only have the contacturi on INVITEs) */
      if (!ast_strlen_zero(p->okcontacturi))
         c = is_strict ? p->route->hop : p->okcontacturi;
      else
         c = REQ_OFFSET_TO_STR(&p->initreq, rlpart2);
   } else if (!ast_strlen_zero(p->okcontacturi))
      c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
   else if (!ast_strlen_zero(p->uri))
      c = p->uri;
   else {
      char *n;
      /* We have no URI, use To: or From:  header as URI (depending on direction) */
      ast_copy_string(stripped, sip_get_header(orig, is_outbound ? "To" : "From"),
            sizeof(stripped));
      n = get_in_brackets(stripped);
      c = remove_uri_parameters(n);
   }  
   init_req(req, sipmethod, c);

   snprintf(tmp, sizeof(tmp), "%u %s", seqno, sip_methods[sipmethod].text);

   add_header(req, "Via", p->via);
   /*
    * Use the learned route set unless this is a CANCEL on an ACK for a non-2xx
    * final response. For a CANCEL or ACK, we have to send to the same destination
    * as the original INVITE.
    */
   if (p->route &&
         !(sipmethod == SIP_CANCEL ||
            (sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)))) {
      set_destination(p, p->route->hop);
      add_route(req, is_strict ? p->route->next : p->route);
   }
   add_max_forwards(p, req);

   ot = sip_get_header(orig, "To");
   of = sip_get_header(orig, "From");

   /* Add tag *unless* this is a CANCEL, in which case we need to send it exactly
      as our original request, including tag (or presumably lack thereof) */
   if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) {
      /* Add the proper tag if we don't have it already.  If they have specified
         their tag, use it.  Otherwise, use our own tag */
      if (is_outbound && !ast_strlen_zero(p->theirtag))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
      else if (!is_outbound)
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
      else
         snprintf(newto, sizeof(newto), "%s", ot);
      ot = newto;
   }

   if (is_outbound) {
      add_header(req, "From", of);
      add_header(req, "To", ot);
   } else {
      add_header(req, "From", ot);
      add_header(req, "To", of);
   }
   /* Do not add Contact for MESSAGE, BYE and Cancel requests */
   if (sipmethod != SIP_BYE && sipmethod != SIP_CANCEL && sipmethod != SIP_MESSAGE)
      add_header(req, "Contact", p->our_contact);

   copy_header(req, orig, "Call-ID");
   add_header(req, "CSeq", tmp);

   if (!ast_strlen_zero(global_useragent))
      add_header(req, "User-Agent", global_useragent);

   if (!ast_strlen_zero(p->url)) {
      add_header(req, "Access-URL", p->url);
      ast_string_field_set(p, url, NULL);
   }

   /* Add Session-Timers related headers if the feature is active for this session.
      An exception to this behavior is the ACK request. Since Asterisk never requires
      session-timers support from a remote end-point (UAS) in an INVITE, it must
      not send 'Require: timer' header in the ACK request.
      This should only be added in the INVITE transactions, not MESSAGE or REFER or other
      in-dialog messages.
   */
   if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE
       && sipmethod == SIP_INVITE) {
      char se_hdr[256];
      snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
         p->stimer->st_ref == SESSION_TIMER_REFRESHER_US ? "uac" : "uas");
      add_header(req, "Session-Expires", se_hdr);
      snprintf(se_hdr, sizeof(se_hdr), "%d", st_get_se(p, FALSE));
      add_header(req, "Min-SE", se_hdr);
   }

   return 0;
}

static int resp_needs_contact	(	const char *	msg,
		enum sipmethod	method
	)		[inline, static]

Test if this response needs a contact header.

Definition at line 11739 of file chan_sip.c.

Referenced by respprep().

                                                                             {
   /* Requirements for Contact header inclusion in responses generated
    * from the header tables found in the following RFCs.  Where the
    * Contact header was marked mandatory (m) or optional (o) this
    * function returns 1.
    *
    * - RFC 3261 (ACK, BYE, CANCEL, INVITE, OPTIONS, REGISTER)
    * - RFC 2976 (INFO)
    * - RFC 3262 (PRACK)
    * - RFC 3265 (SUBSCRIBE, NOTIFY)
    * - RFC 3311 (UPDATE)
    * - RFC 3428 (MESSAGE)
    * - RFC 3515 (REFER)
    * - RFC 3903 (PUBLISH)
    */

   switch (method) {
      /* 1xx, 2xx, 3xx, 485 */
      case SIP_INVITE:
      case SIP_UPDATE:
      case SIP_SUBSCRIBE:
      case SIP_NOTIFY:
         if ((msg[0] >= '1' && msg[0] <= '3') || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 2xx, 3xx, 485 */
      case SIP_REGISTER:
      case SIP_OPTIONS:
         if (msg[0] == '2' || msg[0] == '3' || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 3xx, 485 */
      case SIP_BYE:
      case SIP_PRACK:
      case SIP_MESSAGE:
      case SIP_PUBLISH:
         if (msg[0] == '3' || !strncmp(msg, "485", 3))
            return 1;
         break;

      /* 2xx, 3xx, 4xx, 5xx, 6xx */
      case SIP_REFER:
         if (msg[0] >= '2' && msg[0] <= '6')
            return 1;
         break;

      /* contact will not be included for everything else */
      case SIP_ACK:
      case SIP_CANCEL:
      case SIP_INFO:
      case SIP_PING:
      default:
         return 0;
   }
   return 0;
}

static int respprep	(	struct sip_request *	resp,
		struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)		[static]

Prepare SIP response packet.

Definition at line 11799 of file chan_sip.c.

References add_expires(), add_header(), add_supported(), ast_copy_string(), ast_log(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, copy_all_header(), copy_header(), copy_via_headers(), init_resp(), LOG_WARNING, process_via(), resp_needs_contact(), sip_get_header(), TRUE, and url.

Referenced by __transmit_response(), transmit_response_with_allow(), transmit_response_with_auth(), transmit_response_with_date(), transmit_response_with_minexpires(), transmit_response_with_minse(), transmit_response_with_retry_after(), transmit_response_with_sdp(), transmit_response_with_sip_etag(), transmit_response_with_t38_sdp(), transmit_response_with_unsupported(), update_connectedline(), and update_redirecting().

{
   char newto[256];
   const char *ot;

   init_resp(resp, msg);
   copy_via_headers(p, resp, req, "Via");
   if (msg[0] == '1' || msg[0] == '2')
      copy_all_header(resp, req, "Record-Route");
   copy_header(resp, req, "From");
   ot = sip_get_header(req, "To");
   if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
      /* Add the proper tag if we don't have it already.  If they have specified
         their tag, use it.  Otherwise, use our own tag */
      if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
      else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING))
         snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
      else
         ast_copy_string(newto, ot, sizeof(newto));
      ot = newto;
   }
   add_header(resp, "To", ot);
   copy_header(resp, req, "Call-ID");
   copy_header(resp, req, "CSeq");
   if (!ast_strlen_zero(global_useragent))
      add_header(resp, "Server", global_useragent);
   add_header(resp, "Allow", ALLOWED_METHODS);
   add_supported(p, resp);

   /* If this is an invite, add Session-Timers related headers if the feature is active for this session */
   if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE) {
      char se_hdr[256];
      snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
         p->stimer->st_ref == SESSION_TIMER_REFRESHER_US ? "uas" : "uac");
      add_header(resp, "Session-Expires", se_hdr);
      /* RFC 2048, Section 9
       * If the refresher parameter in the Session-Expires header field in the
       * 2xx response has a value of 'uac', the UAS MUST place a Require
       * header field into the response with the value 'timer'.
       * ...
       * If the refresher parameter in
       * the 2xx response has a value of 'uas' and the Supported header field
       * in the request contained the value 'timer', the UAS SHOULD place a
       * Require header field into the response with the value 'timer'
       */
      if (p->stimer->st_ref == SESSION_TIMER_REFRESHER_THEM ||
            (p->stimer->st_ref == SESSION_TIMER_REFRESHER_US &&
             p->stimer->st_active_peer_ua == TRUE)) {
         resp->reqsipoptions |= SIP_OPT_TIMER;
      }
   }

   if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_PUBLISH)) {
      /* For registration responses, we also need expiry and
         contact info */
      add_expires(resp, p->expiry);
      if (p->expiry) {  /* Only add contact if we have an expiry time */
         char contact[SIPBUFSIZE];
         const char *contact_uri = p->method == SIP_SUBSCRIBE ? p->our_contact : p->fullcontact;
         char *brackets = strchr(contact_uri, '<');
         snprintf(contact, sizeof(contact), "%s%s%s;expires=%d", brackets ? "" : "<", contact_uri, brackets ? "" : ">", p->expiry);
         add_header(resp, "Contact", contact);  /* Not when we unregister */
      }
   } else if (!ast_strlen_zero(p->our_contact) && resp_needs_contact(msg, p->method)) {
      add_header(resp, "Contact", p->our_contact);
   }

   if (!ast_strlen_zero(p->url)) {
      add_header(resp, "Access-URL", p->url);
      ast_string_field_set(p, url, NULL);
   }

   /* default to routing the response to the address where the request
    * came from.  Since we don't have a transport layer, we do this here.
    * The process_via() function will update the port to either the port
    * specified in the via header or the default port later on (per RFC
    * 3261 section 18.2.2).
    */
   p->sa = p->recv;

   if (process_via(p, req)) {
      ast_log(LOG_WARNING, "error processing via header, will send response to originating address\n");
   }

   return 0;
}

static int restart_monitor

(

void

)

[static]

Start the channel monitor thread.

Definition at line 29134 of file chan_sip.c.

References ast_log(), ast_mutex_lock, ast_mutex_unlock, ast_pthread_create_background, AST_PTHREADT_NULL, AST_PTHREADT_STOP, do_monitor(), LOG_ERROR, LOG_WARNING, and monlock.

Referenced by acl_change_event_cb(), load_module(), sip_reload(), and sip_request_call().

{
   /* If we're supposed to be stopped -- stay stopped */
   if (monitor_thread == AST_PTHREADT_STOP)
      return 0;
   ast_mutex_lock(&monlock);
   if (monitor_thread == pthread_self()) {
      ast_mutex_unlock(&monlock);
      ast_log(LOG_WARNING, "Cannot kill myself\n");
      return -1;
   }
   if (monitor_thread != AST_PTHREADT_NULL) {
      /* Wake up the thread */
      pthread_kill(monitor_thread, SIGURG);
   } else {
      /* Start a new monitor */
      if (ast_pthread_create_background(&monitor_thread, NULL, do_monitor, NULL) < 0) {
         ast_mutex_unlock(&monlock);
         ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
         return -1;
      }
   }
   ast_mutex_unlock(&monlock);
   return 0;
}

static void restart_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Restart session timer.

Definition at line 29179 of file chan_sip.c.

References ast_debug, ast_log(), AST_SCHED_DEL_UNREF, LOG_WARNING, start_session_timer(), and TRUE.

Referenced by handle_request_invite().

{
   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in restart_session_timer - %s\n", p->callid);
      return;
   }

   if (p->stimer->st_active == TRUE) {
      ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
      AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
            dialog_unref(p, "Removing session timer ref"));
      start_session_timer(p);
   }
}

static int retrans_pkt

(

const void *

data

)

[static]

Retransmit SIP message if no answer (Called from scheduler)

Definition at line 4085 of file chan_sip.c.

References __sip_xmit(), append_history, AST_CAUSE_NO_USER_RESPONSE, ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_trylock, ast_channel_unlock, ast_debug, ast_free, ast_log(), ast_queue_hangup_with_cause(), ast_sockaddr_stringify(), ast_str_buffer(), ast_test_flag, ast_tvdiff_ms(), ast_tvnow(), ast_verbose(), DEFAULT_RETRANS, LOG_WARNING, pvt_set_needdestroy(), sip_alreadygone(), sip_debug_test_pvt(), sip_methods, sip_nat_mode(), sip_pvt_lock, sip_pvt_unlock, sip_real_dst(), cfsip_methods::text, and UNLINK.

Referenced by __sip_reliable_xmit(), and sip_show_sched().

{
   struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
   int reschedule = DEFAULT_RETRANS;
   int xmitres = 0;
   /* how many ms until retrans timeout is reached */
   int64_t diff = pkt->retrans_stop_time - ast_tvdiff_ms(ast_tvnow(), pkt->time_sent);

   /* Do not retransmit if time out is reached. This will be negative if the time between
    * the first transmission and now is larger than our timeout period. This is a fail safe
    * check in case the scheduler gets behind or the clock is changed. */
   if ((diff <= 0) || (diff > pkt->retrans_stop_time)) {
      pkt->retrans_stop = 1;
   }

   /* Lock channel PVT */
   sip_pvt_lock(pkt->owner);

   if (!pkt->retrans_stop) {
      pkt->retrans++;
      if (!pkt->timer_t1) {   /* Re-schedule using timer_a and timer_t1 */
         if (sipdebug) {
            ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n",
               pkt->retransid,
               sip_methods[pkt->method].text,
               pkt->method);
         }
      } else {
         int siptimer_a;

         if (sipdebug) {
            ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n",
               pkt->retransid,
               pkt->retrans,
               sip_methods[pkt->method].text,
               pkt->method);
         }
         if (!pkt->timer_a) {
            pkt->timer_a = 2 ;
         } else {
            pkt->timer_a = 2 * pkt->timer_a;
         }

         /* For non-invites, a maximum of 4 secs */
         siptimer_a = pkt->timer_t1 * pkt->timer_a;   /* Double each time */
         if (pkt->method != SIP_INVITE && siptimer_a > 4000) {
            siptimer_a = 4000;
         }

         /* Reschedule re-transmit */
         reschedule = siptimer_a;
         ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n",
            pkt->retrans + 1,
            siptimer_a,
            pkt->timer_t1,
            pkt->retransid);
      }

      if (sip_debug_test_pvt(pkt->owner)) {
         const struct ast_sockaddr *dst = sip_real_dst(pkt->owner);
         ast_verbose("Retransmitting #%d (%s) to %s:\n%s\n---\n",
            pkt->retrans, sip_nat_mode(pkt->owner),
            ast_sockaddr_stringify(dst),
            ast_str_buffer(pkt->data));
      }

      append_history(pkt->owner, "ReTx", "%d %s", reschedule, ast_str_buffer(pkt->data));
      xmitres = __sip_xmit(pkt->owner, pkt->data);

      /* If there was no error during the network transmission, schedule the next retransmission,
       * but if the next retransmission is going to be beyond our timeout period, mark the packet's
       * stop_retrans value and set the next retransmit to be the exact time of timeout.  This will
       * allow any responses to the packet to be processed before the packet is destroyed on the next
       * call to this function by the scheduler. */
      if (xmitres != XMIT_ERROR) {
         if (reschedule >= diff) {
            pkt->retrans_stop = 1;
            reschedule = diff;
         }
         sip_pvt_unlock(pkt->owner);
         return  reschedule;
      }
   }

   /* At this point, either the packet's retransmission timed out, or there was a
    * transmission error, either way destroy the scheduler item and this packet. */

   pkt->retransid = -1; /* Kill this scheduler item */

   if (pkt->method != SIP_OPTIONS && xmitres == 0) {
      if (pkt->is_fatal || sipdebug) { /* Tell us if it's critical or if we're debugging */
         ast_log(LOG_WARNING, "Retransmission timeout reached on transmission %s for seqno %u (%s %s) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n"
            "Packet timed out after %dms with no response\n",
            pkt->owner->callid,
            pkt->seqno,
            pkt->is_fatal ? "Critical" : "Non-critical",
            pkt->is_resp ? "Response" : "Request",
            (int) ast_tvdiff_ms(ast_tvnow(), pkt->time_sent));
      }
   } else if (pkt->method == SIP_OPTIONS && sipdebug) {
      ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s)  -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions\n", pkt->owner->callid);
   }

   if (xmitres == XMIT_ERROR) {
      ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
      append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
   } else {
      append_history(pkt->owner, "MaxRetries", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
   }

   if (pkt->is_fatal) {
      while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
         sip_pvt_unlock(pkt->owner);   /* SIP_PVT, not channel */
         usleep(1);
         sip_pvt_lock(pkt->owner);
      }
      if (pkt->owner->owner && !ast_channel_hangupcause(pkt->owner->owner)) {
         ast_channel_hangupcause_set(pkt->owner->owner, AST_CAUSE_NO_USER_RESPONSE);
      }
      if (pkt->owner->owner) {
         ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).\n", pkt->owner->callid);

         if (pkt->is_resp &&
            (pkt->response_code >= 200) &&
            (pkt->response_code < 300) &&
            pkt->owner->pendinginvite &&
            ast_test_flag(&pkt->owner->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
            /* This is a timeout of the 2XX response to a pending INVITE.  In this case terminate the INVITE
             * transaction just as if we received the ACK, but immediately hangup with a BYE (sip_hangup
             * will send the BYE as long as the dialog is not set as "alreadygone")
             * RFC 3261 section 13.3.1.4.
             * "If the server retransmits the 2xx response for 64*T1 seconds without receiving
             * an ACK, the dialog is confirmed, but the session SHOULD be terminated.  This is
             * accomplished with a BYE, as described in Section 15." */
            pkt->owner->invitestate = INV_TERMINATED;
            pkt->owner->pendinginvite = 0;
         } else {
            /* there is nothing left to do, mark the dialog as gone */
            sip_alreadygone(pkt->owner);
         }
         ast_queue_hangup_with_cause(pkt->owner->owner, AST_CAUSE_NO_USER_RESPONSE);
         ast_channel_unlock(pkt->owner->owner);
      } else {
         /* If no channel owner, destroy now */

         /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
         if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) {
            pvt_set_needdestroy(pkt->owner, "no response to critical packet");
            sip_alreadygone(pkt->owner);
            append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately");
         }
      }
   }

   if (pkt->method == SIP_BYE) {
      /* We're not getting answers on SIP BYE's.  Tear down the call anyway. */
      sip_alreadygone(pkt->owner);
      if (pkt->owner->owner) {
         ast_channel_unlock(pkt->owner->owner);
      }
      append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");
      pvt_set_needdestroy(pkt->owner, "no response to BYE");
   }

   /* Remove the packet */
   for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
      if (cur == pkt) {
         UNLINK(cur, pkt->owner->packets, prev);
         sip_pvt_unlock(pkt->owner);
         if (pkt->owner) {
            pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
         }
         if (pkt->data) {
            ast_free(pkt->data);
         }
         pkt->data = NULL;
         ast_free(pkt);
         return 0;
      }
   }
   /* error case */
   ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
   sip_pvt_unlock(pkt->owner);
   return 0;
}

static void send_manager_peer_status	(	struct mansession *	s,
		struct sip_peer *	peer,
		const char *	idText
	)		[static]

Definition at line 19817 of file chan_sip.c.

References astman_append(), and status.

Referenced by manager_sip_peer_status().

{
   char time[128] = "";
   char status[128] = "";
   if (peer->maxms) {
      if (peer->lastms < 0) {
         snprintf(status, sizeof(status), "PeerStatus: Unreachable\r\n");
      } else if (peer->lastms > peer->maxms) {
         snprintf(status, sizeof(status), "PeerStatus: Lagged\r\n");
         snprintf(time, sizeof(time), "Time: %d\r\n", peer->lastms);
      } else if (peer->lastms) {
         snprintf(status, sizeof(status), "PeerStatus: Reachable\r\n");
         snprintf(time, sizeof(time), "Time: %d\r\n", peer->lastms);
      } else {
         snprintf(status, sizeof(status), "PeerStatus: Unknown\r\n");
      }
   } else {
      snprintf(status, sizeof(status), "PeerStatus: Unmonitored\r\n");
   }

   astman_append(s,
   "Event: PeerStatus\r\n"
   "Privilege: System\r\n"
   "ChannelType: SIP\r\n"
   "Peer: SIP/%s\r\n"
   "%s"
   "%s"
   "%s"
   "\r\n",
   peer->name, status, time, idText);
}

static int send_provisional_keepalive

(

const void *

data

)

[static]

Definition at line 4680 of file chan_sip.c.

References send_provisional_keepalive_full().

Referenced by update_provisional_keepalive().

{
   struct sip_pvt *pvt = (struct sip_pvt *) data;

   return send_provisional_keepalive_full(pvt, 0);
}

static int send_provisional_keepalive_full	(	struct sip_pvt *	pvt,
		int	with_sdp
	)		[static]

Definition at line 4630 of file chan_sip.c.

References ast_channel_unlock, ast_channel_unref, FALSE, S_OR, sip_pvt_lock_full(), sip_pvt_unlock, transmit_response(), and transmit_response_with_sdp().

Referenced by send_provisional_keepalive(), and send_provisional_keepalive_with_sdp().

{
   const char *msg = NULL;
   struct ast_channel *chan;
   int res = 0;
   int old_sched_id = pvt->provisional_keepalive_sched_id;

   chan = sip_pvt_lock_full(pvt);
   /* Check that nothing has changed while we were waiting for the lock */
   if (old_sched_id != pvt->provisional_keepalive_sched_id) {
      /* Keepalive has been cancelled or rescheduled, clean up and leave */
      if (chan) {
         ast_channel_unlock(chan);
         chan = ast_channel_unref(chan);
      }
      sip_pvt_unlock(pvt);
      dialog_unref(pvt, "dialog ref for provisional keepalive");
      return 0;
   }

   if (!pvt->last_provisional || !strncasecmp(pvt->last_provisional, "100", 3)) {
      msg = "183 Session Progress";
   }

   if (pvt->invitestate < INV_COMPLETED) {
      if (with_sdp) {
         transmit_response_with_sdp(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
      } else {
         transmit_response(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq);
      }
      res = PROVIS_KEEPALIVE_TIMEOUT;
   }

   if (chan) {
      ast_channel_unlock(chan);
      chan = ast_channel_unref(chan);
   }

   if (!res) {
      pvt->provisional_keepalive_sched_id = -1;
   }

   sip_pvt_unlock(pvt);

   if (!res) {
      dialog_unref(pvt, "dialog ref for provisional keepalive");
   }
   return res;
}

static int send_provisional_keepalive_with_sdp

(

const void *

data

)

[static]

Definition at line 4687 of file chan_sip.c.

References send_provisional_keepalive_full().

Referenced by update_provisional_keepalive().

{
   struct sip_pvt *pvt = (void *) data;

   return send_provisional_keepalive_full(pvt, 1);
}

static int send_request	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum xmittype	reliable,
		uint32_t	seqno
	)		[static]

Definition at line 4773 of file chan_sip.c.

References __sip_reliable_xmit(), __sip_xmit(), add_blank(), append_history, ast_sockaddr_stringify(), ast_str_buffer(), ast_test_flag, ast_verbose(), deinit_req(), finalize_content(), parse_copy(), sip_debug_test_pvt(), sip_get_header(), sip_methods, and cfsip_methods::text.

Referenced by sipinfo_send(), transmit_cc_notify(), transmit_info_with_aoc(), transmit_info_with_digit(), transmit_info_with_vidupdate(), transmit_invite(), transmit_message(), transmit_notify_with_mwi(), transmit_notify_with_sipfrag(), transmit_register(), transmit_reinvite_with_sdp(), transmit_request(), transmit_request_with_auth(), transmit_state_notify(), and update_connectedline().

{
   int res;

   /* If we have an outbound proxy, reset peer address
      Only do this once.
   */
   if (p->outboundproxy) {
      p->sa = p->outboundproxy->ip;
   }

   finalize_content(req);
   add_blank(req);
   if (sip_debug_test_pvt(p)) {
      if (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT)) {
         ast_verbose("%sTransmitting (NAT) to %s:\n%s\n---\n", reliable ? "Reliably " : "", ast_sockaddr_stringify(&p->recv), ast_str_buffer(req->data));
      } else {
         ast_verbose("%sTransmitting (no NAT) to %s:\n%s\n---\n", reliable ? "Reliably " : "", ast_sockaddr_stringify(&p->sa), ast_str_buffer(req->data));
      }
   }
   if (p->do_history) {
      struct sip_request tmp = { .rlpart1 = 0, };
      parse_copy(&tmp, req);
      append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", ast_str_buffer(tmp.data), sip_get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
      deinit_req(&tmp);
   }
   res = (reliable) ?
      __sip_reliable_xmit(p, seqno, 0, req->data, (reliable == XMIT_CRITICAL), req->method) :
      __sip_xmit(p, req->data);
   deinit_req(req);
   return res;
}

static int send_response	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum xmittype	reliable,
		uint32_t	seqno
	)		[static]

Transmit response on SIP request.

Definition at line 4731 of file chan_sip.c.

References __sip_reliable_xmit(), __sip_xmit(), add_blank(), append_history, AST_SCHED_DEL_UNREF, ast_sockaddr_stringify(), ast_str_buffer(), ast_verbose(), deinit_req(), finalize_content(), parse_copy(), sip_debug_test_pvt(), sip_get_header(), sip_methods, sip_nat_mode(), sip_real_dst(), and cfsip_methods::text.

Referenced by __transmit_response(), transmit_response_with_allow(), transmit_response_with_auth(), transmit_response_with_date(), transmit_response_with_minexpires(), transmit_response_with_minse(), transmit_response_with_retry_after(), transmit_response_with_sdp(), transmit_response_with_sip_etag(), transmit_response_with_t38_sdp(), transmit_response_with_unsupported(), update_connectedline(), and update_redirecting().

{
   int res;

   finalize_content(req);
   add_blank(req);
   if (sip_debug_test_pvt(p)) {
      const struct ast_sockaddr *dst = sip_real_dst(p);

      ast_verbose("\n<--- %sTransmitting (%s) to %s --->\n%s\n<------------>\n",
         reliable ? "Reliably " : "", sip_nat_mode(p),
         ast_sockaddr_stringify(dst),
         ast_str_buffer(req->data));
   }
   if (p->do_history) {
      struct sip_request tmp = { .rlpart1 = 0, };
      parse_copy(&tmp, req);
      append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", ast_str_buffer(tmp.data), sip_get_header(&tmp, "CSeq"),
         (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? REQ_OFFSET_TO_STR(&tmp, rlpart2) : sip_methods[tmp.method].text);
      deinit_req(&tmp);
   }

   /* If we are sending a final response to an INVITE, stop retransmitting provisional responses */
   if (p->initreq.method == SIP_INVITE && reliable == XMIT_CRITICAL) {
      AST_SCHED_DEL_UNREF(sched, p->provisional_keepalive_sched_id, dialog_unref(p, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));
   }

   res = (reliable) ?
       __sip_reliable_xmit(p, seqno, 1, req->data, (reliable == XMIT_CRITICAL), req->method) :
      __sip_xmit(p, req->data);
   deinit_req(req);
   if (res > 0) {
      return 0;
   }
   return res;
}

static enum ast_cc_service_type service_string_to_service_type

(

const char *const

service_string

)

[static]

Definition at line 900 of file chan_sip.c.

References AST_CC_CCBS, AST_CC_CCNL, AST_CC_NONE, service, and sip_cc_service_map.

Referenced by sip_get_cc_information().

{
   enum ast_cc_service_type service;
   for (service = AST_CC_CCBS; service <= AST_CC_CCNL; ++service) {
      if (!strcasecmp(service_string, sip_cc_service_map[service].service_string)) {
         return service;
      }
   }
   return AST_CC_NONE;
}

static int set_address_from_contact

(

struct sip_pvt *

pvt

)

[static]

Change the other partys IP address based on given contact.

Todo:

We need to save the TRANSPORT here too

Definition at line 15985 of file chan_sip.c.

References __set_address_from_contact(), and ast_test_flag.

Referenced by handle_response_invite().

{
   if (ast_test_flag(&pvt->flags[0], SIP_NAT_FORCE_RPORT)) {
      /* NAT: Don't trust the contact field.  Just use what they came to us
         with. */
      /*! \todo We need to save the TRANSPORT here too */
      pvt->sa = pvt->recv;
      return 0;
   }

   return __set_address_from_contact(pvt->fullcontact, &pvt->sa, pvt->socket.type == SIP_TRANSPORT_TLS ? 1 : 0);
}

static void set_destination	(	struct sip_pvt *	p,
		char *	uri
	)		[static]

Set destination from SIP URI.

Parse uri to h (host) and port - uri is already just the part inside the <> general form we are expecting is sip[s]:username[:password][;parameter][:port][;...] If there's a port given, turn NAPTR/SRV off. NAPTR might indicate SIPS preference even for SIP: uri's

If there's a sips: uri scheme, TLS will be required.

Todo:

XXX If we have sip_cfg.srvlookup on, then look for NAPTR/SRV, otherwise, just look for A records

Todo:

XXX If we have sip_cfg.srvlookup on, then look for NAPTR/SRV, otherwise, just look for A records

Definition at line 11598 of file chan_sip.c.

References ast_copy_string(), ast_log(), ast_sockaddr_port, ast_sockaddr_resolve_first_transport(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_verbose(), debug, FALSE, hostname, LOG_WARNING, PARSE_PORT_FORBID, sip_debug_test_pvt(), and TRUE.

Referenced by reqprep().

{
   char *trans, *h, *maddr, hostname[256];
   int hn;
   int debug=sip_debug_test_pvt(p);
   int tls_on = FALSE;

   if (debug)
      ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);

   if ((trans = strcasestr(uri, ";transport="))) {
      trans += strlen(";transport=");

      if (!strncasecmp(trans, "ws", 2)) {
         if (debug)
            ast_verbose("set_destination: URI is for WebSocket, we can't set destination\n");
         return;
      }
   }

   /* Find and parse hostname */
   h = strchr(uri, '@');
   if (h)
      ++h;
   else {
      h = uri;
      if (!strncasecmp(h, "sip:", 4)) {
         h += 4;
      } else if (!strncasecmp(h, "sips:", 5)) {
         h += 5;
         tls_on = TRUE;
      }
   }
   hn = strcspn(h, ";>") + 1;
   if (hn > sizeof(hostname))
      hn = sizeof(hostname);
   ast_copy_string(hostname, h, hn);
   /* XXX bug here if string has been trimmed to sizeof(hostname) */
   h += hn - 1;

   /*! \todo XXX If we have sip_cfg.srvlookup on, then look for NAPTR/SRV,
    * otherwise, just look for A records */
   if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, 0, p->socket.type)) {
      ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
      return;
   }

   /* Got the hostname - but maybe there's a "maddr=" to override address? */
   maddr = strstr(h, "maddr=");
   if (maddr) {
      int port;

      maddr += 6;
      hn = strspn(maddr, "abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ"
                    "0123456789-.:[]") + 1;
      if (hn > sizeof(hostname))
         hn = sizeof(hostname);
      ast_copy_string(hostname, maddr, hn);

      port = ast_sockaddr_port(&p->sa);

      /*! \todo XXX If we have sip_cfg.srvlookup on, then look for
       * NAPTR/SRV, otherwise, just look for A records */
      if (ast_sockaddr_resolve_first_transport(&p->sa, hostname, PARSE_PORT_FORBID, p->socket.type)) {
         ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
         return;
      }

      ast_sockaddr_set_port(&p->sa, port);
   }

   if (!ast_sockaddr_port(&p->sa)) {
      ast_sockaddr_set_port(&p->sa, tls_on ?
                  STANDARD_TLS_PORT : STANDARD_SIP_PORT);
   }

   if (debug) {
      ast_verbose("set_destination: set destination to %s\n",
             ast_sockaddr_stringify(&p->sa));
   }
}

static void set_insecure_flags	(	struct ast_flags *	flags,
		const char *	value,
		int	lineno
	)		[static]

Parse insecure= setting in sip.conf and set flags according to setting.

Definition at line 30046 of file chan_sip.c.

References ast_copy_string(), ast_false(), ast_log(), ast_set_flag, ast_strlen_zero(), LOG_WARNING, and word.

Referenced by get_insecure_variable_from_config(), get_insecure_variable_from_sipregs(), and handle_common_options().

{
   if (ast_strlen_zero(value))
      return;

   if (!ast_false(value)) {
      char buf[64];
      char *word, *next;

      ast_copy_string(buf, value, sizeof(buf));
      next = buf;
      while ((word = strsep(&next, ","))) {
         if (!strcasecmp(word, "port"))
            ast_set_flag(&flags[0], SIP_INSECURE_PORT);
         else if (!strcasecmp(word, "invite"))
            ast_set_flag(&flags[0], SIP_INSECURE_INVITE);
         else
            ast_log(LOG_WARNING, "Unknown insecure mode '%s' on line %d\n", value, lineno);
      }
   }
}

static int set_message_vars_from_req	(	struct ast_msg *	msg,
		struct sip_request *	req
	)		[static]

Definition at line 18533 of file chan_sip.c.

References ast_copy_string(), ast_msg_set_var(), ast_skip_blanks(), ast_trim_blanks(), find_full_alias(), MIN, and name.

Referenced by receive_message().

{
   size_t x;
   char name_buf[1024];
   char val_buf[1024];
   const char *name;
   char *c;
   int res = 0;

   for (x = 0; x < req->headers; x++) {
      const char *header = REQ_OFFSET_TO_STR(req, header[x]);

      if ((c = strchr(header, ':'))) {
         ast_copy_string(name_buf, header, MIN((c - header + 1), sizeof(name_buf)));
         ast_copy_string(val_buf, ast_skip_blanks(c + 1), sizeof(val_buf));
         ast_trim_blanks(name_buf);

         /* Convert header name to full name alias. */
         name = find_full_alias(name_buf, name_buf);

         res = ast_msg_set_var(msg, name, val_buf);
         if (res) {
            break;
         }
      }
   }
   return res;
}

static void set_peer_defaults

(

struct sip_peer *

peer

)

[static]

Set peer defaults before configuring specific configurations.

Definition at line 30471 of file chan_sip.c.

References ao2_ref, ast_copy_flags, ast_format_cap_copy(), ast_sockaddr_setnull(), ast_string_field_set, cid_name, cid_num, clear_peer_mailboxes(), context, default_keepalive, default_maxcallbitrate, default_prefs, default_primary_transport, default_qualify, default_transports, global_autoframing, global_max_se, global_min_se, global_qualifyfreq, global_rtpholdtimeout, global_rtpkeepalive, global_rtptimeout, global_st_mode, global_st_refresher, global_t1, global_t38_maxdatagram, global_timer_b, language, mohinterpret, mohsuggest, secret, set_socket_transport(), sip_cfg, and vmexten.

Referenced by build_peer(), and temp_peer().

{
   if (peer->expire == 0) {
      /* Don't reset expire or port time during reload
         if we have an active registration
      */
      peer->expire = -1;
      peer->pokeexpire = -1;
      peer->keepalivesend = -1;
      set_socket_transport(&peer->socket, SIP_TRANSPORT_UDP);
   }
   peer->type = SIP_TYPE_PEER;
   ast_copy_flags(&peer->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&peer->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&peer->flags[2], &global_flags[2], SIP_PAGE3_FLAGS_TO_COPY);
   ast_string_field_set(peer, context, sip_cfg.default_context);
   ast_string_field_set(peer, record_on_feature, sip_cfg.default_record_on_feature);
   ast_string_field_set(peer, record_off_feature, sip_cfg.default_record_off_feature);
   ast_string_field_set(peer, messagecontext, sip_cfg.messagecontext);
   ast_string_field_set(peer, subscribecontext, sip_cfg.default_subscribecontext);
   ast_string_field_set(peer, language, default_language);
   ast_string_field_set(peer, mohinterpret, default_mohinterpret);
   ast_string_field_set(peer, mohsuggest, default_mohsuggest);
   ast_string_field_set(peer, engine, default_engine);
   ast_sockaddr_setnull(&peer->addr);
   ast_sockaddr_setnull(&peer->defaddr);
   ast_format_cap_copy(peer->caps, sip_cfg.caps);
   peer->maxcallbitrate = default_maxcallbitrate;
   peer->rtptimeout = global_rtptimeout;
   peer->rtpholdtimeout = global_rtpholdtimeout;
   peer->rtpkeepalive = global_rtpkeepalive;
   peer->allowtransfer = sip_cfg.allowtransfer;
   peer->autoframing = global_autoframing;
   peer->t38_maxdatagram = global_t38_maxdatagram;
   peer->qualifyfreq = global_qualifyfreq;
   if (global_callcounter)
      peer->call_limit=INT_MAX;
   ast_string_field_set(peer, vmexten, default_vmexten);
   ast_string_field_set(peer, secret, "");
   ast_string_field_set(peer, description, "");
   ast_string_field_set(peer, remotesecret, "");
   ast_string_field_set(peer, md5secret, "");
   ast_string_field_set(peer, cid_num, "");
   ast_string_field_set(peer, cid_name, "");
   ast_string_field_set(peer, cid_tag, "");
   ast_string_field_set(peer, fromdomain, "");
   ast_string_field_set(peer, fromuser, "");
   ast_string_field_set(peer, regexten, "");
   peer->callgroup = 0;
   peer->pickupgroup = 0;
   peer->maxms = default_qualify;
   peer->keepalive = default_keepalive;
   peer->prefs = default_prefs;
   ast_string_field_set(peer, zone, default_zone);
   peer->stimer.st_mode_oper = global_st_mode;  /* Session-Timers */
   peer->stimer.st_ref = global_st_refresher;
   peer->stimer.st_min_se = global_min_se;
   peer->stimer.st_max_se = global_max_se;
   peer->timer_t1 = global_t1;
   peer->timer_b = global_timer_b;
   clear_peer_mailboxes(peer);
   peer->disallowed_methods = sip_cfg.disallowed_methods;
   peer->transports = default_transports;
   peer->default_outbound_transport = default_primary_transport;
   if (peer->outboundproxy) {
      ao2_ref(peer->outboundproxy, -1);
      peer->outboundproxy = NULL;
   }
}

static void set_peer_nat	(	const struct sip_pvt *	p,
		struct sip_peer *	peer
	)		[static]

Set the peers nat flags if they are using auto_* settings.

Definition at line 18026 of file chan_sip.c.

References ast_clear_flag, ast_set_flag, and ast_test_flag.

Referenced by check_peer_ok(), register_verify(), and sip_request_call().

{

   if (!p || !peer) {
      return;
   }

   if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_RPORT)) {
      if (p->natdetected) {
         ast_set_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
      } else {
         ast_clear_flag(&peer->flags[0], SIP_NAT_FORCE_RPORT);
      }
   }

   if (ast_test_flag(&peer->flags[2], SIP_PAGE3_NAT_AUTO_COMEDIA)) {
      if (p->natdetected) {
         ast_set_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
      } else {
         ast_clear_flag(&peer->flags[1], SIP_PAGE2_SYMMETRICRTP);
      }
   }
}

static unsigned int set_pvt_allowed_methods	(	struct sip_pvt *	pvt,
		struct sip_request *	req
	)		[static]

A wrapper for parse_allowed_methods geared toward sip_pvts

This function, in addition to setting the allowed methods for a sip_pvt also will take into account the setting of the SIP_PAGE2_RPID_UPDATE flag.

Parameters:

pvt	The sip_pvt we are setting the allowed_methods for
req	The request which we are parsing

Return values:

The	methods alloweded by the sip_pvt

Definition at line 9536 of file chan_sip.c.

References ast_test_flag, mark_method_allowed(), and parse_allowed_methods().

Referenced by check_peer_ok(), handle_request_invite(), handle_request_options(), handle_request_subscribe(), handle_response_invite(), handle_response_subscribe(), and receive_message().

{
   pvt->allowed_methods = parse_allowed_methods(req);
   
   if (ast_test_flag(&pvt->flags[1], SIP_PAGE2_RPID_UPDATE)) {
      mark_method_allowed(&pvt->allowed_methods, SIP_UPDATE);
   }
   pvt->allowed_methods &= ~(pvt->disallowed_methods);

   return pvt->allowed_methods;
}

static void set_socket_transport	(	struct sip_socket *	socket,
		int	transport
	)		[static]

Definition at line 15733 of file chan_sip.c.

References ao2_ref, and ast_websocket_unref().

Referenced by __sip_subscribe_mwi_do(), _sip_tcp_helper_thread(), build_peer(), create_addr(), expire_register(), get_transport_pvt(), parse_moved_contact(), parse_register_contact(), set_peer_defaults(), sip_alloc(), sip_request_call(), sip_send_mwi_to_peer(), sip_websocket_callback(), sipsock_read(), and transmit_register().

{
   /* if the transport type changes, clear all socket data */
   if (socket->type != transport) {
      socket->fd = -1;
      socket->type = transport;
      if (socket->tcptls_session) {
         ao2_ref(socket->tcptls_session, -1);
         socket->tcptls_session = NULL;
      } else if (socket->ws_session) {
         ast_websocket_unref(socket->ws_session);
         socket->ws_session = NULL;
      }
   }
}

static void set_t38_capabilities

(

struct sip_pvt *

p

)

[static]

Set the global T38 capabilities on a SIP dialog structure.

Definition at line 5806 of file chan_sip.c.

References ast_test_flag, ast_udptl_set_error_correction_scheme(), UDPTL_ERROR_CORRECTION_FEC, UDPTL_ERROR_CORRECTION_NONE, and UDPTL_ERROR_CORRECTION_REDUNDANCY.

Referenced by check_peer_ok(), and initialize_udptl().

{
   if (p->udptl) {
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL_REDUNDANCY) {
                        ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_REDUNDANCY);
      } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL_FEC) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_FEC);
      } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT) == SIP_PAGE2_T38SUPPORT_UDPTL) {
         ast_udptl_set_error_correction_scheme(p->udptl, UDPTL_ERROR_CORRECTION_NONE);
      }
   }
}

static int setup_srtp

(

struct sip_srtp **

srtp

)

[static]

Definition at line 33289 of file chan_sip.c.

References ast_log(), ast_rtp_engine_srtp_is_registered(), LOG_ERROR, and sip_srtp_alloc().

Referenced by process_crypto(), and sip_call().

{
   if (!ast_rtp_engine_srtp_is_registered()) {
      ast_log(LOG_ERROR, "No SRTP module loaded, can't setup SRTP session.\n");
      return -1;
   }

   if (!(*srtp = sip_srtp_alloc())) { /* Allocate SRTP data structure */
      return -1;
   }

   return 0;
}

static int show_channels_cb	(	void *	__cur,
		void *	__arg,
		int	flags
	)		[static]

callback for show channel|subscription

Definition at line 20948 of file chan_sip.c.

References ast_channel_nativeformats(), ast_cli(), AST_CLI_YESNO, ast_extension_state2str(), ast_getformatname_multiple(), ast_sockaddr_stringify_addr(), ast_str_alloca, ast_str_buffer(), ast_test_flag, FORMAT, FORMAT4, NONE, peer_mailboxes_to_str(), referstatus2str(), S_OR, sip_pvt_lock, sip_pvt_unlock, sip_real_dst(), and subscription_type2str().

Referenced by sip_show_channels().

{
   struct sip_pvt *cur = __cur;
   struct __show_chan_arg *arg = __arg;
   const struct ast_sockaddr *dst;

   sip_pvt_lock(cur);
   dst = sip_real_dst(cur);

   /* XXX indentation preserved to reduce diff. Will be fixed later */
   if (cur->subscribed == NONE && !arg->subscriptions) {
      /* set if SIP transfer in progress */
      const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : "";
      char formatbuf[SIPBUFSIZE/2];
      
      ast_cli(arg->fd, FORMAT, ast_sockaddr_stringify_addr(dst),
            S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
            cur->callid,
            cur->owner ? ast_getformatname_multiple(formatbuf, sizeof(formatbuf), ast_channel_nativeformats(cur->owner)) : "(nothing)",
            AST_CLI_YESNO(ast_test_flag(&cur->flags[1], SIP_PAGE2_CALL_ONHOLD)),
            cur->needdestroy ? "(d)" : "",
            cur->lastmsg ,
            referstatus,
            cur->relatedpeer ? cur->relatedpeer->name : "<guest>"
         );
      arg->numchans++;
   }
   if (cur->subscribed != NONE && arg->subscriptions) {
      struct ast_str *mailbox_str = ast_str_alloca(512);
      if (cur->subscribed == MWI_NOTIFICATION && cur->relatedpeer)
         peer_mailboxes_to_str(&mailbox_str, cur->relatedpeer);
      ast_cli(arg->fd, FORMAT4, ast_sockaddr_stringify_addr(dst),
            S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
               cur->callid,
            /* the 'complete' exten/context is hidden in the refer_to field for subscriptions */
            cur->subscribed == MWI_NOTIFICATION ? "--" : cur->subscribeuri,
            cur->subscribed == MWI_NOTIFICATION ? "<none>" : ast_extension_state2str(cur->laststate),
            subscription_type2str(cur->subscribed),
            cur->subscribed == MWI_NOTIFICATION ? S_OR(ast_str_buffer(mailbox_str), "<none>") : "<none>",
            cur->expiry
         );
      arg->numchans++;
   }
   sip_pvt_unlock(cur);
   return 0;   /* don't care, we scan all channels */
}

static int show_chanstats_cb	(	void *	__cur,
		void *	__arg,
		int	flags
	)		[static]

Callback for show_chanstats.

Definition at line 20541 of file chan_sip.c.

References ast_channel_cdr(), ast_cli(), ast_log(), ast_rtp_instance_get_stats(), AST_RTP_INSTANCE_STAT_ALL, ast_sockaddr_stringify_addr(), ast_tvdiff_ms(), ast_tvnow(), ast_tvzero(), invstate2stringtable::desc, FORMAT, invitestate2string, LOG_WARNING, NONE, ast_rtp_instance_stats::rxcount, ast_rtp_instance_stats::rxjitter, ast_rtp_instance_stats::rxploss, sip_pvt_lock, sip_pvt_unlock, ast_rtp_instance_stats::txcount, ast_rtp_instance_stats::txjitter, and ast_rtp_instance_stats::txploss.

Referenced by sip_show_channelstats().

{
#define FORMAT2 "%-15.15s  %-11.11s  %-8.8s %-10.10s  %-10.10s (     %%) %-6.6s %-10.10s  %-10.10s (     %%) %-6.6s\n"
#define FORMAT  "%-15.15s  %-11.11s  %-8.8s %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf %-10.10u%-1.1s %-10.10u (%5.2f%%) %-6.4lf\n"
   struct sip_pvt *cur = __cur;
   struct ast_rtp_instance_stats stats;
   char durbuf[10];
   int duration;
   int durh, durm, durs;
   struct ast_channel *c;
   struct __show_chan_arg *arg = __arg;
   int fd = arg->fd;

   sip_pvt_lock(cur);
   c = cur->owner;

   if (cur->subscribed != NONE) {
      /* Subscriptions */
      sip_pvt_unlock(cur);
      return 0;   /* don't care, we scan all channels */
   }

   if (!cur->rtp) {
      if (sipdebug) {
         ast_cli(fd, "%-15.15s  %-11.11s (inv state: %s) -- %s\n",
            ast_sockaddr_stringify_addr(&cur->sa), cur->callid,
            invitestate2string[cur->invitestate].desc,
            "-- No RTP active");
      }
      sip_pvt_unlock(cur);
      return 0;   /* don't care, we scan all channels */
   }

   if (ast_rtp_instance_get_stats(cur->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
      sip_pvt_unlock(cur);
      ast_log(LOG_WARNING, "Could not get RTP stats.\n");
      return 0;
   }

   if (c && ast_channel_cdr(c) && !ast_tvzero(ast_channel_cdr(c)->start)) {
      duration = (int)(ast_tvdiff_ms(ast_tvnow(), ast_channel_cdr(c)->start) / 1000);
      durh = duration / 3600;
      durm = (duration % 3600) / 60;
      durs = duration % 60;
      snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
   } else {
      durbuf[0] = '\0';
   }

   ast_cli(fd, FORMAT,
      ast_sockaddr_stringify_addr(&cur->sa),
      cur->callid,
      durbuf,
      stats.rxcount > (unsigned int) 100000 ? (unsigned int) (stats.rxcount)/(unsigned int) 1000 : stats.rxcount,
      stats.rxcount > (unsigned int) 100000 ? "K":" ",
      stats.rxploss,
      (stats.rxcount + stats.rxploss) > 0 ? (double) stats.rxploss / (stats.rxcount + stats.rxploss) * 100 : 0,
      stats.rxjitter,
      stats.txcount > (unsigned int) 100000 ? (unsigned int) (stats.txcount)/(unsigned int) 1000 : stats.txcount,
      stats.txcount > (unsigned int) 100000 ? "K":" ",
      stats.txploss,
      stats.txcount > 0 ? (double) stats.txploss / stats.txcount * 100 : 0,
      stats.txjitter
   );
   arg->numchans++;
   sip_pvt_unlock(cur);

   return 0;   /* don't care, we scan all channels */
}

static int sip_addheader	(	struct ast_channel *	chan,
		const char *	data
	)		[static]

Add a SIP header to an outbound INVITE.

Definition at line 33029 of file chan_sip.c.

References ast_alloca, ast_channel_lock, ast_channel_unlock, ast_debug, ast_get_encoded_str(), ast_log(), ast_strlen_zero(), FALSE, inbuf(), len(), LOG_WARNING, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), and TRUE.

Referenced by load_module().

{
   int no = 0;
   int ok = FALSE;
   char varbuf[30];
   const char *inbuf = data;
   char *subbuf;
   
   if (ast_strlen_zero(inbuf)) {
      ast_log(LOG_WARNING, "This application requires the argument: Header\n");
      return 0;
   }
   ast_channel_lock(chan);

   /* Check for headers */
   while (!ok && no <= 50) {
      no++;
      snprintf(varbuf, sizeof(varbuf), "__SIPADDHEADER%.2d", no);

      /* Compare without the leading underscores */
      if ((pbx_builtin_getvar_helper(chan, (const char *) varbuf + 2) == (const char *) NULL)) {
         ok = TRUE;
      }
   }
   if (ok) {
      size_t len = strlen(inbuf);
      subbuf = ast_alloca(len + 1);
      ast_get_encoded_str(inbuf, subbuf, len + 1);
      pbx_builtin_setvar_helper(chan, varbuf, subbuf);
      if (sipdebug) {
         ast_debug(1, "SIP Header added \"%s\" as %s\n", inbuf, varbuf);
      }
   } else {
      ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
   }
   ast_channel_unlock(chan);
   return 0;
}

struct sip_pvt* sip_alloc	(	ast_string_field	callid,
		struct ast_sockaddr *	addr,
		int	useglobal_nat,
		const int	intended_method,
		struct sip_request *	req,
		struct ast_callid *	logger_callid
	)		[read]

Allocate sip_pvt structure, set defaults and link in the container. Returns a reference to the object so whoever uses it later must remember to release the reference.

Definition at line 8631 of file chan_sip.c.

References ao2_t_alloc, ao2_t_link, ao2_t_ref, ast_cc_config_params_init, ast_copy_flags, ast_copy_string(), ast_debug, ast_format_cap_alloc_nolock(), ast_format_cap_copy(), ast_format_cap_destroy(), AST_LIST_HEAD_INIT_NOLOCK, ast_random(), AST_RTP_DTMF, ast_sip_ouraddrfor(), ast_sockaddr_copy(), ast_string_field_init, ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_callid_pvt(), build_via(), check_via(), context, default_fromdomainport, default_maxcallbitrate, default_prefs, do_setnat(), free_via(), global_autoframing, global_t1, global_timer_b, internip, make_our_tag(), mohinterpret, mohsuggest, NONE, parkinglot, parse_via(), recordhistory, set_socket_transport(), sip_cfg, sip_destroy_fn(), sip_get_header(), sip_methods, sip_pvt_callid_set(), cfsip_methods::text, and TRUE.

Referenced by __sip_subscribe_mwi_do(), find_call(), forked_invite_init(), manager_sipnotify(), sip_cc_monitor_request_cc(), sip_cli_notify(), sip_msg_send(), sip_poke_peer(), sip_request_call(), sip_send_mwi_to_peer(), transmit_publish(), and transmit_register().

{
   struct sip_pvt *p;

   if (!(p = ao2_t_alloc(sizeof(*p), sip_destroy_fn, "allocate a dialog(pvt) struct")))
      return NULL;

   if (ast_string_field_init(p, 512)) {
      ao2_t_ref(p, -1, "failed to string_field_init, drop p");
      return NULL;
   }

   if (!(p->cc_params = ast_cc_config_params_init())) {
      ao2_t_ref(p, -1, "Yuck, couldn't allocate cc_params struct. Get rid o' p");
      return NULL;
   }

   if (logger_callid) {
      sip_pvt_callid_set(p, logger_callid);
   }

   p->caps = ast_format_cap_alloc_nolock();
   p->jointcaps = ast_format_cap_alloc_nolock();
   p->peercaps = ast_format_cap_alloc_nolock();
   p->redircaps = ast_format_cap_alloc_nolock();
   p->prefcaps = ast_format_cap_alloc_nolock();

   if (!p->caps|| !p->jointcaps || !p->peercaps || !p->redircaps) {
      p->caps = ast_format_cap_destroy(p->caps);
      p->jointcaps = ast_format_cap_destroy(p->jointcaps);
      p->peercaps = ast_format_cap_destroy(p->peercaps);
      p->redircaps = ast_format_cap_destroy(p->redircaps);
      p->prefcaps = ast_format_cap_destroy(p->prefcaps);
      ao2_t_ref(p, -1, "Yuck, couldn't allocate format capabilities. Get rid o' p");
      return NULL;
   }


   /* If this dialog is created as a result of a request or response, lets store
    * some information about it in the dialog. */
   if (req) {
      struct sip_via *via;
      const char *cseq = sip_get_header(req, "Cseq");
      uint32_t seqno;

      /* get branch parameter from initial Request that started this dialog */
      via = parse_via(sip_get_header(req, "Via"));
      if (via) {
         /* only store the branch if it begins with the magic prefix "z9hG4bK", otherwise
          * it is not useful to us to have it */
         if (!ast_strlen_zero(via->branch) && !strncasecmp(via->branch, "z9hG4bK", 7)) {
            ast_string_field_set(p, initviabranch, via->branch);
            ast_string_field_set(p, initviasentby, via->sent_by);
         }
         free_via(via);
      }

      /* Store initial incoming cseq. An error in sscanf here is ignored.  There is no approperiate
       * except not storing the number.  CSeq validation must take place before dialog creation in find_call */
      if (!ast_strlen_zero(cseq) && (sscanf(cseq, "%30u", &seqno) == 1)) {
         p->init_icseq = seqno;
      }
      /* Later in ast_sip_ouraddrfor we need this to choose the right ip and port for the specific transport */
      set_socket_transport(&p->socket, req->socket.type);
   } else {
      set_socket_transport(&p->socket, SIP_TRANSPORT_UDP);
   }

   p->socket.fd = -1;
   p->method = intended_method;
   p->initid = -1;
   p->waitid = -1;
   p->reinviteid = -1;
   p->autokillid = -1;
   p->request_queue_sched_id = -1;
   p->provisional_keepalive_sched_id = -1;
   p->t38id = -1;
   p->subscribed = NONE;
   p->stateid = -1;
   p->sessionversion_remote = -1;
   p->session_modify = TRUE;
   p->stimer = NULL;
   p->prefs = default_prefs;     /* Set default codecs for this call */
   ast_copy_string(p->zone, default_zone, sizeof(p->zone));
   p->maxforwards = sip_cfg.default_max_forwards;

   if (intended_method != SIP_OPTIONS) {  /* Peerpoke has it's own system */
      p->timer_t1 = global_t1;   /* Default SIP retransmission timer T1 (RFC 3261) */
      p->timer_b = global_timer_b;  /* Default SIP transaction timer B (RFC 3261) */
   }

   if (!addr) {
      p->ourip = internip;
   } else {
      ast_sockaddr_copy(&p->sa, addr);
      ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
   }

   /* Copy global flags to this PVT at setup. */
   ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[2], &global_flags[2], SIP_PAGE3_FLAGS_TO_COPY);

   p->do_history = recordhistory;

   p->branch = ast_random();  
   make_our_tag(p);
   p->ocseq = INITIAL_CSEQ;
   p->allowed_methods = UINT_MAX;

   if (sip_methods[intended_method].need_rtp) {
      p->maxcallbitrate = default_maxcallbitrate;
      p->autoframing = global_autoframing;
   }

   if (useglobal_nat && addr) {
      /* Setup NAT structure according to global settings if we have an address */
      ast_sockaddr_copy(&p->recv, addr);
      check_via(p, req);
      do_setnat(p);
   }

   if (p->method != SIP_REGISTER) {
      ast_string_field_set(p, fromdomain, default_fromdomain);
      p->fromdomainport = default_fromdomainport;
   }
   build_via(p);
   if (!callid)
      build_callid_pvt(p);
   else
      ast_string_field_set(p, callid, callid);
   /* Assign default music on hold class */
   ast_string_field_set(p, mohinterpret, default_mohinterpret);
   ast_string_field_set(p, mohsuggest, default_mohsuggest);
   ast_format_cap_copy(p->caps, sip_cfg.caps);
   p->allowtransfer = sip_cfg.allowtransfer;
   if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
       (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      p->noncodeccapability |= AST_RTP_DTMF;
   }
   ast_string_field_set(p, context, sip_cfg.default_context);
   ast_string_field_set(p, parkinglot, default_parkinglot);
   ast_string_field_set(p, engine, default_engine);

   AST_LIST_HEAD_INIT_NOLOCK(&p->request_queue);
   AST_LIST_HEAD_INIT_NOLOCK(&p->offered_media);

   /* Add to active dialog list */

   ao2_t_link(dialogs, p, "link pvt into dialogs table");

   ast_debug(1, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : p->callid, sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
   return p;
}

static int sip_allow_anyrtp_remote	(	struct ast_channel *	chan1,
		struct ast_channel *	chan2,
		char *	rtptype
	)		[static]

Definition at line 32660 of file chan_sip.c.

References apply_directmedia_acl(), ast_channel_tech_pvt(), ast_duplicate_acl_list(), ast_free_acl_list(), ast_test_flag, sip_pvt_lock, and sip_pvt_unlock.

Referenced by sip_allow_rtp_remote(), and sip_allow_vrtp_remote().

{
   struct sip_pvt *p1 = NULL, *p2 = NULL;
   struct ast_acl_list *p2_directmediaacl = NULL; /* opposed directmediaha for comparing against first channel host address */
   struct ast_acl_list *p1_directmediaacl = NULL; /* opposed directmediaha for comparing against second channel host address */
   int res = 1;

   if (!(p1 = ast_channel_tech_pvt(chan1))) {
      return 0;
   }

   if (!(p2 = ast_channel_tech_pvt(chan2))) {
      return 0;
   }

   sip_pvt_lock(p2);
   if (p2->relatedpeer && p2->relatedpeer->directmediaacl) {
      p2_directmediaacl = ast_duplicate_acl_list(p2->relatedpeer->directmediaacl);
   }
   sip_pvt_unlock(p2);

   sip_pvt_lock(p1);
   if (p1->relatedpeer && p1->relatedpeer->directmediaacl) {
      p1_directmediaacl = ast_duplicate_acl_list(p1->relatedpeer->directmediaacl);
   }

   if (p2_directmediaacl && ast_test_flag(&p1->flags[0], SIP_DIRECT_MEDIA)) {
      if (!apply_directmedia_acl(p1, p2_directmediaacl, rtptype)) {
         res = 0;
      }
   }
   sip_pvt_unlock(p1);

   if (res == 0) {
      goto allow_anyrtp_remote_end;
   }

   sip_pvt_lock(p2);
   if (p1_directmediaacl && ast_test_flag(&p2->flags[0], SIP_DIRECT_MEDIA)) {
      if (!apply_directmedia_acl(p2, p1_directmediaacl, rtptype)) {
         res = 0;
      }
   }
   sip_pvt_unlock(p2);

allow_anyrtp_remote_end:

   if (p2_directmediaacl) {
      p2_directmediaacl = ast_free_acl_list(p2_directmediaacl);
   }

   if (p1_directmediaacl) {
      p1_directmediaacl = ast_free_acl_list(p1_directmediaacl);
   }

   return res;
}

static int sip_allow_rtp_remote	(	struct ast_channel *	chan1,
		struct ast_channel *	chan2
	)		[static]

Definition at line 32718 of file chan_sip.c.

References sip_allow_anyrtp_remote().

{
   return sip_allow_anyrtp_remote(chan1, chan2, "audio");
}

static int sip_allow_vrtp_remote	(	struct ast_channel *	chan1,
		struct ast_channel *	chan2
	)		[static]

Definition at line 32723 of file chan_sip.c.

References sip_allow_anyrtp_remote().

{
   return sip_allow_anyrtp_remote(chan1, chan2, "video");
}

static void sip_alreadygone

(

struct sip_pvt *

dialog

)

[static]

Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging.

Definition at line 3530 of file chan_sip.c.

References ast_debug.

Referenced by handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), handle_response_publish(), handle_response_subscribe(), retrans_pkt(), sip_indicate(), and sip_sipredirect().

{
   ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
   dialog->alreadygone = 1;
}

static int sip_answer

(

struct ast_channel *

ast

)

[static]

sip_answer: Answer SIP call , send 200 OK on Invite Part of PBX interface

Definition at line 7262 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_rtp_instance_update_source(), ast_set_flag, ast_setstate(), AST_STATE_UP, ast_test_flag, FALSE, sip_pvt_lock, sip_pvt_unlock, transmit_response_with_sdp(), TRUE, and try_suggested_sip_codec().

{
   int res = 0;
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int oldsdp = FALSE;

   if (!p) {
      ast_debug(1, "Asked to answer channel %s without tech pvt; ignoring\n",
            ast_channel_name(ast));
      return res;
   }
   sip_pvt_lock(p);
   if (ast_channel_state(ast) != AST_STATE_UP) {
      try_suggested_sip_codec(p);

      if (ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
         oldsdp = TRUE;
      }

      ast_setstate(ast, AST_STATE_UP);
      ast_debug(1, "SIP answering channel: %s\n", ast_channel_name(ast));
      ast_rtp_instance_update_source(p->rtp);
      res = transmit_response_with_sdp(p, "200 OK", &p->initreq, XMIT_CRITICAL, oldsdp, TRUE);
      ast_set_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   }
   sip_pvt_unlock(p);
   return res;
}

void sip_auth_headers	(	enum sip_auth_type	code,
		char **	header,
		char **	respheader
	)

return the request and response header for a 401 or 407 code

Definition at line 15654 of file chan_sip.c.

References ast_verbose().

Referenced by check_auth(), do_message_auth(), do_proxy_auth(), do_register_auth(), sip_report_security_event(), and transmit_request_with_auth().

{
   if (code == WWW_AUTH) {       /* 401 */
      *header = "WWW-Authenticate";
      *respheader = "Authorization";
   } else if (code == PROXY_AUTH) { /* 407 */
      *header = "Proxy-Authenticate";
      *respheader = "Proxy-Authorization";
   } else {
      ast_verbose("-- wrong response code %d\n", code);
      *header = *respheader = "Invalid";
   }
}

static int sip_call	(	struct ast_channel *	ast,
		const char *	dest,
		int	timeout
	)		[static]

Initiate SIP call from PBX used from the dial() application.

Definition at line 6255 of file chan_sip.c.

References ao2_t_ref, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_USER_BUSY, ast_cc_get_monitor_by_recall_core_id(), ast_cc_is_recall(), ast_channel_caller(), ast_channel_get_device_name(), ast_channel_hangupcause_set(), AST_CHANNEL_NAME, ast_channel_name(), ast_channel_queue_connected_line_update(), ast_channel_tech_pvt(), ast_channel_varshead(), ast_clear_flag, AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, ast_copy_string(), ast_debug, ast_format_cap_has_type(), AST_FORMAT_TYPE_AUDIO, AST_LIST_TRAVERSE, ast_log(), ast_party_connected_line_init(), ast_party_id_presentation(), AST_PRES_ALLOWED, AST_PRES_RESTRICTION, ast_rtp_instance_available_formats(), AST_SCHED_REPLACE_UNREF, ast_set_flag, ast_set_party_id_all(), AST_STATE_DOWN, AST_STATE_RESERVED, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_var_name(), ast_var_value(), auto_congest(), cid_name, connected, ast_party_connected_line::id, ast_set_party_connected_line::id, LOG_WARNING, ast_party_id::name, ast_set_party_id::name, ast_party_id::number, ast_set_party_id::number, ast_party_name::presentation, ast_party_number::presentation, ast_set_party_connected_line::priv, ast_cc_monitor::private_data, setup_srtp(), sip_pvt_lock, sip_pvt_unlock, ast_party_connected_line::source, ast_party_name::str, ast_party_number::str, ast_party_id::tag, transmit_invite(), update_call_counter(), ast_party_name::valid, and ast_party_number::valid.

{
   int res;
   struct sip_pvt *p = ast_channel_tech_pvt(ast);  /* chan is locked, so the reference cannot go away */
   struct varshead *headp;
   struct ast_var_t *current;
   const char *referer = NULL;   /* SIP referrer */
   int cc_core_id;
   char uri[SIPBUFSIZE] = "";

   if ((ast_channel_state(ast) != AST_STATE_DOWN) && (ast_channel_state(ast) != AST_STATE_RESERVED)) {
      ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast_channel_name(ast));
      return -1;
   }

   if (ast_cc_is_recall(ast, &cc_core_id, "SIP")) {
      char device_name[AST_CHANNEL_NAME];
      struct ast_cc_monitor *recall_monitor;
      struct sip_monitor_instance *monitor_instance;
      ast_channel_get_device_name(ast, device_name, sizeof(device_name));
      if ((recall_monitor = ast_cc_get_monitor_by_recall_core_id(cc_core_id, device_name))) {
         monitor_instance = recall_monitor->private_data;
         ast_copy_string(uri, monitor_instance->notify_uri, sizeof(uri));
         ao2_t_ref(recall_monitor, -1, "Got the URI we need so unreffing monitor");
      }
   }

   /* Check whether there is vxml_url, distinctive ring variables */
   headp=ast_channel_varshead(ast);
   AST_LIST_TRAVERSE(headp, current, entries) {
      /* Check whether there is a VXML_URL variable */
      if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
         p->options->vxml_url = ast_var_value(current);
      } else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
         p->options->uri_options = ast_var_value(current);
      } else if (!p->options->addsipheaders && !strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
         /* Check whether there is a variable with a name starting with SIPADDHEADER */
         p->options->addsipheaders = 1;
      } else if (!strcasecmp(ast_var_name(current), "SIPFROMDOMAIN")) {
         ast_string_field_set(p, fromdomain, ast_var_value(current));
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER")) {
         /* This is a transferred call */
         p->options->transfer = 1;
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REFERER")) {
         /* This is the referrer */
         referer = ast_var_value(current);
      } else if (!strcasecmp(ast_var_name(current), "SIPTRANSFER_REPLACES")) {
         /* We're replacing a call. */
         p->options->replaces = ast_var_value(current);
      } else if (!strcasecmp(ast_var_name(current), "SIP_MAX_FORWARDS")) {
         if (sscanf(ast_var_value(current), "%30d", &(p->maxforwards)) != 1) {
            ast_log(LOG_WARNING, "The SIP_MAX_FORWARDS channel variable is not a valid integer.\n");
         }
      }
   }

   /* Check to see if we should try to force encryption */
   if (p->req_secure_signaling && p->socket.type != SIP_TRANSPORT_TLS) {
      ast_log(LOG_WARNING, "Encrypted signaling is required\n");
      ast_channel_hangupcause_set(ast, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
      return -1;
   }

   if (ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP)) {
      if (ast_test_flag(&p->flags[0], SIP_REINVITE)) {
         ast_debug(1, "Direct media not possible when using SRTP, ignoring canreinvite setting\n");
         ast_clear_flag(&p->flags[0], SIP_REINVITE);
      }

      if (p->rtp && !p->srtp && setup_srtp(&p->srtp) < 0) {
         ast_log(LOG_WARNING, "SRTP audio setup failed\n");
         return -1;
      }

      if (p->vrtp && !p->vsrtp && setup_srtp(&p->vsrtp) < 0) {
         ast_log(LOG_WARNING, "SRTP video setup failed\n");
         return -1;
      }

      if (p->trtp && !p->tsrtp && setup_srtp(&p->tsrtp) < 0) {
         ast_log(LOG_WARNING, "SRTP text setup failed\n");
         return -1;
      }
   }

   res = 0;
   ast_set_flag(&p->flags[0], SIP_OUTGOING);

   /* T.38 re-INVITE FAX detection should never be done for outgoing calls,
    * so ensure it is disabled.
    */
   ast_clear_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_T38);

   if (p->options->transfer) {
      char buf[SIPBUFSIZE/2];

      if (referer) {
         if (sipdebug)
            ast_debug(3, "Call for %s transferred by %s\n", p->username, referer);
         snprintf(buf, sizeof(buf)-1, "-> %s (via %s)", p->cid_name, referer);
      } else
         snprintf(buf, sizeof(buf)-1, "-> %s", p->cid_name);
      ast_string_field_set(p, cid_name, buf);
   }
   ast_debug(1, "Outgoing Call for %s\n", p->username);

   res = update_call_counter(p, INC_CALL_RINGING);

   if (res == -1) {
      ast_channel_hangupcause_set(ast, AST_CAUSE_USER_BUSY);
      return res;
   }
   p->callingpres = ast_party_id_presentation(&ast_channel_caller(ast)->id);
   ast_rtp_instance_available_formats(p->rtp, p->caps, p->prefcaps, p->jointcaps);
   p->jointnoncodeccapability = p->noncodeccapability;

   /* If there are no audio formats left to offer, punt */
   if (!(ast_format_cap_has_type(p->jointcaps, AST_FORMAT_TYPE_AUDIO))) {
      ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
      res = -1;
   } else {
      int xmitres;
      struct ast_party_connected_line connected;
      struct ast_set_party_connected_line update_connected;

      sip_pvt_lock(p);

      /* Supply initial connected line information if available. */
      memset(&update_connected, 0, sizeof(update_connected));
      ast_party_connected_line_init(&connected);
      if (!ast_strlen_zero(p->cid_num)
         || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
         update_connected.id.number = 1;
         connected.id.number.valid = 1;
         connected.id.number.str = (char *) p->cid_num;
         connected.id.number.presentation = p->callingpres;
      }
      if (!ast_strlen_zero(p->cid_name)
         || (p->callingpres & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED) {
         update_connected.id.name = 1;
         connected.id.name.valid = 1;
         connected.id.name.str = (char *) p->cid_name;
         connected.id.name.presentation = p->callingpres;
      }
      if (update_connected.id.number || update_connected.id.name) {
         /* Invalidate any earlier private connected id representation */
         ast_set_party_id_all(&update_connected.priv);

         connected.id.tag = (char *) p->cid_tag;
         connected.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
         ast_channel_queue_connected_line_update(ast, &connected, &update_connected);
      }

      xmitres = transmit_invite(p, SIP_INVITE, 1, 2, uri);
      if (xmitres == XMIT_ERROR) {
         sip_pvt_unlock(p);
         return -1;
      }
      p->invitestate = INV_CALLING;

      /* Initialize auto-congest time */
      AST_SCHED_REPLACE_UNREF(p->initid, sched, p->timer_b, auto_congest, p,
                        dialog_unref(_data, "dialog ptr dec when SCHED_REPLACE del op succeeded"),
                        dialog_unref(p, "dialog ptr dec when SCHED_REPLACE add failed"),
                        dialog_ref(p, "dialog ptr inc when SCHED_REPLACE add succeeded") );
      sip_pvt_unlock(p);
   }
   return res;
}

int sip_cancel_destroy

(

struct sip_pvt *

p

)

Cancel destruction of SIP dialog. Be careful as this also absorbs the reference - if you call it from within the scheduler, this might be the last reference.

Definition at line 4492 of file chan_sip.c.

References append_history, and AST_SCHED_DEL_UNREF.

Referenced by handle_request_invite(), handle_request_subscribe(), handle_response(), handle_response_invite(), register_verify(), sip_hangup(), and sip_scheddestroy().

{
   if (p->final_destruction_scheduled) {
      return 0;
   }

   if (p->autokillid > -1) {
      append_history(p, "CancelDestroy", "");
      AST_SCHED_DEL_UNREF(sched, p->autokillid, dialog_unref(p, "remove ref for autokillid"));
   }
   return 0;
}

static void sip_cc_agent_destructor

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1918 of file chan_sip.c.

References ast_free, ast_test_flag, ast_cc_agent::private_data, sip_cc_agent_stop_offer_timer(), sip_pvt_lock, sip_pvt_unlock, and transmit_response().

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

   if (!agent_pvt) {
      /* The agent constructor probably failed. */
      return;
   }

   sip_cc_agent_stop_offer_timer(agent);
   if (agent_pvt->subscribe_pvt) {
      sip_pvt_lock(agent_pvt->subscribe_pvt);
      if (!ast_test_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
         /* If we haven't sent a 200 OK for the SUBSCRIBE dialog yet, then we need to send a response letting
          * the subscriber know something went wrong
          */
         transmit_response(agent_pvt->subscribe_pvt, "500 Internal Server Error", &agent_pvt->subscribe_pvt->initreq);
      }
      sip_pvt_unlock(agent_pvt->subscribe_pvt);
      agent_pvt->subscribe_pvt = dialog_unref(agent_pvt->subscribe_pvt, "SIP CC agent destructor: Remove ref to subscription");
   }
   ast_free(agent_pvt);
}

static int sip_cc_agent_init	(	struct ast_cc_agent *	agent,
		struct ast_channel *	chan
	)		[static]

Definition at line 1805 of file chan_sip.c.

References ast_assert, ast_calloc, ast_channel_tech(), ast_channel_tech_pvt(), ast_copy_string(), ast_set_flag, ast_cc_agent::private_data, sip_pvt_lock, sip_pvt_unlock, and type.

{
   struct sip_cc_agent_pvt *agent_pvt = ast_calloc(1, sizeof(*agent_pvt));
   struct sip_pvt *call_pvt = ast_channel_tech_pvt(chan);

   if (!agent_pvt) {
      return -1;
   }

   ast_assert(!strcmp(ast_channel_tech(chan)->type, "SIP"));

   ast_copy_string(agent_pvt->original_callid, call_pvt->callid, sizeof(agent_pvt->original_callid));
   ast_copy_string(agent_pvt->original_exten, call_pvt->exten, sizeof(agent_pvt->original_exten));
   agent_pvt->offer_timer_id = -1;
   agent->private_data = agent_pvt;
   sip_pvt_lock(call_pvt);
   ast_set_flag(&call_pvt->flags[0], SIP_OFFER_CC);
   sip_pvt_unlock(call_pvt);
   return 0;
}

static int sip_cc_agent_recall

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1898 of file chan_sip.c.

References ast_cc_agent_caller_busy(), ast_cc_agent::core_id, ast_cc_agent::device_name, ast_cc_agent::private_data, sip_pvt_lock, sip_pvt_unlock, and transmit_cc_notify().

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   /* If we have received a PUBLISH beforehand stating that the caller in question
    * is not available, we can save ourself a bit of effort here and just report
    * the caller as busy
    */
   if (!agent_pvt->is_available) {
      return ast_cc_agent_caller_busy(agent->core_id, "Caller %s is busy, reporting to the core",
            agent->device_name);
   }
   /* Otherwise, we transmit a NOTIFY to the caller and await either
    * a PUBLISH or an INVITE
    */
   sip_pvt_lock(agent_pvt->subscribe_pvt);
   transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_READY);
   sip_pvt_unlock(agent_pvt->subscribe_pvt);
   return 0;
}

static void sip_cc_agent_respond	(	struct ast_cc_agent *	agent,
		enum ast_cc_agent_response_reason	reason
	)		[static]

Definition at line 1854 of file chan_sip.c.

References AST_CC_AGENT_RESPONSE_SUCCESS, ast_set_flag, ast_strlen_zero(), ast_cc_agent::private_data, sip_pvt_lock, sip_pvt_unlock, transmit_cc_notify(), transmit_response(), and TRUE.

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

   sip_pvt_lock(agent_pvt->subscribe_pvt);
   ast_set_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
   if (reason == AST_CC_AGENT_RESPONSE_SUCCESS || !ast_strlen_zero(agent_pvt->notify_uri)) {
      /* The second half of this if statement may be a bit hard to grasp,
       * so here's an explanation. When a subscription comes into
       * chan_sip, as long as it is not malformed, it will be passed
       * to the CC core. If the core senses an out-of-order state transition,
       * then the core will call this callback with the "reason" set to a
       * failure condition.
       * However, an out-of-order state transition will occur during a resubscription
       * for CC. In such a case, we can see that we have already generated a notify_uri
       * and so we can detect that this isn't a *real* failure. Rather, it is just
       * something the core doesn't recognize as a legitimate SIP state transition.
       * Thus we respond with happiness and flowers.
       */
      transmit_response(agent_pvt->subscribe_pvt, "200 OK", &agent_pvt->subscribe_pvt->initreq);
      transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_QUEUED);
   } else {
      transmit_response(agent_pvt->subscribe_pvt, "500 Internal Error", &agent_pvt->subscribe_pvt->initreq);
   }
   sip_pvt_unlock(agent_pvt->subscribe_pvt);
   agent_pvt->is_available = TRUE;
}

static int sip_cc_agent_start_monitoring

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1889 of file chan_sip.c.

{
   /* To start monitoring just means to wait for an incoming PUBLISH
    * to tell us that the caller has become available again. No special
    * action is needed
    */
   return 0;
}

static int sip_cc_agent_start_offer_timer

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1836 of file chan_sip.c.

References ast_get_cc_offer_timer(), ast_sched_add(), ast_cc_agent::cc_params, ast_cc_agent::private_data, and sip_offer_timer_expire().

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   int when;

   when = ast_get_cc_offer_timer(agent->cc_params) * 1000;
   agent_pvt->offer_timer_id = ast_sched_add(sched, when, sip_offer_timer_expire, agent);
   return 0;
}

static int sip_cc_agent_status_request

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1882 of file chan_sip.c.

References ast_cc_agent_status_response(), AST_DEVICE_INUSE, AST_DEVICE_NOT_INUSE, ast_cc_agent::core_id, and ast_cc_agent::private_data.

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   enum ast_device_state state = agent_pvt->is_available ? AST_DEVICE_NOT_INUSE : AST_DEVICE_INUSE;
   return ast_cc_agent_status_response(agent->core_id, state);
}

static int sip_cc_agent_stop_offer_timer

(

struct ast_cc_agent *

agent

)

[static]

Definition at line 1846 of file chan_sip.c.

References AST_SCHED_DEL, and ast_cc_agent::private_data.

Referenced by sip_cc_agent_destructor().

{
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

   AST_SCHED_DEL(sched, agent_pvt->offer_timer_id);
   return 0;
}

static int sip_cc_monitor_cancel_available_timer	(	struct ast_cc_monitor *	monitor,
		int *	sched_id
	)		[static]

Definition at line 2157 of file chan_sip.c.

References ao2_t_ref, and AST_SCHED_DEL.

{
   if (*sched_id != -1) {
      AST_SCHED_DEL(sched, *sched_id);
      ao2_t_ref(monitor, -1, "Removing scheduler's reference to the monitor");
   }
   return 0;
}

static void sip_cc_monitor_destructor

(

void *

private_data

)

[static]

Definition at line 2166 of file chan_sip.c.

References ao2_unlink, ast_module_unref(), and ast_module_info::self.

{
   struct sip_monitor_instance *monitor_instance = private_data;
   ao2_unlink(sip_monitor_instances, monitor_instance);
   ast_module_unref(ast_module_info->self);
}

static int sip_cc_monitor_request_cc	(	struct ast_cc_monitor *	monitor,
		int *	available_timer_id
	)		[static]

Definition at line 2024 of file chan_sip.c.

References ao2_t_ref, ast_cc_available_timer_expire(), AST_CC_CCBS, ast_get_ccbs_available_timer(), ast_get_ccnr_available_timer(), ast_sched_add(), ast_set_flag, ast_sip_ouraddrfor(), ast_cc_interface::config_params, create_addr(), FALSE, ast_cc_monitor::interface, ast_cc_monitor::private_data, service, ast_cc_monitor::service_offered, sip_alloc(), sip_pvt_lock, sip_pvt_unlock, and transmit_invite().

{
   struct sip_monitor_instance *monitor_instance = monitor->private_data;
   enum ast_cc_service_type service = monitor->service_offered;
   int when;

   if (!monitor_instance) {
      return -1;
   }

   if (!(monitor_instance->subscription_pvt = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, NULL))) {
      return -1;
   }

   when = service == AST_CC_CCBS ? ast_get_ccbs_available_timer(monitor->interface->config_params) :
      ast_get_ccnr_available_timer(monitor->interface->config_params);

   sip_pvt_lock(monitor_instance->subscription_pvt);
   ast_set_flag(&monitor_instance->subscription_pvt->flags[0], SIP_OUTGOING);
   create_addr(monitor_instance->subscription_pvt, monitor_instance->peername, 0, 1);
   ast_sip_ouraddrfor(&monitor_instance->subscription_pvt->sa, &monitor_instance->subscription_pvt->ourip, monitor_instance->subscription_pvt);
   monitor_instance->subscription_pvt->subscribed = CALL_COMPLETION;
   monitor_instance->subscription_pvt->expiry = when;

   transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 2, monitor_instance->subscribe_uri);
   sip_pvt_unlock(monitor_instance->subscription_pvt);

   ao2_t_ref(monitor, +1, "Adding a ref to the monitor for the scheduler");
   *available_timer_id = ast_sched_add(sched, when * 1000, ast_cc_available_timer_expire, monitor);
   return 0;
}

static int sip_cc_monitor_suspend

(

struct ast_cc_monitor *

monitor

)

[static]

Definition at line 2082 of file chan_sip.c.

References ao2_ref, ast_calloc, ast_log(), ast_strlen_zero(), construct_pidf_body(), ast_cc_monitor::core_id, create_epa_entry(), LOG_WARNING, ast_cc_monitor::private_data, and transmit_publish().

{
   struct sip_monitor_instance *monitor_instance = monitor->private_data;
   enum sip_publish_type publish_type;
   struct cc_epa_entry *cc_entry;

   if (!monitor_instance) {
      return -1;
   }

   if (!monitor_instance->suspension_entry) {
      /* We haven't yet allocated the suspension entry, so let's give it a shot */
      if (!(monitor_instance->suspension_entry = create_epa_entry("call-completion", monitor_instance->peername))) {
         ast_log(LOG_WARNING, "Unable to allocate sip EPA entry for call-completion\n");
         ao2_ref(monitor_instance, -1);
         return -1;
      }
      if (!(cc_entry = ast_calloc(1, sizeof(*cc_entry)))) {
         ast_log(LOG_WARNING, "Unable to allocate space for instance data of EPA entry for call-completion\n");
         ao2_ref(monitor_instance, -1);
         return -1;
      }
      cc_entry->core_id = monitor->core_id;
      monitor_instance->suspension_entry->instance_data = cc_entry;
      publish_type = SIP_PUBLISH_INITIAL;
   } else {
      publish_type = SIP_PUBLISH_MODIFY;
      cc_entry = monitor_instance->suspension_entry->instance_data;
   }

   cc_entry->current_state = CC_CLOSED;

   if (ast_strlen_zero(monitor_instance->notify_uri)) {
      /* If we have no set notify_uri, then what this means is that we have
       * not received a NOTIFY from this destination stating that he is
       * currently available.
       *
       * This situation can arise when the core calls the suspend callbacks
       * of multiple destinations. If one of the other destinations aside
       * from this one notified Asterisk that he is available, then there
       * is no reason to take any suspension action on this device. Rather,
       * we should return now and if we receive a NOTIFY while monitoring
       * is still "suspended" then we can immediately respond with the
       * proper PUBLISH to let this endpoint know what is going on.
       */
      return 0;
   }
   construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
   return transmit_publish(monitor_instance->suspension_entry, publish_type, monitor_instance->notify_uri);
}

static int sip_cc_monitor_unsuspend

(

struct ast_cc_monitor *

monitor

)

[static]

Definition at line 2133 of file chan_sip.c.

References ast_assert, ast_strlen_zero(), construct_pidf_body(), ast_cc_monitor::private_data, and transmit_publish().

{
   struct sip_monitor_instance *monitor_instance = monitor->private_data;
   struct cc_epa_entry *cc_entry;

   if (!monitor_instance) {
      return -1;
   }

   ast_assert(monitor_instance->suspension_entry != NULL);

   cc_entry = monitor_instance->suspension_entry->instance_data;
   cc_entry->current_state = CC_OPEN;
   if (ast_strlen_zero(monitor_instance->notify_uri)) {
      /* This means we are being asked to unsuspend a call leg we never
       * sent a PUBLISH on. As such, there is no reason to send another
       * PUBLISH at this point either. We can just return instead.
       */
      return 0;
   }
   construct_pidf_body(CC_OPEN, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
   return transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_MODIFY, monitor_instance->notify_uri);
}

static int sip_check_authtimeout

(

time_t

start

)

[static]

Check if the authtimeout has expired.

Parameters:

start

the time when the session started

Return values:

0	the timeout has expired
-1	error

Returns:

the number of milliseconds until the timeout will expire

Definition at line 2642 of file chan_sip.c.

References ast_log(), errno, and LOG_ERROR.

Referenced by _sip_tcp_helper_thread(), sip_tcp_read(), and sip_tls_read().

{
   int timeout;
   time_t now;
   if(time(&now) == -1) {
      ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
      return -1;
   }

   timeout = (authtimeout - (now - start)) * 1000;
   if (timeout < 0) {
      /* we have timed out */
      return 0;
   }

   return timeout;
}

static char * sip_cli_notify	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Cli command to send SIP notify to peer.

Definition at line 21675 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_copy_string(), ast_log(), ast_set_flag, ast_sip_ouraddrfor(), ast_str_append(), ast_str_strlen(), ast_unescape_semicolon(), ast_variable_browse(), ast_variable_new(), build_via(), change_callid_pvt(), CLI_FAILURE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_notify(), create_addr(), dialog_unlink_all(), ast_cli_args::fd, ast_cli_args::line, LOG_WARNING, ast_cli_args::n, ast_variable::name, ast_variable::next, ast_cli_args::pos, sip_alloc(), sip_notify_alloc(), sip_scheddestroy(), transmit_invite(), ast_cli_entry::usage, ast_variable::value, var, and ast_cli_args::word.

{
   struct ast_variable *varlist;
   int i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip notify";
      e->usage =
         "Usage: sip notify <type> <peer> [<peer>...]\n"
         "       Send a NOTIFY message to a SIP peer or peers\n"
         "       Message types are defined in sip_notify.conf\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_notify(a->line, a->word, a->pos, a->n);
   }

   if (a->argc < 4)
      return CLI_SHOWUSAGE;

   if (!notify_types) {
      ast_cli(a->fd, "No %s file found, or no types listed there\n", notify_config);
      return CLI_FAILURE;
   }

   varlist = ast_variable_browse(notify_types, a->argv[2]);

   if (!varlist) {
      ast_cli(a->fd, "Unable to find notify type '%s'\n", a->argv[2]);
      return CLI_FAILURE;
   }

   for (i = 3; i < a->argc; i++) {
      struct sip_pvt *p;
      char buf[512];
      struct ast_variable *header, *var;

      if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, NULL))) {
         ast_log(LOG_WARNING, "Unable to build sip pvt data for notify (memory/socket error)\n");
         return CLI_FAILURE;
      }

      if (create_addr(p, a->argv[i], NULL, 1)) {
         /* Maybe they're not registered, etc. */
         dialog_unlink_all(p);
         dialog_unref(p, "unref dialog inside for loop" );
         /* sip_destroy(p); */
         ast_cli(a->fd, "Could not create address for '%s'\n", a->argv[i]);
         continue;
      }

      /* Notify is outgoing call */
      ast_set_flag(&p->flags[0], SIP_OUTGOING);
      sip_notify_alloc(p);
      p->notify->headers = header = ast_variable_new("Subscription-State", "terminated", "");

      for (var = varlist; var; var = var->next) {
         ast_copy_string(buf, var->value, sizeof(buf));
         ast_unescape_semicolon(buf);

         if (!strcasecmp(var->name, "Content")) {
            if (ast_str_strlen(p->notify->content))
               ast_str_append(&p->notify->content, 0, "\r\n");
            ast_str_append(&p->notify->content, 0, "%s", buf);
         } else if (!strcasecmp(var->name, "Content-Length")) {
            ast_log(LOG_WARNING, "it is not necessary to specify Content-Length in sip_notify.conf, ignoring\n");
         } else {
            header->next = ast_variable_new(var->name, buf, "");
            header = header->next;
         }
      }

      /* Now that we have the peer's address, set our ip and change callid */
      ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
      build_via(p);

      change_callid_pvt(p, NULL);

      ast_cli(a->fd, "Sending NOTIFY of type '%s' to '%s'\n", a->argv[2], a->argv[i]);
      sip_scheddestroy(p, SIP_TRANS_TIMEOUT);
      transmit_invite(p, SIP_NOTIFY, 0, 2, NULL);
      dialog_unref(p, "bump down the count of p since we're done with it.");
   }

   return CLI_SUCCESS;
}

static int sip_debug_test_addr

(

const struct ast_sockaddr *

addr

)

[inline, static]

See if we pass debug IP filter.

Definition at line 3680 of file chan_sip.c.

References ast_sockaddr_cmp(), ast_sockaddr_cmp_addr(), ast_sockaddr_isnull(), ast_sockaddr_port, and debugaddr.

Referenced by check_peer_ok(), handle_request_do(), and sip_debug_test_pvt().

{
   /* Can't debug if sipdebug is not enabled */
   if (!sipdebug) {
      return 0;
   }

   /* A null debug_addr means we'll debug any address */
   if (ast_sockaddr_isnull(&debugaddr)) {
      return 1;
   }

   /* If no port was specified for a debug address, just compare the
    * addresses, otherwise compare the address and port
    */
   if (ast_sockaddr_port(&debugaddr)) {
      return !ast_sockaddr_cmp(&debugaddr, addr);
   } else {
      return !ast_sockaddr_cmp_addr(&debugaddr, addr);
   }
}

static int sip_debug_test_pvt

(

struct sip_pvt *

p

)

[inline, static]

Test PVT for debugging output.

Definition at line 3719 of file chan_sip.c.

References sip_debug_test_addr(), and sip_real_dst().

Referenced by __sip_destroy(), add_sdp(), build_route(), check_via(), do_register_auth(), get_also_info(), get_destination(), get_rdnis(), get_refer_info(), process_sdp(), process_sdp_a_audio(), process_sdp_a_text(), process_sdp_a_video(), receive_message(), retrans_pkt(), send_request(), send_response(), set_destination(), sip_scheddestroy(), sip_sendhtml(), sip_sendtext(), and transmit_register().

{
   if (!sipdebug) {
      return 0;
   }
   return sip_debug_test_addr(sip_real_dst(p));
}

struct sip_pvt* sip_destroy

(

struct sip_pvt *

p

)

[read]

Destroy SIP call structure. Make it return NULL so the caller can do things like foo = sip_destroy(foo); and reduce the chance of bugs due to dangling pointers.

Definition at line 6829 of file chan_sip.c.

References __sip_destroy(), ast_debug, and TRUE.

Referenced by sip_destroy_fn(), and sip_subscribe_mwi_destroy().

{
   ast_debug(3, "Destroying SIP dialog %s\n", p->callid);
   __sip_destroy(p, TRUE, TRUE);
   return NULL;
}

static void sip_destroy_fn

(

void *

p

)

[static]

Definition at line 6819 of file chan_sip.c.

References sip_destroy().

Referenced by sip_alloc().

{
   sip_destroy(p);
}

static void sip_destroy_peer

(

struct sip_peer *

peer

)

[static]

Destroy peer object from memory.

Definition at line 5186 of file chan_sip.c.

References ao2_ref, ao2_t_ref, ast_atomic_fetchadd_int(), ast_cc_config_params_destroy(), ast_debug, ast_format_cap_destroy(), ast_free_acl_list(), ast_rtp_dtls_cfg_free(), ast_string_field_free_memory, ast_test_flag, ast_unref_namedgroups(), ast_variables_destroy(), ast_websocket_unref(), clear_peer_mailboxes(), dialog_unlink_all(), FALSE, and register_peer_exten().

Referenced by sip_destroy_peer_fn().

{
   ast_debug(3, "Destroying SIP peer %s\n", peer->name);

   /*
    * Remove any mailbox event subscriptions for this peer before
    * we destroy anything.  An event subscription callback may be
    * happening right now.
    */
   clear_peer_mailboxes(peer);

   if (peer->outboundproxy) {
      ao2_ref(peer->outboundproxy, -1);
      peer->outboundproxy = NULL;
   }

   /* Delete it, it needs to disappear */
   if (peer->call) {
      dialog_unlink_all(peer->call);
      peer->call = dialog_unref(peer->call, "peer->call is being unset");
   }

   if (peer->mwipvt) {  /* We have an active subscription, delete it */
      dialog_unlink_all(peer->mwipvt);
      peer->mwipvt = dialog_unref(peer->mwipvt, "unreffing peer->mwipvt");
   }

   if (peer->chanvars) {
      ast_variables_destroy(peer->chanvars);
      peer->chanvars = NULL;
   }

   register_peer_exten(peer, FALSE);
   ast_free_acl_list(peer->acl);
   ast_free_acl_list(peer->directmediaacl);
   if (peer->selfdestruct)
      ast_atomic_fetchadd_int(&apeerobjs, -1);
   else if (!ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS) && peer->is_realtime) {
      ast_atomic_fetchadd_int(&rpeerobjs, -1);
      ast_debug(3, "-REALTIME- peer Destroyed. Name: %s. Realtime Peer objects: %d\n", peer->name, rpeerobjs);
   } else
      ast_atomic_fetchadd_int(&speerobjs, -1);
   if (peer->auth) {
      ao2_t_ref(peer->auth, -1, "Removing peer authentication");
      peer->auth = NULL;
   }

   if (peer->socket.tcptls_session) {
      ao2_ref(peer->socket.tcptls_session, -1);
      peer->socket.tcptls_session = NULL;
   } else if (peer->socket.ws_session) {
      ast_websocket_unref(peer->socket.ws_session);
      peer->socket.ws_session = NULL;
   }

   peer->named_callgroups = ast_unref_namedgroups(peer->named_callgroups);
   peer->named_pickupgroups = ast_unref_namedgroups(peer->named_pickupgroups);

   ast_cc_config_params_destroy(peer->cc_params);

   ast_string_field_free_memory(peer);

   peer->caps = ast_format_cap_destroy(peer->caps);

   ast_rtp_dtls_cfg_free(&peer->dtls_cfg);
}

static void sip_destroy_peer_fn

(

void *

peer

)

[static]

Definition at line 5180 of file chan_sip.c.

References sip_destroy_peer().

Referenced by build_peer(), and temp_peer().

{
   sip_destroy_peer(peer);
}

static int sip_devicestate

(

const char *

data

)

[static]

Part of PBX channel interface.

Note:

Return values:---

If we have qualify on and the device is not reachable, regardless of registration state we return AST_DEVICE_UNAVAILABLE

For peers with call limit:

• not registered AST_DEVICE_UNAVAILABLE
• registered, no call AST_DEVICE_NOT_INUSE
• registered, active calls AST_DEVICE_INUSE
• registered, call limit reached AST_DEVICE_BUSY
• registered, onhold AST_DEVICE_ONHOLD
• registered, ringing AST_DEVICE_RINGING

For peers without call limit:

• not registered AST_DEVICE_UNAVAILABLE
• registered AST_DEVICE_NOT_INUSE
• fixed IP (!dynamic) AST_DEVICE_NOT_INUSE

Peers that does not have a known call and can't be reached by OPTIONS

• unreachable AST_DEVICE_UNAVAILABLE

If we return AST_DEVICE_UNKNOWN, the device state engine will try to find out a state by walking the channel list.

The queue system (app_queue.c) treats a member as "active" if devicestate is != AST_DEVICE_UNAVAILBALE && != AST_DEVICE_INVALID

When placing a call to the queue member, queue system sets a member to busy if != AST_DEVICE_NOT_INUSE and != AST_DEVICE_UNKNOWN

Definition at line 29730 of file chan_sip.c.

References ast_debug, AST_DEVICE_BUSY, AST_DEVICE_INUSE, AST_DEVICE_INVALID, AST_DEVICE_NOT_INUSE, AST_DEVICE_ONHOLD, AST_DEVICE_RINGING, AST_DEVICE_RINGINUSE, AST_DEVICE_UNAVAILABLE, ast_sockaddr_isnull(), FALSE, sip_find_peer(), sip_unref_peer(), and TRUE.

{
   char *host;
   char *tmp;
   struct sip_peer *p;

   int res = AST_DEVICE_INVALID;

   /* make sure data is not null. Maybe unnecessary, but better be safe */
   host = ast_strdupa(data ? data : "");
   if ((tmp = strchr(host, '@')))
      host = tmp + 1;

   ast_debug(3, "Checking device state for peer %s\n", host);

   /* If sip_find_peer asks for a realtime peer, then this breaks rtautoclear.  This
    * is because when a peer tries to autoexpire, the last thing it does is to
    * queue up an event telling the system that the devicestate has changed
    * (presumably to unavailable).  If we ask for a realtime peer here, this would
    * load it BACK into memory, thus defeating the point of trying to clear dead
    * hosts out of memory.
    */
   if ((p = sip_find_peer(host, NULL, FALSE, FINDALLDEVICES, TRUE, 0))) {
      if (!(ast_sockaddr_isnull(&p->addr) && ast_sockaddr_isnull(&p->defaddr))) {
         /* we have an address for the peer */

         /* Check status in this order
            - Hold
            - Ringing
            - Busy (enforced only by call limit)
            - Inuse (we have a call)
            - Unreachable (qualify)
            If we don't find any of these state, report AST_DEVICE_NOT_INUSE
            for registered devices */

         if (p->onhold)
            /* First check for hold or ring states */
            res = AST_DEVICE_ONHOLD;
         else if (p->ringing) {
            if (p->ringing == p->inuse)
               res = AST_DEVICE_RINGING;
            else
               res = AST_DEVICE_RINGINUSE;
         } else if (p->call_limit && (p->inuse == p->call_limit))
            /* check call limit */
            res = AST_DEVICE_BUSY;
         else if (p->call_limit && p->busy_level && p->inuse >= p->busy_level)
            /* We're forcing busy before we've reached the call limit */
            res = AST_DEVICE_BUSY;
         else if (p->call_limit && p->inuse)
            /* Not busy, but we do have a call */
            res = AST_DEVICE_INUSE;
         else if (p->maxms && ((p->lastms > p->maxms) || (p->lastms < 0)))
            /* We don't have a call. Are we reachable at all? Requires qualify= */
            res = AST_DEVICE_UNAVAILABLE;
         else  /* Default reply if we're registered and have no other data */
            res = AST_DEVICE_NOT_INUSE;
      } else {
         /* there is no address, it's unavailable */
         res = AST_DEVICE_UNAVAILABLE;
      }
      sip_unref_peer(p, "sip_unref_peer, from sip_devicestate, release ref from sip_find_peer");
   }

   return res;
}

void sip_digest_parser	(	char *	c,
		struct digestkeys *	keys
	)

Takes the digest response and parses it.

Definition at line 16369 of file chan_sip.c.

References ast_skip_blanks().

Referenced by check_auth(), and sip_report_security_event().

{
        struct digestkeys *i = i;

        while(c && *(c = ast_skip_blanks(c)) ) { /* lookup for keys */
                for (i = keys; i->key != NULL; i++) {
                        const char *separator = ",";    /* default */

                        if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
                                continue;
                        }
                        /* Found. Skip keyword, take text in quotes or up to the separator. */
                        c += strlen(i->key);
                        if (*c == '"') { /* in quotes. Skip first and look for last */
                                c++;
                                separator = "\"";
                        }
                        i->s = c;
                        strsep(&c, separator);
                        break;
                }
                if (i->key == NULL) { /* not found, jump after space or comma */
         strsep(&c, " ,");
      }
        }
}

static char * sip_do_debug	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Turn on SIP debugging (CLI command)

Note:

this can be a special debug command - "sip debug text" or something

Definition at line 21632 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_peer(), debugaddr, ast_cli_args::fd, ast_cli_args::n, ast_cli_args::pos, sip_do_debug_ip(), sip_do_debug_peer(), ast_cli_entry::usage, and ast_cli_args::word.

{
   int oldsipdebug = sipdebug & sip_debug_console;
   const char *what;

   if (cmd == CLI_INIT) {
      e->command = "sip set debug {on|off|ip|peer}";
      e->usage =
         "Usage: sip set debug {off|on|ip addr[:port]|peer peername}\n"
         "       Globally disables dumping of SIP packets,\n"
         "       or enables it either globally or for a (single)\n"
         "       IP address or registered peer.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE) {
      if (a->pos == 4 && !strcasecmp(a->argv[3], "peer"))
         return complete_sip_peer(a->word, a->n, 0);
      return NULL;
        }

   what = a->argv[e->args-1];      /* guaranteed to exist */
   if (a->argc == e->args) {       /* on/off */
      if (!strcasecmp(what, "on")) {
         sipdebug |= sip_debug_console;
         sipdebug_text = 1;   /*! \note this can be a special debug command - "sip debug text" or something */
         memset(&debugaddr, 0, sizeof(debugaddr));
         ast_cli(a->fd, "SIP Debugging %senabled\n", oldsipdebug ? "re-" : "");
         return CLI_SUCCESS;
      } else if (!strcasecmp(what, "off")) {
         sipdebug &= ~sip_debug_console;
         sipdebug_text = 0;
         ast_cli(a->fd, "SIP Debugging Disabled\n");
         return CLI_SUCCESS;
      }
   } else if (a->argc == e->args +1) {/* ip/peer */
      if (!strcasecmp(what, "ip"))
         return sip_do_debug_ip(a->fd, a->argv[e->args]);
      else if (!strcasecmp(what, "peer"))
         return sip_do_debug_peer(a->fd, a->argv[e->args]);
   }
   return CLI_SHOWUSAGE;   /* default, failure */
}

static char * sip_do_debug_ip	(	int	fd,
		const char *	arg
	)		[static]

Enable SIP Debugging for a single IP.

Definition at line 21600 of file chan_sip.c.

References ast_cli(), ast_sockaddr_resolve_first_af(), ast_sockaddr_stringify_addr(), CLI_SHOWUSAGE, CLI_SUCCESS, and debugaddr.

Referenced by sip_do_debug().

{
   if (ast_sockaddr_resolve_first_af(&debugaddr, arg, 0, 0)) {
      return CLI_SHOWUSAGE;
   }

   ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_sockaddr_stringify_addr(&debugaddr));
   sipdebug |= sip_debug_console;

   return CLI_SUCCESS;
}

static char * sip_do_debug_peer	(	int	fd,
		const char *	arg
	)		[static]

Turn on SIP debugging for a given peer.

Definition at line 21613 of file chan_sip.c.

References ast_cli(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_stringify_addr(), CLI_SUCCESS, debugaddr, FALSE, sip_find_peer(), sip_unref_peer(), and TRUE.

Referenced by sip_do_debug().

{
   struct sip_peer *peer = sip_find_peer(arg, NULL, TRUE, FINDPEERS, FALSE, 0);
   if (!peer) {
      ast_cli(fd, "No such peer '%s'\n", arg);
   } else if (ast_sockaddr_isnull(&peer->addr)) {
      ast_cli(fd, "Unable to get IP address of peer '%s'\n", arg);
   } else {
      ast_sockaddr_copy(&debugaddr, &peer->addr);
      ast_cli(fd, "SIP Debugging Enabled for IP: %s\n", ast_sockaddr_stringify_addr(&debugaddr));
      sipdebug |= sip_debug_console;
   }
   if (peer) {
      sip_unref_peer(peer, "sip_do_debug_peer: sip_unref_peer, from sip_find_peer call");
   }
   return CLI_SUCCESS;
}

static int sip_do_reload

(

enum channelreloadreason

reason

)

[static]

Reload module.

Definition at line 33346 of file chan_sip.c.

References ast_debug, ast_sched_dump(), reload_config(), sip_keepalive_all_peers(), sip_poke_all_peers(), sip_send_all_mwi_subscriptions(), sip_send_all_registers(), and unlink_marked_peers_from_tables().

Referenced by do_monitor().

{
   time_t start_poke, end_poke;

   reload_config(reason);
   ast_sched_dump(sched);

   start_poke = time(0);
   /* Prune peers who still are supposed to be deleted */
   unlink_marked_peers_from_tables();

   ast_debug(4, "--------------- Done destroying pruned peers\n");

   /* Send qualify (OPTIONS) to all peers */
   sip_poke_all_peers();

   /* Send keepalive to all peers */
   sip_keepalive_all_peers();

   /* Register with all services */
   sip_send_all_registers();

   sip_send_all_mwi_subscriptions();

   end_poke = time(0);

   ast_debug(4, "do_reload finished. peer poke/prune reg contact time = %d sec.\n", (int)(end_poke-start_poke));

   ast_debug(4, "--------------- SIP reload done\n");

   return 0;
}

static int sip_dtmfmode	(	struct ast_channel *	chan,
		const char *	data
	)		[static]

Set the DTMFmode for an outbound SIP call (application)

Definition at line 32975 of file chan_sip.c.

References ast_channel_lock, ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_unlock, ast_clear_flag, ast_log(), AST_RTP_DTMF, ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_DTMF, ast_set_flag, ast_test_flag, disable_dsp_detect(), enable_dsp_detect(), LOG_WARNING, sip_pvt_lock, and sip_pvt_unlock.

Referenced by load_module().

{
   struct sip_pvt *p;
   const char *mode = data;

   if (!data) {
      ast_log(LOG_WARNING, "This application requires the argument: info, inband, rfc2833\n");
      return 0;
   }
   ast_channel_lock(chan);
   if (!IS_SIP_TECH(ast_channel_tech(chan))) {
      ast_log(LOG_WARNING, "Call this application only on SIP incoming calls\n");
      ast_channel_unlock(chan);
      return 0;
   }
   p = ast_channel_tech_pvt(chan);
   if (!p) {
      ast_channel_unlock(chan);
      return 0;
   }
   sip_pvt_lock(p);
   if (!strcasecmp(mode, "info")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_INFO);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "shortinfo")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_SHORTINFO);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "rfc2833")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
      p->jointnoncodeccapability |= AST_RTP_DTMF;
   } else if (!strcasecmp(mode, "inband")) {
      ast_clear_flag(&p->flags[0], SIP_DTMF);
      ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
      p->jointnoncodeccapability &= ~AST_RTP_DTMF;
   } else {
      ast_log(LOG_WARNING, "I don't know about this dtmf mode: %s\n", mode);
   }
   if (p->rtp)
      ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_DTMF, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
   if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
       (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      enable_dsp_detect(p);
   } else {
      disable_dsp_detect(p);
   }
   sip_pvt_unlock(p);
   ast_channel_unlock(chan);
   return 0;
}

static void sip_dump_history

(

struct sip_pvt *

dialog

)

[static]

Dump SIP history to debug log file at end of lifespan for SIP dialog.

Definition at line 21379 of file chan_sip.c.

References ast_debug, AST_LIST_TRAVERSE, ast_log(), LOG_NOTICE, and option_debug.

Referenced by __sip_destroy().

{
   int x = 0;
   struct sip_history *hist;
   static int errmsg = 0;

   if (!dialog) {
      return;
   }

   if (!option_debug && !sipdebug) {
      if (!errmsg) {
         ast_log(LOG_NOTICE, "You must have debugging enabled (SIP or Asterisk) in order to dump SIP history.\n");
         errmsg = 1;
      }
      return;
   }

   ast_debug(1, "\n---------- SIP HISTORY for '%s' \n", dialog->callid);
   if (dialog->subscribed) {
      ast_debug(1, "  * Subscription\n");
   } else {
      ast_debug(1, "  * SIP Call\n");
   }
   if (dialog->history) {
      AST_LIST_TRAVERSE(dialog->history, hist, list)
         ast_debug(1, "  %-3.3d. %s\n", ++x, hist->event);
   }
   if (!x) {
      ast_debug(1, "Call '%s' has no history\n", dialog->callid);
   }
   ast_debug(1, "\n---------- END SIP HISTORY for '%s' \n", dialog->callid);
}

static int sip_epa_register

(

const struct epa_static_data *

static_data

)

[static]

Definition at line 921 of file chan_sip.c.

References ast_calloc, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, and AST_LIST_UNLOCK.

Referenced by load_module().

{
   struct epa_backend *backend = ast_calloc(1, sizeof(*backend));

   if (!backend) {
      return -1;
   }

   backend->static_data = static_data;

   AST_LIST_LOCK(&epa_static_data_list);
   AST_LIST_INSERT_TAIL(&epa_static_data_list, backend, next);
   AST_LIST_UNLOCK(&epa_static_data_list);
   return 0;
}

static void sip_epa_unregister_all

(

void

)

[static]

Definition at line 937 of file chan_sip.c.

References ast_free, AST_LIST_LOCK, AST_LIST_REMOVE_HEAD, and AST_LIST_UNLOCK.

Referenced by unload_module().

{
   struct epa_backend *backend;

   AST_LIST_LOCK(&epa_static_data_list);
   while ((backend = AST_LIST_REMOVE_HEAD(&epa_static_data_list, next))) {
      ast_free(backend);
   }
   AST_LIST_UNLOCK(&epa_static_data_list);
}

struct sip_peer* sip_find_peer	(	const char *	peer,
		struct ast_sockaddr *	addr,
		int	realtime,
		int	which_objects,
		int	devstate_only,
		int	transport
	)		[read]

Locate device by name or ip address.

Parameters:

peer,sin,realtime,devstate_only,transport
which_objects	Define which objects should be matched when doing a lookup by name. Valid options are FINDUSERS, FINDPEERS, or FINDALLDEVICES. Note that this option is not used at all when doing a lookup by IP.

This is used on find matching device on name or ip/port. If the device was declared as type=peer, we don't match on peer name on incoming INVITEs.

Note:

Avoid using this function in new functions if there is a way to avoid it, since it might cause a database lookup.

Definition at line 5718 of file chan_sip.c.

References sip_find_peer_full().

Referenced by _sip_qualify_peer(), _sip_show_peer(), check_peer_ok(), create_addr(), function_sippeer(), handle_request_notify(), manager_sip_peer_status(), receive_message(), register_verify(), sip_devicestate(), sip_do_debug_peer(), sip_msg_send(), sip_report_security_event(), sip_show_user(), sip_unregister(), and transmit_register().

{
   return sip_find_peer_full(peer, addr, NULL, realtime, which_objects, devstate_only, transport);
}

static struct sip_peer* sip_find_peer_by_ip_and_exten	(	struct ast_sockaddr *	addr,
		char *	callbackexten,
		int	transport
	)		[static, read]

Definition at line 5723 of file chan_sip.c.

References FALSE, sip_find_peer_full(), and TRUE.

Referenced by check_peer_ok().

{
   return sip_find_peer_full(NULL, addr, callbackexten, TRUE, FINDPEERS, FALSE, transport);
}

static struct sip_peer* sip_find_peer_full	(	const char *	peer,
		struct ast_sockaddr *	addr,
		char *	callbackexten,
		int	realtime,
		int	which_objects,
		int	devstate_only,
		int	transport
	)		[static, read]

Definition at line 5656 of file chan_sip.c.

References ao2_t_callback_data, ast_copy_string(), ast_set_flag, ast_sockaddr_copy(), find_by_name(), OBJ_POINTER, peer_ipcmp_cb_full(), realtime_peer(), and sip_unref_peer().

Referenced by sip_find_peer(), and sip_find_peer_by_ip_and_exten().

{
   struct sip_peer *p = NULL;
   struct sip_peer tmp_peer;

   if (peer) {
      ast_copy_string(tmp_peer.name, peer, sizeof(tmp_peer.name));
      p = ao2_t_callback_data(peers, OBJ_POINTER, find_by_name, &tmp_peer, &which_objects, "ao2_find in peers table");
   } else if (addr) { /* search by addr? */
      ast_sockaddr_copy(&tmp_peer.addr, addr);
      tmp_peer.flags[0].flags = 0;
      tmp_peer.transports = transport;
      p = ao2_t_callback_data(peers_by_ip, OBJ_POINTER, peer_ipcmp_cb_full, &tmp_peer, callbackexten, "ao2_find in peers_by_ip table");
      if (!p) {
         ast_set_flag(&tmp_peer.flags[0], SIP_INSECURE_PORT);
         p = ao2_t_callback_data(peers_by_ip, OBJ_POINTER, peer_ipcmp_cb_full, &tmp_peer, callbackexten, "ao2_find in peers_by_ip table 2");
         if (p) {
            return p;
         }
      }
   }

   if (!p && (realtime || devstate_only)) {
      /* realtime_peer will return a peer with matching callbackexten if possible, otherwise one matching
       * without the callbackexten */
      p = realtime_peer(peer, addr, callbackexten, devstate_only, which_objects);
      if (p) {
         switch (which_objects) {
         case FINDUSERS:
            if (!(p->type & SIP_TYPE_USER)) {
               sip_unref_peer(p, "Wrong type of realtime SIP endpoint");
               return NULL;
            }
            break;
         case FINDPEERS:
            if (!(p->type & SIP_TYPE_PEER)) {
               sip_unref_peer(p, "Wrong type of realtime SIP endpoint");
               return NULL;
            }
            break;
         case FINDALLDEVICES:
            break;
         }
      }
   }

   return p;
}

static int sip_fixup	(	struct ast_channel *	oldchan,
		struct ast_channel *	newchan
	)		[static]

sip_fixup: Fix up a channel: If a channel is consumed, this is called. Basically update any ->owner links

Definition at line 7400 of file chan_sip.c.

References append_history, ast_channel_flags(), ast_channel_name(), ast_channel_tech_pvt(), ast_debug, AST_FLAG_ZOMBIE, ast_log(), ast_test_flag, LOG_WARNING, sip_pvt_lock, sip_pvt_unlock, and sip_set_rtp_peer().

{
   int ret = -1;
   struct sip_pvt *p;

   if (newchan && ast_test_flag(ast_channel_flags(newchan), AST_FLAG_ZOMBIE))
      ast_debug(1, "New channel is zombie\n");
   if (oldchan && ast_test_flag(ast_channel_flags(oldchan), AST_FLAG_ZOMBIE))
      ast_debug(1, "Old channel is zombie\n");

   if (!newchan || !ast_channel_tech_pvt(newchan)) {
      if (!newchan)
         ast_log(LOG_WARNING, "No new channel! Fixup of %s failed.\n", ast_channel_name(oldchan));
      else
         ast_log(LOG_WARNING, "No SIP tech_pvt! Fixup of %s failed.\n", ast_channel_name(oldchan));
      return -1;
   }
   p = ast_channel_tech_pvt(newchan);

   sip_pvt_lock(p);
   append_history(p, "Masq", "Old channel: %s\n", ast_channel_name(oldchan));
   append_history(p, "Masq (cont)", "...new owner: %s\n", ast_channel_name(newchan));
   if (p->owner != oldchan)
      ast_log(LOG_WARNING, "old channel wasn't %p but was %p\n", oldchan, p->owner);
   else {
      p->owner = newchan;
      /* Re-invite RTP back to Asterisk. Needed if channel is masqueraded out of a native
         RTP bridge (i.e., RTP not going through Asterisk): RTP bridge code might not be
         able to do this if the masquerade happens before the bridge breaks (e.g., AMI
         redirect of both channels). Note that a channel can not be masqueraded *into*
         a native bridge. So there is no danger that this breaks a native bridge that
         should stay up. */
      sip_set_rtp_peer(newchan, NULL, NULL, 0, 0, 0);
      ret = 0;
   }
   ast_debug(3, "SIP Fixup: New owner for dialogue %s: %s (Old parent: %s)\n", p->callid, ast_channel_name(p->owner), ast_channel_name(oldchan));

   sip_pvt_unlock(p);
   return ret;
}

static const char * sip_get_callid

(

struct ast_channel *

chan

)

[static]

Deliver SIP call ID for the call.

Definition at line 5030 of file chan_sip.c.

References ast_channel_tech_pvt().

{
   return ast_channel_tech_pvt(chan) ? ((struct sip_pvt *) ast_channel_tech_pvt(chan))->callid : "";
}

static int sip_get_cc_information	(	struct sip_request *	req,
		char *	subscribe_uri,
		size_t	size,
		enum ast_cc_service_type *	service
	)		[static]

Definition at line 2173 of file chan_sip.c.

References AST_CC_NONE, ast_copy_string(), ast_strlen_zero(), get_in_brackets(), service_string_to_service_type(), and sip_get_header().

Referenced by sip_handle_cc().

{
   char *call_info = ast_strdupa(sip_get_header(req, "Call-Info"));
   char *uri;
   char *purpose;
   char *service_str;
   static const char cc_purpose[] = "purpose=call-completion";
   static const int cc_purpose_len = sizeof(cc_purpose) - 1;

   if (ast_strlen_zero(call_info)) {
      /* No Call-Info present. Definitely no CC offer */
      return -1;
   }

   uri = strsep(&call_info, ";");

   while ((purpose = strsep(&call_info, ";"))) {
      if (!strncmp(purpose, cc_purpose, cc_purpose_len)) {
         break;
      }
   }
   if (!purpose) {
      /* We didn't find the appropriate purpose= parameter. Oh well */
      return -1;
   }

   /* Okay, call-completion has been offered. Let's figure out what type of service this is */
   while ((service_str = strsep(&call_info, ";"))) {
      if (!strncmp(service_str, "m=", 2)) {
         break;
      }
   }
   if (!service_str) {
      /* So they didn't offer a particular service, We'll just go with CCBS since it really
       * doesn't matter anyway
       */
      service_str = "BS";
   } else {
      /* We already determined that there is an "m=" so no need to check
       * the result of this strsep
       */
      strsep(&service_str, "=");
   }

   if ((*service = service_string_to_service_type(service_str)) == AST_CC_NONE) {
      /* Invalid service offered */
      return -1;
   }

   ast_copy_string(subscribe_uri, get_in_brackets(uri), size);

   return 0;
}

static void sip_get_codec	(	struct ast_channel *	chan,
		struct ast_format_cap *	result
	)		[static]

Definition at line 32950 of file chan_sip.c.

References ast_channel_tech_pvt(), ast_format_cap_append(), and ast_format_cap_is_empty().

{
   struct sip_pvt *p = ast_channel_tech_pvt(chan);
   ast_format_cap_append(result, ast_format_cap_is_empty(p->peercaps) ? p->caps : p->peercaps);
}

const char* sip_get_header	(	const struct sip_request *	req,
		const char *	name
	)

Get header from SIP request.

Returns:

Always return something, so don't check for NULL because it won't happen :-)

Definition at line 8296 of file chan_sip.c.

References __get_header().

Referenced by __transmit_response(), build_route(), cc_handle_publish_error(), change_redirecting_information(), check_auth(), check_user_full(), check_via(), copy_header(), extract_uri(), find_call(), find_sdp(), get_also_info(), get_destination(), get_pai(), get_rdnis(), get_realm(), get_refer_info(), get_rpid(), gettag(), handle_cc_notify(), handle_cc_subscribe(), handle_incoming(), handle_request_bye(), handle_request_do(), handle_request_info(), handle_request_invite(), handle_request_invite_st(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_register(), handle_request_subscribe(), handle_request_update(), handle_response(), handle_response_invite(), handle_response_notify(), handle_response_publish(), handle_response_refer(), handle_response_register(), handle_response_subscribe(), handle_response_update(), parse_allowed_methods(), parse_moved_contact(), parse_ok_contact(), parse_oli(), parse_register_contact(), proc_422_rsp(), process_via(), receive_message(), register_verify(), reply_digest(), reqprep(), respprep(), send_request(), send_response(), sip_alloc(), sip_get_cc_information(), sip_pidf_validate(), sip_report_security_event(), sip_sipredirect(), sip_tls_read(), transmit_fake_auth_response(), transmit_refer(), transmit_response_with_auth(), transmit_response_with_sdp(), transmit_response_with_t38_sdp(), and transmit_state_notify().

{
   int start = 0;
   return __get_header(req, name, &start);
}

static enum ast_rtp_glue_result sip_get_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance **	instance
	)		[static]

Definition at line 32728 of file chan_sip.c.

References ao2_ref, ast_channel_tech_pvt(), AST_JB_FORCED, AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_LOCAL, AST_RTP_GLUE_RESULT_REMOTE, ast_test_flag, global_jbconf, sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_LOCAL;

   if (!(p = ast_channel_tech_pvt(chan))) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   sip_pvt_lock(p);
   if (!(p->rtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   ao2_ref(p->rtp, +1);
   *instance = p->rtp;

   if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
      res = AST_RTP_GLUE_RESULT_REMOTE;
   } else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
      res = AST_RTP_GLUE_RESULT_REMOTE;
   } else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
      res = AST_RTP_GLUE_RESULT_FORBID;
   }

   if (p->srtp) {
      res = AST_RTP_GLUE_RESULT_FORBID;
   }

   sip_pvt_unlock(p);

   return res;
}

const char* sip_get_transport

(

enum sip_transport

t

)

Return transport as string.

Definition at line 3795 of file chan_sip.c.

Referenced by _sip_show_peer(), ast_sip_ouraddrfor(), build_contact(), get_transport_pvt(), handle_request_do(), parse_moved_contact(), sip_report_security_event(), sip_show_settings(), sip_show_tcp(), and transmit_notify_with_mwi().

{
   switch (t) {
   case SIP_TRANSPORT_UDP:
      return "UDP";
   case SIP_TRANSPORT_TCP:
      return "TCP";
   case SIP_TRANSPORT_TLS:
      return "TLS";
   case SIP_TRANSPORT_WS:
   case SIP_TRANSPORT_WSS:
      return "WS";
   }

   return "UNKNOWN";
}

static enum ast_rtp_glue_result sip_get_trtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance **	instance
	)		[static]

Definition at line 32790 of file chan_sip.c.

References ao2_ref, ast_channel_tech_pvt(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_REMOTE, ast_test_flag, sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;

   if (!(p = ast_channel_tech_pvt(chan))) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   sip_pvt_lock(p);
   if (!(p->trtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   ao2_ref(p->trtp, +1);
   *instance = p->trtp;

   if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
      res = AST_RTP_GLUE_RESULT_REMOTE;
   }

   sip_pvt_unlock(p);

   return res;
}

static struct ast_udptl * sip_get_udptl_peer

(

struct ast_channel *

chan

)

[static, read]

Definition at line 32599 of file chan_sip.c.

References ast_channel_tech_pvt(), ast_test_flag, sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p;
   struct ast_udptl *udptl = NULL;
   
   p = ast_channel_tech_pvt(chan);
   if (!p) {
      return NULL;
   }

   sip_pvt_lock(p);
   if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
      udptl = p->udptl;
   }
   sip_pvt_unlock(p);
   return udptl;
}

static enum ast_rtp_glue_result sip_get_vrtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance **	instance
	)		[static]

Definition at line 32763 of file chan_sip.c.

References ao2_ref, ast_channel_tech_pvt(), AST_RTP_GLUE_RESULT_FORBID, AST_RTP_GLUE_RESULT_REMOTE, ast_test_flag, sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p = NULL;
   enum ast_rtp_glue_result res = AST_RTP_GLUE_RESULT_FORBID;

   if (!(p = ast_channel_tech_pvt(chan))) {
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   sip_pvt_lock(p);
   if (!(p->vrtp)) {
      sip_pvt_unlock(p);
      return AST_RTP_GLUE_RESULT_FORBID;
   }

   ao2_ref(p->vrtp, +1);
   *instance = p->vrtp;

   if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
      res = AST_RTP_GLUE_RESULT_REMOTE;
   }

   sip_pvt_unlock(p);

   return res;
}

static void sip_handle_cc	(	struct sip_pvt *	pvt,
		struct sip_request *	req,
		enum ast_cc_service_type	service
	)		[static]

Definition at line 2244 of file chan_sip.c.

References ao2_ref, AST_CC_GENERIC_MONITOR_TYPE, ast_cc_get_current_core_id(), AST_CC_MONITOR_ALWAYS, AST_CC_MONITOR_GENERIC, AST_CC_MONITOR_NATIVE, AST_CC_MONITOR_NEVER, ast_channel_get_device_name(), AST_CHANNEL_NAME, ast_get_cc_monitor_policy(), ast_module_ref(), ast_queue_cc_frame(), ast_module_info::self, sip_get_cc_information(), and sip_monitor_instance_init().

Referenced by handle_response(), and handle_response_invite().

{
   enum ast_cc_monitor_policies monitor_policy = ast_get_cc_monitor_policy(pvt->cc_params);
   int core_id;
   char interface_name[AST_CHANNEL_NAME];

   if (monitor_policy == AST_CC_MONITOR_NEVER) {
      /* Don't bother, just return */
      return;
   }

   if ((core_id = ast_cc_get_current_core_id(pvt->owner)) == -1) {
      /* For some reason, CC is invalid, so don't try it! */
      return;
   }

   ast_channel_get_device_name(pvt->owner, interface_name, sizeof(interface_name));

   if (monitor_policy == AST_CC_MONITOR_ALWAYS || monitor_policy == AST_CC_MONITOR_NATIVE) {
      char subscribe_uri[SIPBUFSIZE];
      char device_name[AST_CHANNEL_NAME];
      enum ast_cc_service_type offered_service;
      struct sip_monitor_instance *monitor_instance;
      if (sip_get_cc_information(req, subscribe_uri, sizeof(subscribe_uri), &offered_service)) {
         /* If CC isn't being offered to us, or for some reason the CC offer is
          * not formatted correctly, then it may still be possible to use generic
          * call completion since the monitor policy may be "always"
          */
         goto generic;
      }
      ast_channel_get_device_name(pvt->owner, device_name, sizeof(device_name));
      if (!(monitor_instance = sip_monitor_instance_init(core_id, subscribe_uri, pvt->peername, device_name))) {
         /* Same deal. We can try using generic still */
         goto generic;
      }
      /* We bump the refcount of chan_sip because once we queue this frame, the CC core
       * will have a reference to callbacks in this module. We decrement the module
       * refcount once the monitor destructor is called
       */
      ast_module_ref(ast_module_info->self);
      ast_queue_cc_frame(pvt->owner, "SIP", pvt->dialstring, offered_service, monitor_instance);
      ao2_ref(monitor_instance, -1);
      return;
   }

generic:
   if (monitor_policy == AST_CC_MONITOR_GENERIC || monitor_policy == AST_CC_MONITOR_ALWAYS) {
      ast_queue_cc_frame(pvt->owner, AST_CC_GENERIC_MONITOR_TYPE, interface_name, service, NULL);
   }
}

static int sip_hangup

(

struct ast_channel *

ast

)

[static]

sip_hangup: Hangup SIP call Part of PBX interface, called from ast_hangup

Definition at line 7025 of file chan_sip.c.

References __sip_semi_ack(), append_history, ast_bridged_channel(), ast_cause2str(), AST_CAUSE_ANSWERED_ELSEWHERE, ast_channel_hangupcause(), ast_channel_name(), ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_tech_pvt_set(), ast_channel_trylock, ast_channel_unlock, ast_clear_flag, ast_debug, ast_log(), AST_MAX_USER_FIELD, ast_module_unref(), ast_rtp_instance_get_quality(), ast_rtp_instance_set_stats_vars(), AST_RTP_INSTANCE_STAT_FIELD_QUALITY, ast_sched_add(), AST_SCHED_DEL_UNREF, ast_set_flag, ast_state2str(), AST_STATE_UP, ast_str_buffer(), ast_str_strlen(), ast_test_flag, CHANNEL_DEADLOCK_AVOIDANCE, disable_dsp_detect(), FALSE, find_sip_method(), hangup_cause2sip(), LOG_WARNING, pbx_builtin_setvar_helper(), pvt_set_needdestroy(), quality, reinvite_timeout(), ast_module_info::self, sip_cancel_destroy(), sip_pvt_lock, sip_pvt_trylock, sip_pvt_unlock, sip_scheddestroy(), stop_media_flows(), stop_session_timer(), transmit_request(), transmit_request_with_auth(), transmit_response_reliable(), TRUE, and update_call_counter().

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int needcancel = FALSE;
   int needdestroy = 0;
   struct ast_channel *oldowner = ast;

   if (!p) {
      ast_debug(1, "Asked to hangup channel that was not connected\n");
      return 0;
   }
   if (ast_channel_hangupcause(ast) == AST_CAUSE_ANSWERED_ELSEWHERE) {
      ast_debug(1, "This call was answered elsewhere\n");
      append_history(p, "Cancel", "Call answered elsewhere");
      p->answered_elsewhere = TRUE;
   }

   /* Store hangupcause locally in PVT so we still have it before disconnect */
   if (p->owner)
      p->hangupcause = ast_channel_hangupcause(p->owner);

   if (ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
      if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
         if (sipdebug)
            ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
         update_call_counter(p, DEC_CALL_LIMIT);
      }
      ast_debug(4, "SIP Transfer: Not hanging up right now... Rescheduling hangup for %s.\n", p->callid);
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      ast_clear_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER); /* Really hang up next time */
      if (p->owner) {
         ast_channel_tech_pvt_set(p->owner, dialog_unref(ast_channel_tech_pvt(p->owner), "unref p->owner->tech_pvt"));
         sip_pvt_lock(p);
         p->owner = NULL;  /* Owner will be gone after we return, so take it away */
         sip_pvt_unlock(p);
      }
      ast_module_unref(ast_module_info->self);
      return 0;
   }

   ast_debug(1, "Hangup call %s, SIP callid %s\n", ast_channel_name(ast), p->callid);

   sip_pvt_lock(p);
   if (ast_test_flag(&p->flags[0], SIP_INC_COUNT) || ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD)) {
      if (sipdebug)
         ast_debug(1, "update_call_counter(%s) - decrement call limit counter on hangup\n", p->username);
      update_call_counter(p, DEC_CALL_LIMIT);
   }

   /* Determine how to disconnect */
   if (p->owner != ast) {
      ast_log(LOG_WARNING, "Huh?  We aren't the owner? Can't hangup call.\n");
      sip_pvt_unlock(p);
      return 0;
   }
   /* If the call is not UP, we need to send CANCEL instead of BYE */
   /* In case of re-invites, the call might be UP even though we have an incomplete invite transaction */
   if (p->invitestate < INV_COMPLETED && ast_channel_state(p->owner) != AST_STATE_UP) {
      needcancel = TRUE;
      ast_debug(4, "Hanging up channel in state %s (not UP)\n", ast_state2str(ast_channel_state(ast)));
   }

   stop_media_flows(p); /* Immediately stop RTP, VRTP and UDPTL as applicable */

   append_history(p, needcancel ? "Cancel" : "Hangup", "Cause %s", ast_cause2str(p->hangupcause));

   /* Disconnect */
   disable_dsp_detect(p);

   p->owner = NULL;
   ast_channel_tech_pvt_set(ast, dialog_unref(ast_channel_tech_pvt(ast), "unref ast->tech_pvt"));

   ast_module_unref(ast_module_info->self);
   /* Do not destroy this pvt until we have timeout or
      get an answer to the BYE or INVITE/CANCEL
      If we get no answer during retransmit period, drop the call anyway.
      (Sorry, mother-in-law, you can't deny a hangup by sending
      603 declined to BYE...)
   */
   if (p->alreadygone)
      needdestroy = 1;  /* Set destroy flag at end of this function */
   else if (p->invitestate != INV_CALLING)
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);

   /* Start the process if it's not already started */
   if (!p->alreadygone && p->initreq.data && ast_str_strlen(p->initreq.data)) {
      if (needcancel) { /* Outgoing call, not up */
         if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
            /* if we can't send right now, mark it pending */
            if (p->invitestate == INV_CALLING) {
               /* We can't send anything in CALLING state */
               ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
               /* Do we need a timer here if we don't hear from them at all? Yes we do or else we will get hung dialogs and those are no fun. */
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
               append_history(p, "DELAY", "Not sending cancel, waiting for timeout");
            } else {
               struct sip_pkt *cur;

               for (cur = p->packets; cur; cur = cur->next) {
                  __sip_semi_ack(p, cur->seqno, cur->is_resp, cur->method ? cur->method : find_sip_method(ast_str_buffer(cur->data)));
               }
               p->invitestate = INV_CANCELLED;
               /* Send a new request: CANCEL */
               transmit_request(p, SIP_CANCEL, p->lastinvite, XMIT_RELIABLE, FALSE);
               /* Actually don't destroy us yet, wait for the 487 on our original
                  INVITE, but do set an autodestruct just in case we never get it. */
               needdestroy = 0;
               sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
            }
         } else { /* Incoming call, not up */
            const char *res;
            AST_SCHED_DEL_UNREF(sched, p->provisional_keepalive_sched_id, dialog_unref(p, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));
            if (p->hangupcause && (res = hangup_cause2sip(p->hangupcause)))
               transmit_response_reliable(p, res, &p->initreq);
            else
               transmit_response_reliable(p, "603 Declined", &p->initreq);
            p->invitestate = INV_TERMINATED;
         }
      } else { /* Call is in UP state, send BYE */
         if (p->stimer->st_active == TRUE) {
            stop_session_timer(p);
         }

         if (!p->pendinginvite) {
            struct ast_channel *bridge = ast_bridged_channel(oldowner);
            char quality_buf[AST_MAX_USER_FIELD], *quality;

            /* We need to get the lock on bridge because ast_rtp_instance_set_stats_vars will attempt
             * to lock the bridge. This may get hairy...
             */
            while (bridge && ast_channel_trylock(bridge)) {
               sip_pvt_unlock(p);
               do {
                  CHANNEL_DEADLOCK_AVOIDANCE(oldowner);
               } while (sip_pvt_trylock(p));
               bridge = ast_bridged_channel(oldowner);
            }

            if (p->rtp) {
               ast_rtp_instance_set_stats_vars(oldowner, p->rtp);
            }

            if (bridge) {
               struct sip_pvt *q = ast_channel_tech_pvt(bridge);

               if (IS_SIP_TECH(ast_channel_tech(bridge)) && q && q->rtp) {
                  ast_rtp_instance_set_stats_vars(bridge, q->rtp);
               }
               ast_channel_unlock(bridge);
            }

            /*
             * The channel variables are set below just to get the AMI
             * VarSet event because the channel is being hungup.
             */
            if (p->rtp && (quality = ast_rtp_instance_get_quality(p->rtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
               if (p->do_history) {
                  append_history(p, "RTCPaudio", "Quality:%s", quality);
               }
               pbx_builtin_setvar_helper(oldowner, "RTPAUDIOQOS", quality);
            }
            if (p->vrtp && (quality = ast_rtp_instance_get_quality(p->vrtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
               if (p->do_history) {
                  append_history(p, "RTCPvideo", "Quality:%s", quality);
               }
               pbx_builtin_setvar_helper(oldowner, "RTPVIDEOQOS", quality);
            }
            if (p->trtp && (quality = ast_rtp_instance_get_quality(p->trtp, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
               if (p->do_history) {
                  append_history(p, "RTCPtext", "Quality:%s", quality);
               }
               pbx_builtin_setvar_helper(oldowner, "RTPTEXTQOS", quality);
            }

            /* Send a hangup */
            if (ast_channel_state(oldowner) == AST_STATE_UP) {
               transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
            }

         } else {
            /* Note we will need a BYE when this all settles out
               but we can't send one while we have "INVITE" outstanding. */
            ast_set_flag(&p->flags[0], SIP_PENDINGBYE);  
            ast_clear_flag(&p->flags[0], SIP_NEEDREINVITE); 
            AST_SCHED_DEL_UNREF(sched, p->waitid, dialog_unref(p, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));
            if (sip_cancel_destroy(p)) {
               ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
            }
            /* If we have an ongoing reinvite, there is a chance that we have gotten a provisional
             * response, but something weird has happened and we will never receive a final response.
             * So, just in case, check for pending actions after a bit of time to trigger the pending
             * bye that we are setting above */
            if (p->ongoing_reinvite && p->reinviteid < 0) {
               p->reinviteid = ast_sched_add(sched, 32 * p->timer_t1, reinvite_timeout, dialog_ref(p, "ref for reinvite_timeout"));
            }
         }
      }
   }
   if (needdestroy) {
      pvt_set_needdestroy(p, "hangup");
   }
   sip_pvt_unlock(p);
   return 0;
}

static int sip_indicate	(	struct ast_channel *	ast,
		int	condition,
		const void *	data,
		size_t	datalen
	)		[static]

Play indication to user With SIP a lot of indications is sent as messages, letting the device play the indication - busy signal, congestion etc.

Returns:

-1 to force ast_indicate to send indication in audio, 0 if SIP can handle the indication by sending a message

Definition at line 7705 of file chan_sip.c.

References AST_AOC_D, ast_aoc_decode(), ast_aoc_destroy_decoded(), AST_AOC_E, ast_aoc_get_msg_type(), ast_aoc_get_termination_request(), AST_AOC_REQUEST, AST_AOC_S, ast_channel_name(), ast_channel_tech_pvt(), AST_CONTROL_AOC, AST_CONTROL_BUSY, AST_CONTROL_CONGESTION, AST_CONTROL_CONNECTED_LINE, AST_CONTROL_FLASH, AST_CONTROL_HOLD, AST_CONTROL_INCOMPLETE, AST_CONTROL_PROCEEDING, AST_CONTROL_PROGRESS, AST_CONTROL_PVT_CAUSE_CODE, AST_CONTROL_REDIRECTING, AST_CONTROL_RINGING, AST_CONTROL_SRCCHANGE, AST_CONTROL_SRCUPDATE, AST_CONTROL_T38_PARAMETERS, AST_CONTROL_UNHOLD, AST_CONTROL_UPDATE_RTP_PEER, AST_CONTROL_VIDUPDATE, ast_debug, ast_log(), ast_moh_start(), ast_moh_stop(), ast_rtp_instance_change_source(), ast_rtp_instance_update_source(), ast_set_flag, AST_SOFTHANGUP_DEV, ast_softhangup_nolock(), AST_STATE_RING, AST_STATE_UP, ast_test_flag, initialize_udptl(), interpret_t38_parameters(), LOG_ERROR, LOG_WARNING, sip_alreadygone(), sip_pvt_lock, sip_pvt_unlock, transmit_info_with_aoc(), transmit_info_with_vidupdate(), transmit_provisional_response(), transmit_response(), transmit_response_reliable(), TRUE, update_connectedline(), and update_redirecting().

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int res = 0;

   if (!p) {
      ast_debug(1, "Asked to indicate condition on channel %s with no pvt; ignoring\n",
            ast_channel_name(ast));
      return res;
   }

   sip_pvt_lock(p);
   switch(condition) {
   case AST_CONTROL_RINGING:
      if (ast_channel_state(ast) == AST_STATE_RING) {
         p->invitestate = INV_EARLY_MEDIA;
         if (!ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) ||
             (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER)) {            
            /* Send 180 ringing if out-of-band seems reasonable */
            transmit_provisional_response(p, "180 Ringing", &p->initreq, 0);
            ast_set_flag(&p->flags[0], SIP_RINGING);
            if (ast_test_flag(&p->flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_YES)
               break;
         } else {
            /* Well, if it's not reasonable, just send in-band */
         }
      }
      res = -1;
      break;
   case AST_CONTROL_BUSY:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         transmit_response_reliable(p, "486 Busy Here", &p->initreq);
         p->invitestate = INV_COMPLETED;
         sip_alreadygone(p);
         ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_CONGESTION:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         transmit_response_reliable(p, "503 Service Unavailable", &p->initreq);
         p->invitestate = INV_COMPLETED;
         sip_alreadygone(p);
         ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_INCOMPLETE:
      if (ast_channel_state(ast) != AST_STATE_UP) {
         switch (ast_test_flag(&p->flags[1], SIP_PAGE2_ALLOWOVERLAP)) {
         case SIP_PAGE2_ALLOWOVERLAP_YES:
            transmit_response_reliable(p, "484 Address Incomplete", &p->initreq);
            p->invitestate = INV_COMPLETED;
            sip_alreadygone(p);
            ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
            break;
         case SIP_PAGE2_ALLOWOVERLAP_DTMF:
            /* Just wait for inband DTMF digits */
            break;
         default:
            /* it actually means no support for overlap */
            transmit_response_reliable(p, "404 Not Found", &p->initreq);
            p->invitestate = INV_COMPLETED;
            sip_alreadygone(p);
            ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
            break;
         }
      }
      break;
   case AST_CONTROL_PROCEEDING:
      if ((ast_channel_state(ast) != AST_STATE_UP) &&
          !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
          !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         transmit_response(p, "100 Trying", &p->initreq);
         p->invitestate = INV_PROCEEDING;
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_PROGRESS:
      if ((ast_channel_state(ast) != AST_STATE_UP) &&
          !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
          !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         p->invitestate = INV_EARLY_MEDIA;
         transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
         ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
         break;
      }
      res = -1;
      break;
   case AST_CONTROL_HOLD:
      ast_rtp_instance_update_source(p->rtp);
      ast_moh_start(ast, data, p->mohinterpret);
      break;
   case AST_CONTROL_UNHOLD:
      ast_rtp_instance_update_source(p->rtp);
      ast_moh_stop(ast);
      break;
   case AST_CONTROL_VIDUPDATE:   /* Request a video frame update */
      if (p->vrtp && !p->novideo) {
         transmit_info_with_vidupdate(p);
         /* ast_rtcp_send_h261fur(p->vrtp); */
      } else
         res = -1;
      break;
   case AST_CONTROL_T38_PARAMETERS:
      res = -1;
      if (datalen != sizeof(struct ast_control_t38_parameters)) {
         ast_log(LOG_ERROR, "Invalid datalen for AST_CONTROL_T38_PARAMETERS. Expected %d, got %d\n", (int) sizeof(struct ast_control_t38_parameters), (int) datalen);
      } else {
         const struct ast_control_t38_parameters *parameters = data;
         if (!initialize_udptl(p)) {
            res = interpret_t38_parameters(p, parameters);
         }
      }
      break;
   case AST_CONTROL_SRCUPDATE:
      ast_rtp_instance_update_source(p->rtp);
      break;
   case AST_CONTROL_SRCCHANGE:
      ast_rtp_instance_change_source(p->rtp);
      break;
   case AST_CONTROL_CONNECTED_LINE:
      update_connectedline(p, data, datalen);
      break;
   case AST_CONTROL_REDIRECTING:
      update_redirecting(p, data, datalen);
      break;
   case AST_CONTROL_AOC:
      {
         struct ast_aoc_decoded *decoded = ast_aoc_decode((struct ast_aoc_encoded *) data, datalen, ast);
         if (!decoded) {
            ast_log(LOG_ERROR, "Error decoding indicated AOC data\n");
            res = -1;
            break;
         }
         switch (ast_aoc_get_msg_type(decoded)) {
         case AST_AOC_REQUEST:
            if (ast_aoc_get_termination_request(decoded)) {
               /* TODO, once there is a way to get AOC-E on hangup, attempt that here
                * before hanging up the channel.*/

               /* The other side has already initiated the hangup. This frame
                * just says they are waiting to get AOC-E before completely tearing
                * the call down.  Since SIP does not support this at the moment go
                * ahead and terminate the call here to avoid an unnecessary timeout. */
               ast_debug(1, "AOC-E termination request received on %s. This is not yet supported on sip. Continue with hangup \n", ast_channel_name(p->owner));
               ast_softhangup_nolock(p->owner, AST_SOFTHANGUP_DEV);
            }
            break;
         case AST_AOC_D:
         case AST_AOC_E:
            if (ast_test_flag(&p->flags[2], SIP_PAGE3_SNOM_AOC)) {
               transmit_info_with_aoc(p, decoded);
            }
            break;
         case AST_AOC_S: /* S not supported yet */
         default:
            break;
         }
         ast_aoc_destroy_decoded(decoded);
      }
      break;
   case AST_CONTROL_UPDATE_RTP_PEER: /* Absorb this since it is handled by the bridge */
      break;
   case AST_CONTROL_FLASH: /* We don't currently handle AST_CONTROL_FLASH here, but it is expected, so we don't need to warn either. */
      res = -1;
      break;
   case AST_CONTROL_PVT_CAUSE_CODE: /* these should be handled by the code in channel.c */
   case -1:
      res = -1;
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", condition);
      res = -1;
      break;
   }
   sip_pvt_unlock(p);
   return res;
}

static int sip_is_xml_parsable

(

void

)

[static]

Definition at line 33183 of file chan_sip.c.

References FALSE, and TRUE.

Referenced by load_module().

{
#ifdef HAVE_LIBXML2
   return TRUE;
#else
   return FALSE;
#endif
}

static void sip_keepalive_all_peers

(

void

)

[static]

Send a keepalive to all known peers.

Definition at line 33226 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, AST_SCHED_REPLACE_UNREF, sip_ref_peer(), sip_send_keepalive(), and sip_unref_peer().

Referenced by load_module(), and sip_do_reload().

{
   struct ao2_iterator i;
   struct sip_peer *peer;

   if (!speerobjs) {       /* No peers, just give up */
      return;
   }

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      ao2_lock(peer);
      AST_SCHED_REPLACE_UNREF(peer->keepalivesend, sched, 0, sip_send_keepalive, peer,
               sip_unref_peer(_data, "removing poke peer ref"),
               sip_unref_peer(peer, "removing poke peer ref"),
               sip_ref_peer(peer, "adding poke peer ref"));
      ao2_unlock(peer);
      sip_unref_peer(peer, "toss iterator peer ptr");
   }
   ao2_iterator_destroy(&i);
}

static int sip_monitor_instance_cmp_fn	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 1950 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_monitor_instance *monitor_instance1 = obj;
   struct sip_monitor_instance *monitor_instance2 = arg;

   return monitor_instance1->core_id == monitor_instance2->core_id ? CMP_MATCH | CMP_STOP : 0;
}

static void sip_monitor_instance_destructor

(

void *

data

)

[static]

Definition at line 1958 of file chan_sip.c.

References ao2_t_ref, ast_string_field_free_memory, FALSE, sip_pvt_lock, sip_pvt_unlock, transmit_invite(), and transmit_publish().

Referenced by sip_monitor_instance_init().

{
   struct sip_monitor_instance *monitor_instance = data;
   if (monitor_instance->subscription_pvt) {
      sip_pvt_lock(monitor_instance->subscription_pvt);
      monitor_instance->subscription_pvt->expiry = 0;
      transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 0, monitor_instance->subscribe_uri);
      sip_pvt_unlock(monitor_instance->subscription_pvt);
      dialog_unref(monitor_instance->subscription_pvt, "Unref monitor instance ref of subscription pvt");
   }
   if (monitor_instance->suspension_entry) {
      monitor_instance->suspension_entry->body[0] = '\0';
      transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_REMOVE ,monitor_instance->notify_uri);
      ao2_t_ref(monitor_instance->suspension_entry, -1, "Decrementing suspension entry refcount in sip_monitor_instance_destructor");
   }
   ast_string_field_free_memory(monitor_instance);
}

static int sip_monitor_instance_hash_fn	(	const void *	obj,
		const int	flags
	)		[static]

Definition at line 1944 of file chan_sip.c.

Referenced by load_module().

{
   const struct sip_monitor_instance *monitor_instance = obj;
   return monitor_instance->core_id;
}

static struct sip_monitor_instance* sip_monitor_instance_init	(	int	core_id,
		const char *const	subscribe_uri,
		const char *const	peername,
		const char *const	device_name
	)		[static, read]

Definition at line 1976 of file chan_sip.c.

References ao2_alloc, ao2_link, ao2_ref, ast_string_field_init, ast_string_field_set, and sip_monitor_instance_destructor().

Referenced by sip_handle_cc().

{
   struct sip_monitor_instance *monitor_instance = ao2_alloc(sizeof(*monitor_instance), sip_monitor_instance_destructor);

   if (!monitor_instance) {
      return NULL;
   }

   if (ast_string_field_init(monitor_instance, 256)) {
      ao2_ref(monitor_instance, -1);
      return NULL;
   }

   ast_string_field_set(monitor_instance, subscribe_uri, subscribe_uri);
   ast_string_field_set(monitor_instance, peername, peername);
   ast_string_field_set(monitor_instance, device_name, device_name);
   monitor_instance->core_id = core_id;
   ao2_link(sip_monitor_instances, monitor_instance);
   return monitor_instance;
}

static int sip_msg_send	(	const struct ast_msg *	msg,
		const char *	to,
		const char *	from
	)		[static]

Definition at line 26994 of file chan_sip.c.

References add_msg_header(), ast_callerid_parse(), ast_log(), ast_msg_get_body(), ast_msg_var_iterator_destroy(), ast_msg_var_iterator_init(), ast_msg_var_iterator_next(), ast_msg_var_unref_current(), ast_set_flag, ast_sip_ouraddrfor(), ast_string_field_set, ast_strlen_zero(), block_msg_header(), build_via(), create_addr(), dialog_unlink_all(), extract_host_from_hostport(), get_in_brackets(), LOG_NOTICE, LOG_WARNING, name, parse_uri(), S_OR, sip_alloc(), sip_find_peer(), SIP_PEDANTIC_DECODE, sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), sip_unref_peer(), transmit_message(), TRUE, and var.

{
   struct sip_pvt *pvt;
   int res;
   char *to_uri;
   char *to_host;
   char *to_user;
   const char *var;
   const char *val;
   struct ast_msg_var_iterator *iter;
   struct sip_peer *peer_ptr;

   if (!(pvt = sip_alloc(NULL, NULL, 0, SIP_MESSAGE, NULL, NULL))) {
      return -1;
   }

        for (iter = ast_msg_var_iterator_init(msg);
        ast_msg_var_iterator_next(msg, iter, &var, &val);
        ast_msg_var_unref_current(iter)) {
      if (!strcasecmp(var, "Request-URI")) {
         ast_string_field_set(pvt, fullcontact, val);
         ast_msg_var_unref_current(iter);
         break;
      }
   }
   ast_msg_var_iterator_destroy(iter);

   to_uri = ast_strdupa(to);
   to_uri = get_in_brackets(to_uri);
   parse_uri(to_uri, "sip:,sips:", &to_user, NULL, &to_host, NULL);

   if (ast_strlen_zero(to_host)) {
      ast_log(LOG_WARNING, "MESSAGE(to) is invalid for SIP - '%s'\n", to);
      dialog_unlink_all(pvt);
      dialog_unref(pvt, "MESSAGE(to) is invalid for SIP");
      return -1;
   }

   if (!ast_strlen_zero(from)) {
      if ((peer_ptr = sip_find_peer(from, NULL, 0, 1, 0, 0))) {
         ast_string_field_set(pvt, fromname, S_OR(peer_ptr->cid_name, peer_ptr->name));
         ast_string_field_set(pvt, fromuser, S_OR(peer_ptr->cid_num, peer_ptr->name));
         sip_unref_peer(peer_ptr, "sip_unref_peer, from sip_msg_send, sip_find_peer");
      } else if (strchr(from, '<')) { /* from is callerid-style */
         char *sender;
         char *name = NULL, *location = NULL, *user = NULL, *domain = NULL;

         sender = ast_strdupa(from);
         ast_callerid_parse(sender, &name, &location);
         if (ast_strlen_zero(location)) {
            /* This can occur if either
             *  1) A name-addr style From header does not close the angle brackets
             *  properly.
             *  2) The From header is not in name-addr style and the content of the
             *  From contains characters other than 0-9, *, #, or +.
             *
             *  In both cases, ast_callerid_parse() should have parsed the From header
             *  as a name rather than a number. So we just need to set the location
             *  to what was parsed as a name, and set the name NULL since there was
             *  no name present.
             */
            location = name;
            name = NULL;
         }
         ast_string_field_set(pvt, fromname, name);
         if (strchr(location, ':')) { /* Must be a URI */
            parse_uri(location, "sip:,sips:", &user, NULL, &domain, NULL);
            SIP_PEDANTIC_DECODE(user);
            SIP_PEDANTIC_DECODE(domain);
            extract_host_from_hostport(&domain);
            ast_string_field_set(pvt, fromuser, user);
            ast_string_field_set(pvt, fromdomain, domain);
         } else { /* Treat it as an exten/user */
            ast_string_field_set(pvt, fromuser, location);
         }
      } else { /* assume we just have the name, use defaults for the rest */
         ast_string_field_set(pvt, fromname, from);
      }
   }

   sip_pvt_lock(pvt);

   /* Look up the host to contact */
   if (create_addr(pvt, to_host, NULL, TRUE)) {
      sip_pvt_unlock(pvt);
      dialog_unlink_all(pvt);
      dialog_unref(pvt, "create_addr failed sending a MESSAGE");
      return -1;
   }

   if (!ast_strlen_zero(to_user)) {
      ast_string_field_set(pvt, username, to_user);
   }
   ast_sip_ouraddrfor(&pvt->sa, &pvt->ourip, pvt);
   build_via(pvt);
   ast_set_flag(&pvt->flags[0], SIP_OUTGOING);

   /* XXX Does pvt->expiry need to be set? */

   /* Save additional MESSAGE headers in case of authentication request. */
   for (iter = ast_msg_var_iterator_init(msg);
      ast_msg_var_iterator_next(msg, iter, &var, &val);
      ast_msg_var_unref_current(iter)) {
      if (!strcasecmp(var, "Max-Forwards")) {
         /* Decrement Max-Forwards for SIP loop prevention. */
         if (sscanf(val, "%30d", &pvt->maxforwards) != 1 || pvt->maxforwards < 1) {
            sip_pvt_unlock(pvt);
            dialog_unlink_all(pvt);
            dialog_unref(pvt, "MESSAGE(Max-Forwards) reached zero.");
            ast_log(LOG_NOTICE,
               "MESSAGE(Max-Forwards) reached zero.  MESSAGE not sent.\n");
            return -1;
         }
         --pvt->maxforwards;
         continue;
      }
      if (block_msg_header(var)) {
         /* Block addition of this header. */
         continue;
      }
      add_msg_header(pvt, var, val);
   }
   ast_msg_var_iterator_destroy(iter);

   ast_string_field_set(pvt, msg_body, ast_msg_get_body(msg));
   res = transmit_message(pvt, 1, 0);

   sip_pvt_unlock(pvt);
   sip_scheddestroy(pvt, DEFAULT_TRANS_TIMEOUT);
   dialog_unref(pvt, "sent a MESSAGE");

   return res;
}

static const char * sip_nat_mode

(

const struct sip_pvt *

p

)

[static]

Display SIP nat mode.

Definition at line 3713 of file chan_sip.c.

References ast_test_flag.

Referenced by check_via(), retrans_pkt(), and send_response().

{
   return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) ? "NAT" : "no NAT";
}

static struct ast_channel* sip_new	(	struct sip_pvt *	i,
		int	state,
		const char *	title,
		const char *	linkedid,
		struct ast_callid *	callid
	)		[static, read]

Initiate a call in the SIP channel.

Note:

called from sip_request_call (calls from the pbx ) for outbound channels and from handle_request_invite for inbound channels

Precondition:

i is locked

Returns:

New ast_channel locked.

Definition at line 7899 of file chan_sip.c.

References ast_channel::accountcode, ast_channel::amaflags, ast_party_caller::ani, append_history, AST_ADSI_UNAVAILABLE, ast_atomic_fetchadd_int(), ast_channel_adsicpe_set(), ast_channel_alloc(), ast_channel_amaflags_set(), ast_channel_caller(), ast_channel_callgroup_set(), ast_channel_callid_set(), ast_channel_cc_params_init(), ast_channel_context_set(), ast_channel_dialed(), ast_channel_exten_set(), ast_channel_flags(), ast_channel_lock, ast_channel_name(), ast_channel_named_callgroups_set(), ast_channel_named_pickupgroups_set(), ast_channel_nativeformats(), ast_channel_pickupgroup_set(), ast_channel_priority_set(), ast_channel_rawreadformat(), ast_channel_rawwriteformat(), ast_channel_readformat(), ast_channel_redirecting(), ast_channel_rings_set(), ast_channel_set_fd(), ast_channel_tech_pvt_set(), ast_channel_tech_set(), ast_channel_uniqueid(), ast_channel_unlock, ast_channel_writeformat(), ast_channel_zone_set(), ast_codec_choose(), ast_debug, ast_exists_extension(), AST_FLAG_DISABLE_DEVSTATE_CACHE, ast_format_cap_add(), ast_format_cap_copy(), ast_format_cap_has_type(), ast_format_cap_is_empty(), ast_format_cap_remove_bytype(), ast_format_copy(), AST_FORMAT_TYPE_AUDIO, AST_FORMAT_TYPE_TEXT, AST_FORMAT_TYPE_VIDEO, ast_get_encoded_str(), ast_get_indication_zone(), ast_getformatname(), ast_getformatname_multiple(), ast_jb_configure(), ast_log(), ast_module_ref(), AST_RTP_DTMF_MODE_INBAND, AST_RTP_DTMF_MODE_RFC2833, ast_rtp_instance_dtmf_mode_set(), ast_rtp_instance_fd(), ast_rtp_instance_set_read_format(), ast_rtp_instance_set_write_format(), ast_set_flag, AST_STATE_RING, ast_strdup, ast_strlen_zero(), ast_test_flag, ast_udptl_fd(), ast_uri_decode(), ast_uri_sip_user, ast_channel::callgroup, ast_channel::callid, ast_channel::context, enable_dsp_detect(), EVENT_FLAG_SYSTEM, exten, ast_party_redirecting::from, global_jbconf, ast_party_caller::id, ast_channel::language, LOG_ERROR, LOG_WARNING, manager_event, ast_variable::name, ast_party_id::name, ast_channel::named_callgroups, ast_channel::named_pickupgroups, ast_variable::next, ast_party_id::number, ast_party_dialed::number, ast_channel::parkinglot, pbx_builtin_setvar_helper(), ast_channel::pickupgroup, ast_party_name::presentation, ast_party_number::presentation, ast_module_info::self, sip_cfg, sip_pvt_lock, sip_pvt_unlock, sip_tech_info, ast_party_number::str, ast_party_dialed::str, ast_party_id::tag, ast_party_number::valid, and ast_variable::value.

Referenced by handle_request_invite(), and sip_request_call().

{
   struct ast_channel *tmp;
   struct ast_variable *v = NULL;
   struct ast_format fmt;
   struct ast_format_cap *what = NULL; /* SHALLOW COPY DO NOT DESTROY! */
   int needvideo = 0;
   int needtext = 0;
   char buf[SIPBUFSIZE];
   char *exten;

   {
      const char *my_name; /* pick a good name */
   
      if (title) {
         my_name = title;
      } else {
         my_name = ast_strdupa(i->fromdomain);
      }

      sip_pvt_unlock(i);
      /* Don't hold a sip pvt lock while we allocate a channel */
      tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, linkedid, i->amaflags, "SIP/%s-%08x", my_name, ast_atomic_fetchadd_int((int *)&chan_idx, +1));
   }
   if (!tmp) {
      ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
      sip_pvt_lock(i);
      return NULL;
   }

   /* If we sent in a callid, bind it to the channel. */
   if (callid) {
      ast_channel_callid_set(tmp, callid);
   }

   ast_channel_lock(tmp);
   sip_pvt_lock(i);
   ast_channel_cc_params_init(tmp, i->cc_params);
   ast_channel_caller(tmp)->id.tag = ast_strdup(i->cid_tag);

   ast_channel_tech_set(tmp, (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INFO || ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO) ? &sip_tech_info : &sip_tech);

   /* Select our native format based on codec preference until we receive
      something from another device to the contrary. */
   if (!(ast_format_cap_is_empty(i->jointcaps))) { /* The joint capabilities of us and peer */
      what = i->jointcaps;
   } else if (!(ast_format_cap_is_empty(i->caps))) {     /* Our configured capability for this peer */
      what = i->caps;
   } else {
      what = sip_cfg.caps;
   }

   /* Set the native formats */
   ast_format_cap_copy(ast_channel_nativeformats(tmp), what);
   /* choose and use only the best audio format for our native formats */
   ast_codec_choose(&i->prefs, ast_channel_nativeformats(tmp), 1, &fmt); /* get the best audio format */
   ast_format_cap_remove_bytype(ast_channel_nativeformats(tmp), AST_FORMAT_TYPE_AUDIO); /* remove only the other audio formats */
   ast_format_cap_add(ast_channel_nativeformats(tmp), &fmt); /* add our best choice back */

   ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_channel_nativeformats(tmp)));
   ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcaps));
   ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->caps));
   ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname(&fmt));
   if (!ast_format_cap_is_empty(i->prefcaps)) {
      ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcaps));
   }

   /* If we have a prefcodec setting, we have an inbound channel that set a
      preferred format for this call. Otherwise, we check the jointcapability
      We also check for vrtp. If it's not there, we are not allowed do any video anyway.
    */
   if (i->vrtp) {
      if (ast_test_flag(&i->flags[1], SIP_PAGE2_VIDEOSUPPORT))
         needvideo = 1;
      else if (!ast_format_cap_is_empty(i->prefcaps))
         needvideo = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_VIDEO);   /* Outbound call */
      else
         needvideo = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_VIDEO);  /* Inbound call */
   }

   if (i->trtp) {
      if (!ast_format_cap_is_empty(i->prefcaps))
         needtext = ast_format_cap_has_type(i->prefcaps, AST_FORMAT_TYPE_TEXT);  /* Outbound call */
      else
         needtext = ast_format_cap_has_type(i->jointcaps, AST_FORMAT_TYPE_TEXT); /* Inbound call */
   }

   if (needvideo) {
      ast_debug(3, "This channel can handle video! HOLLYWOOD next!\n");
   } else {
      ast_debug(3, "This channel will not be able to handle video.\n");
   }

   enable_dsp_detect(i);

   if ((ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
       (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
      if (i->rtp) {
         ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_INBAND);
      }
   } else if (ast_test_flag(&i->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) {
      if (i->rtp) {
         ast_rtp_instance_dtmf_mode_set(i->rtp, AST_RTP_DTMF_MODE_RFC2833);
      }
   }

   /* Set file descriptors for audio, video, and realtime text.  Since
    * UDPTL is created as needed in the lifetime of a dialog, its file
    * descriptor is set in initialize_udptl */
   if (i->rtp) {
      ast_channel_set_fd(tmp, 0, ast_rtp_instance_fd(i->rtp, 0));
      ast_channel_set_fd(tmp, 1, ast_rtp_instance_fd(i->rtp, 1));
      ast_rtp_instance_set_write_format(i->rtp, &fmt);
      ast_rtp_instance_set_read_format(i->rtp, &fmt);
   }
   if (needvideo && i->vrtp) {
      ast_channel_set_fd(tmp, 2, ast_rtp_instance_fd(i->vrtp, 0));
      ast_channel_set_fd(tmp, 3, ast_rtp_instance_fd(i->vrtp, 1));
   }
   if (needtext && i->trtp) {
      ast_channel_set_fd(tmp, 4, ast_rtp_instance_fd(i->trtp, 0));
   }
   if (i->udptl) {
      ast_channel_set_fd(tmp, 5, ast_udptl_fd(i->udptl));
   }

   if (state == AST_STATE_RING) {
      ast_channel_rings_set(tmp, 1);
   }
   ast_channel_adsicpe_set(tmp, AST_ADSI_UNAVAILABLE);

   ast_format_copy(ast_channel_writeformat(tmp), &fmt);
   ast_format_copy(ast_channel_rawwriteformat(tmp), &fmt);

   ast_format_copy(ast_channel_readformat(tmp), &fmt);
   ast_format_copy(ast_channel_rawreadformat(tmp), &fmt);

   ast_channel_tech_pvt_set(tmp, dialog_ref(i, "sip_new: set chan->tech_pvt to i"));

   ast_channel_callgroup_set(tmp, i->callgroup);
   ast_channel_pickupgroup_set(tmp, i->pickupgroup);

   ast_channel_named_callgroups_set(tmp, i->named_callgroups);
   ast_channel_named_pickupgroups_set(tmp, i->named_pickupgroups);

   ast_channel_caller(tmp)->id.name.presentation = i->callingpres;
   ast_channel_caller(tmp)->id.number.presentation = i->callingpres;
   if (!ast_strlen_zero(i->parkinglot)) {
      ast_channel_parkinglot_set(tmp, i->parkinglot);
   }
   if (!ast_strlen_zero(i->accountcode)) {
      ast_channel_accountcode_set(tmp, i->accountcode);
   }
   if (i->amaflags) {
      ast_channel_amaflags_set(tmp, i->amaflags);
   }
   if (!ast_strlen_zero(i->language)) {
      ast_channel_language_set(tmp, i->language);
   }
   if (!ast_strlen_zero(i->zone)) {
      struct ast_tone_zone *zone;
      if (!(zone = ast_get_indication_zone(i->zone))) {
         ast_log(LOG_ERROR, "Unknown country code '%s' for tonezone. Check indications.conf for available country codes.\n", i->zone);
      } 
      ast_channel_zone_set(tmp, zone);
   }
   i->owner = tmp;
   ast_module_ref(ast_module_info->self);
   ast_channel_context_set(tmp, i->context);
   /*Since it is valid to have extensions in the dialplan that have unescaped characters in them
    * we should decode the uri before storing it in the channel, but leave it encoded in the sip_pvt
    * structure so that there aren't issues when forming URI's
    */
   exten = ast_strdupa(i->exten);
   sip_pvt_unlock(i);
   ast_channel_unlock(tmp);
   if (!ast_exists_extension(NULL, i->context, i->exten, 1, i->cid_num)) {
      ast_uri_decode(exten, ast_uri_sip_user);
   }
   ast_channel_lock(tmp);
   sip_pvt_lock(i);
   ast_channel_exten_set(tmp, exten);

   /* Don't use ast_set_callerid() here because it will
    * generate an unnecessary NewCallerID event  */
   if (!ast_strlen_zero(i->cid_num)) {
      ast_channel_caller(tmp)->ani.number.valid = 1;
      ast_channel_caller(tmp)->ani.number.str = ast_strdup(i->cid_num);
   }
   if (!ast_strlen_zero(i->rdnis)) {
      ast_channel_redirecting(tmp)->from.number.valid = 1;
      ast_channel_redirecting(tmp)->from.number.str = ast_strdup(i->rdnis);
   }

   if (!ast_strlen_zero(i->exten) && strcmp(i->exten, "s")) {
      ast_channel_dialed(tmp)->number.str = ast_strdup(i->exten);
   }

   ast_channel_priority_set(tmp, 1);
   if (!ast_strlen_zero(i->uri)) {
      pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri);
   }
   if (!ast_strlen_zero(i->domain)) {
      pbx_builtin_setvar_helper(tmp, "SIPDOMAIN", i->domain);
   }
   if (!ast_strlen_zero(i->callid)) {
      pbx_builtin_setvar_helper(tmp, "SIPCALLID", i->callid);
   }
   if (i->rtp) {
      ast_jb_configure(tmp, &global_jbconf);
   }

   if (!i->relatedpeer) {
      ast_set_flag(ast_channel_flags(tmp), AST_FLAG_DISABLE_DEVSTATE_CACHE);
   }
   /* Set channel variables for this call from configuration */
   for (v = i->chanvars ; v ; v = v->next) {
      char valuebuf[1024];
      pbx_builtin_setvar_helper(tmp, v->name, ast_get_encoded_str(v->value, valuebuf, sizeof(valuebuf)));
   }

   if (i->do_history) {
      append_history(i, "NewChan", "Channel %s - from %s", ast_channel_name(tmp), i->callid);
   }

   /* Inform manager user about new channel and their SIP call ID */
   if (sip_cfg.callevents) {
      manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
         "Channel: %s\r\nUniqueid: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\n",
         ast_channel_name(tmp), ast_channel_uniqueid(tmp), "SIP", i->callid, i->fullcontact);
   }

   return tmp;
}

static int sip_notify_alloc

(

struct sip_pvt *

p

)

[static]

Allocate SIP refer structure.

Definition at line 15493 of file chan_sip.c.

References ast_calloc, and ast_str_create().

Referenced by manager_sipnotify(), and sip_cli_notify().

{
   p->notify = ast_calloc(1, sizeof(struct sip_notify));
   if (p->notify) {
      p->notify->content = ast_str_create(128);
   }
   return p->notify ? 1 : 0;
}

static int sip_offer_timer_expire

(

const void *

data

)

[static]

Definition at line 1826 of file chan_sip.c.

References ast_cc_failed(), ast_cc_agent::core_id, ast_cc_agent::device_name, and ast_cc_agent::private_data.

Referenced by sip_cc_agent_start_offer_timer().

{
   struct ast_cc_agent *agent = (struct ast_cc_agent *) data;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

   agent_pvt->offer_timer_id = -1;

   return ast_cc_failed(agent->core_id, "SIP agent %s's offer timer expired", agent->device_name);
}

static int sip_park	(	struct ast_channel *	chan1,
		struct ast_channel *	chan2,
		struct sip_request *	req,
		uint32_t	seqno,
		const char *	park_exten,
		const char *	park_context
	)		[static]

DO NOT hold any locks while calling sip_park

Definition at line 24321 of file chan_sip.c.

References ast_calloc, ast_channel_accountcode(), ast_channel_alloc(), ast_channel_amaflags(), ast_channel_context(), ast_channel_context_set(), ast_channel_exten(), ast_channel_exten_set(), ast_channel_linkedid(), ast_channel_masquerade(), ast_channel_name(), ast_channel_parkinglot(), ast_channel_priority(), ast_channel_priority_set(), ast_channel_readformat(), ast_channel_writeformat(), ast_do_masquerade(), ast_format_copy(), ast_free, ast_hangup(), ast_pthread_create_detached_background, AST_STATE_DOWN, ast_strdup, copy_request(), deinit_req(), and sip_park_thread().

Referenced by handle_request_refer().

{
   struct sip_dual *d;
   struct ast_channel *transferee, *transferer;
   pthread_t th;

   transferee = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, ast_channel_accountcode(chan1), ast_channel_exten(chan1), ast_channel_context(chan1), ast_channel_linkedid(chan1), ast_channel_amaflags(chan1), "Parking/%s", ast_channel_name(chan1));
   transferer = ast_channel_alloc(0, AST_STATE_DOWN, 0, 0, ast_channel_accountcode(chan2), ast_channel_exten(chan2), ast_channel_context(chan2), ast_channel_linkedid(chan2), ast_channel_amaflags(chan2), "SIPPeer/%s", ast_channel_name(chan2));
   d = ast_calloc(1, sizeof(*d));
   if (!transferee || !transferer || !d) {
      if (transferee) {
         ast_hangup(transferee);
      }
      if (transferer) {
         ast_hangup(transferer);
      }
      ast_free(d);
      return -1;
   }
   d->park_exten = ast_strdup(park_exten);
   d->park_context = ast_strdup(park_context);
   if (!d->park_exten || !d->park_context) {
      ast_hangup(transferee);
      ast_hangup(transferer);
      ast_free(d->park_exten);
      ast_free(d->park_context);
      ast_free(d);
      return -1;
   }

   /* Make formats okay */
   ast_format_copy(ast_channel_readformat(transferee), ast_channel_readformat(chan1));
   ast_format_copy(ast_channel_writeformat(transferee), ast_channel_writeformat(chan1));

   /* Prepare for taking over the channel */
   if (ast_channel_masquerade(transferee, chan1)) {
      ast_hangup(transferee);
      ast_hangup(transferer);
      ast_free(d->park_exten);
      ast_free(d->park_context);
      ast_free(d);
      return -1;
   }

   /* Setup the extensions and such */
   ast_channel_context_set(transferee, ast_channel_context(chan1));
   ast_channel_exten_set(transferee, ast_channel_exten(chan1));
   ast_channel_priority_set(transferee, ast_channel_priority(chan1));

   ast_do_masquerade(transferee);

   /* We make a clone of the peer channel too, so we can play
      back the announcement */

   /* Make formats okay */
   ast_format_copy(ast_channel_readformat(transferer), ast_channel_readformat(chan2));
   ast_format_copy(ast_channel_writeformat(transferer), ast_channel_writeformat(chan2));
   ast_channel_parkinglot_set(transferer, ast_channel_parkinglot(chan2));

   /* Prepare for taking over the channel */
   if (ast_channel_masquerade(transferer, chan2)) {
      ast_hangup(transferee);
      ast_hangup(transferer);
      ast_free(d->park_exten);
      ast_free(d->park_context);
      ast_free(d);
      return -1;
   }

   /* Setup the extensions and such */
   ast_channel_context_set(transferer, ast_channel_context(chan2));
   ast_channel_exten_set(transferer, ast_channel_exten(chan2));
   ast_channel_priority_set(transferer, ast_channel_priority(chan2));

   ast_do_masquerade(transferer);

   /* Save original request for followup */
   copy_request(&d->req, req);
   d->chan1 = transferee;  /* Transferee */
   d->chan2 = transferer;  /* Transferer */
   d->seqno = seqno;
   if (ast_pthread_create_detached_background(&th, NULL, sip_park_thread, d) < 0) {
      /* Could not start thread */
      ast_hangup(transferer);
      ast_hangup(transferee);
      deinit_req(&d->req);
      ast_free(d->park_exten);
      ast_free(d->park_context);
      ast_free(d);   /* We don't need it anymore. If thread is created, d will be free'd
               by sip_park_thread() */
      return -1;
   }
   return 0;
}

static void * sip_park_thread

(

void *

stuff

)

[static]

Park SIP call support function Starts in a new thread, then parks the call XXX Should we add a wait period after streaming audio and before hangup?? Sometimes the audio can't be heard before hangup.

Definition at line 24262 of file chan_sip.c.

References append_history, AST_CAUSE_NORMAL_CLEARING, ast_channel_hangupcause_set(), ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_free, ast_hangup(), ast_log(), AST_LOG_NOTICE, ast_park_call_exten(), ast_set_flag, ast_string_field_set, deinit_req(), destroy_msg_headers(), ext, sip_pvt_lock, sip_pvt_unlock, transmit_message(), transmit_notify_with_sipfrag(), and TRUE.

Referenced by sip_park().

{
   struct ast_channel *transferee, *transferer; /* Chan1: The transferee, Chan2: The transferer */
   struct sip_pvt *transferer_pvt;
   struct sip_dual *d;
   int ext;
   int res;

   d = stuff;
   transferee = d->chan1;
   transferer = d->chan2;
   transferer_pvt = ast_channel_tech_pvt(transferer);

   ast_debug(4, "SIP Park: Transferer channel %s, Transferee %s\n", ast_channel_name(transferer), ast_channel_name(transferee));

   res = ast_park_call_exten(transferee, transferer, d->park_exten, d->park_context, 0, &ext);

   sip_pvt_lock(transferer_pvt);
#ifdef WHEN_WE_KNOW_THAT_THE_CLIENT_SUPPORTS_MESSAGE
   if (res) {
      destroy_msg_headers(transferer_pvt);
      ast_string_field_set(transferer_pvt, msg_body, "Unable to park call.");
      transmit_message(transferer_pvt, 0, 0);
   } else {
      /* Then tell the transferer what happened */
      destroy_msg_headers(transferer_pvt);
      sprintf(buf, "Call parked on extension '%d'.", ext);
      ast_string_field_set(transferer_pvt, msg_body, buf);
      transmit_message(transferer_pvt, 0, 0);
   }
#endif

   /* Any way back to the current call??? */
   /* Transmit response to the REFER request */
   ast_set_flag(&transferer_pvt->flags[0], SIP_DEFER_BYE_ON_TRANSFER);
   if (!res)   {
      /* Transfer succeeded */
      append_history(transferer_pvt, "SIPpark", "Parked call on %d", ext);
      transmit_notify_with_sipfrag(transferer_pvt, d->seqno, "200 OK", TRUE);
      sip_pvt_unlock(transferer_pvt);
      ast_channel_hangupcause_set(transferer, AST_CAUSE_NORMAL_CLEARING);
      ast_debug(1, "SIP Call parked on extension '%d'\n", ext);
   } else {
      transmit_notify_with_sipfrag(transferer_pvt, d->seqno, "503 Service Unavailable", TRUE);
      append_history(transferer_pvt, "SIPpark", "Parking failed\n");
      sip_pvt_unlock(transferer_pvt);
      ast_log(AST_LOG_NOTICE, "SIP Call parked failed for %s\n", ast_channel_name(transferee));
      ast_hangup(transferee);
   }
   ast_hangup(transferer);

   deinit_req(&d->req);
   ast_free(d->park_exten);
   ast_free(d->park_context);
   ast_free(d);
   return NULL;
}

static void sip_peer_hold	(	struct sip_pvt *	p,
		int	hold
	)		[static]

Change onhold state of a peer using a pvt structure.

Definition at line 16571 of file chan_sip.c.

References ast_atomic_fetchadd_int(), ast_channel_flags(), AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), AST_DEVSTATE_NOT_CACHABLE, AST_FLAG_DISABLE_DEVSTATE_CACHE, and ast_test_flag.

Referenced by change_hold_state(), and update_call_counter().

{
   if (!p->relatedpeer) {
      return;
   }

   /* If they put someone on hold, increment the value... otherwise decrement it */
   ast_atomic_fetchadd_int(&p->relatedpeer->onhold, (hold ? +1 : -1));

   /* Request device state update */
   ast_devstate_changed(AST_DEVICE_UNKNOWN, (ast_test_flag(ast_channel_flags(p->owner), AST_FLAG_DISABLE_DEVSTATE_CACHE) ? AST_DEVSTATE_NOT_CACHABLE : AST_DEVSTATE_CACHABLE),
              "SIP/%s", p->relatedpeer->name);

   return;
}

static int sip_pickup

(

struct ast_channel *

chan

)

[static]

Pickup a call using the subsystem in features.c This is executed in a separate thread.

Definition at line 24439 of file chan_sip.c.

References ast_channel_name(), ast_channel_ref, ast_channel_unref, ast_debug, ast_pthread_create_detached_background, and sip_pickup_thread().

Referenced by handle_request_invite().

{
   pthread_t threadid;

   ast_channel_ref(chan);

   if (ast_pthread_create_detached_background(&threadid, NULL, sip_pickup_thread, chan)) {
      ast_debug(1, "Unable to start Group pickup thread on channel %s\n", ast_channel_name(chan));
      ast_channel_unref(chan);
      return -1;
   }
   ast_debug(1, "Started Group pickup thread on channel %s\n", ast_channel_name(chan));
   return 0;
}

static void * sip_pickup_thread

(

void *

stuff

)

[static]

SIP pickup support function Starts in a new thread, then pickup the call.

Definition at line 24420 of file chan_sip.c.

References AST_CAUSE_CALL_REJECTED, AST_CAUSE_NORMAL_CLEARING, ast_channel_hangupcause_set(), ast_channel_unref, ast_hangup(), and ast_pickup_call().

Referenced by sip_pickup().

{
   struct ast_channel *chan;
   chan = stuff;

   if (ast_pickup_call(chan)) {
      ast_channel_hangupcause_set(chan, AST_CAUSE_CALL_REJECTED);
   } else {
      ast_channel_hangupcause_set(chan, AST_CAUSE_NORMAL_CLEARING);
   }
   ast_hangup(chan);
   ast_channel_unref(chan);
   chan = NULL;
   return NULL;
}

static int sip_pidf_validate	(	struct sip_request *	req,
		struct ast_xml_doc **	pidf_doc
	)		[static]

Makes sure that body is properly formatted PIDF.

Specifically, we check that the document has a "presence" element at the root and that within that, there is at least one "tuple" element that contains a "status" element.

XXX This function currently assumes a default namespace is used. Of course if you're not using a default namespace, you're probably a stupid jerk anyway.

Parameters:

	req	The SIP request to check
[out]	pidf_doc	The validated PIDF doc.

Return values:

FALSE	The XML was malformed or the basic PIDF structure was marred
TRUE	The PIDF document is of a valid format

Definition at line 27285 of file chan_sip.c.

References ast_log(), ast_strlen_zero(), ast_xml_close(), ast_xml_read_memory(), FALSE, get_content(), LOG_WARNING, pidf_validate_presence(), sip_get_header(), and TRUE.

Referenced by cc_esc_publish_handler().

{
   struct ast_xml_doc *doc;
   const char *content_type = sip_get_header(req, "Content-Type");
   char *pidf_body;
   int res;

   if (ast_strlen_zero(content_type) || strcmp(content_type, "application/pidf+xml")) {
      ast_log(LOG_WARNING, "Content type is not PIDF\n");
      return FALSE;
   }

   if (!(pidf_body = get_content(req))) {
      ast_log(LOG_WARNING, "Unable to get PIDF body\n");
      return FALSE;
   }

   if (!(doc = ast_xml_read_memory(pidf_body, strlen(pidf_body)))) {
      ast_log(LOG_WARNING, "Unable to open XML PIDF document. Is it malformed?\n");
      return FALSE;
   }

   res = pidf_validate_presence(doc);
   if (res == TRUE) {
      *pidf_doc = doc;
   } else {
      ast_xml_close(doc);
   }
   return res;
}

static void sip_poke_all_peers

(

void

)

[static]

Send a poke to all known peers.

Definition at line 33193 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, AST_SCHED_REPLACE_UNREF, global_qualify_gap, sip_poke_peer_s(), sip_ref_peer(), and sip_unref_peer().

Referenced by load_module(), and sip_do_reload().

{
   int ms = 0, num = 0;
   struct ao2_iterator i;
   struct sip_peer *peer;

   if (!speerobjs) { /* No peers, just give up */
      return;
   }

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
      ao2_lock(peer);
      /* Don't schedule poking on a peer without qualify */
      if (peer->maxms) {
         if (num == global_qualify_peers) {
            ms += global_qualify_gap;
            num = 0;
         } else {
            num++;
         }
         AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, ms, sip_poke_peer_s, peer,
               sip_unref_peer(_data, "removing poke peer ref"),
               sip_unref_peer(peer, "removing poke peer ref"),
               sip_ref_peer(peer, "adding poke peer ref"));
      }
      ao2_unlock(peer);
      sip_unref_peer(peer, "toss iterator peer ptr");
   }
   ao2_iterator_destroy(&i);
}

static int sip_poke_noanswer

(

const void *

data

)

[static]

React to lack of answer to Qualify poke.

Definition at line 29561 of file chan_sip.c.

References ast_check_realtime(), AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), ast_log(), AST_SCHED_REPLACE_UNREF, ast_update_realtime(), DEFAULT_FREQ_NOTOK, dialog_unlink_all(), EVENT_FLAG_SYSTEM, FALSE, LOG_NOTICE, manager_event, register_peer_exten(), SENTINEL, sip_cfg, sip_poke_peer_s(), sip_ref_peer(), and sip_unref_peer().

Referenced by sip_poke_peer(), and sip_show_sched().

{
   struct sip_peer *peer = (struct sip_peer *)data;

   peer->pokeexpire = -1;

   if (peer->lastms > -1) {
      ast_log(LOG_NOTICE, "Peer '%s' is now UNREACHABLE!  Last qualify: %d\n", peer->name, peer->lastms);
      if (sip_cfg.peer_rtupdate) {
         ast_update_realtime(ast_check_realtime("sipregs") ? "sipregs" : "sippeers", "name", peer->name, "lastms", "-1", SENTINEL);
      }
      manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Unreachable\r\nTime: %d\r\n", peer->name, -1);
      if (sip_cfg.regextenonqualify) {
         register_peer_exten(peer, FALSE);
      }
   }

   if (peer->call) {
      dialog_unlink_all(peer->call);
      peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      /* peer->call = sip_destroy(peer->call);*/
   }

   /* Don't send a devstate change if nothing changed. */
   if (peer->lastms > -1) {
      peer->lastms = -1;
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", peer->name);
   }

   /* Try again quickly */
   AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched,
         DEFAULT_FREQ_NOTOK, sip_poke_peer_s, peer,
         sip_unref_peer(_data, "removing poke peer ref"),
         sip_unref_peer(peer, "removing poke peer ref"),
         sip_ref_peer(peer, "adding poke peer ref"));

   /* Release the ref held by the running scheduler entry */
   sip_unref_peer(peer, "release peer poke noanswer ref");

   return 0;
}

static int sip_poke_peer	(	struct sip_peer *	peer,
		int	force
	)		[static]

Check availability of peer, also keep NAT open.

Note:

This is done with 60 seconds between each ping, unless forced by cli or manager. If peer is unreachable, we check every 10th second by default.

Do *not* hold a pvt lock while calling this function. This function calls sip_alloc, which can cause a deadlock if another sip_pvt is held.

Definition at line 29611 of file chan_sip.c.

References ast_copy_flags, ast_copy_string(), ast_log(), AST_SCHED_DEL_UNREF, AST_SCHED_REPLACE_UNREF, ast_set_flag, ast_sip_ouraddrfor(), ast_sockaddr_isnull(), ast_sockaddr_stringify_host_remote(), ast_string_field_set, ast_strlen_zero(), ast_tvnow(), build_via(), change_callid_pvt(), copy_socket_data(), dialog_unlink_all(), LOG_NOTICE, sip_alloc(), sip_poke_noanswer(), sip_ref_peer(), sip_unref_peer(), and transmit_invite().

Referenced by _sip_qualify_peer(), build_peer(), parse_register_contact(), and sip_poke_peer_s().

{
   struct sip_pvt *p;
   int xmitres = 0;
   
   if ((!peer->maxms && !force) || ast_sockaddr_isnull(&peer->addr)) {
      /* IF we have no IP, or this isn't to be monitored, return
        immediately after clearing things out */
      AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
            sip_unref_peer(peer, "removing poke peer ref"));
      
      peer->lastms = 0;
      if (peer->call) {
         peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      }
      return 0;
   }
   if (peer->call) {
      if (sipdebug) {
         ast_log(LOG_NOTICE, "Still have a QUALIFY dialog active, deleting\n");
      }
      dialog_unlink_all(peer->call);
      peer->call = dialog_unref(peer->call, "unref dialog peer->call");
      /* peer->call = sip_destroy(peer->call); */
   }
   if (!(p = sip_alloc(NULL, NULL, 0, SIP_OPTIONS, NULL, NULL))) {
      return -1;
   }
   peer->call = dialog_ref(p, "copy sip alloc from p to peer->call");

   p->sa = peer->addr;
   p->recv = peer->addr;
   copy_socket_data(&p->socket, &peer->socket);
   ast_copy_flags(&p->flags[0], &peer->flags[0], SIP_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[1], &peer->flags[1], SIP_PAGE2_FLAGS_TO_COPY);
   ast_copy_flags(&p->flags[2], &peer->flags[2], SIP_PAGE3_FLAGS_TO_COPY);

   /* Send OPTIONs to peer's fullcontact */
   if (!ast_strlen_zero(peer->fullcontact)) {
      ast_string_field_set(p, fullcontact, peer->fullcontact);
   }

   if (!ast_strlen_zero(peer->fromuser)) {
      ast_string_field_set(p, fromuser, peer->fromuser);
   }

   if (!ast_strlen_zero(peer->tohost)) {
      ast_string_field_set(p, tohost, peer->tohost);
   } else {
      ast_string_field_set(p, tohost, ast_sockaddr_stringify_host_remote(&peer->addr));
   }

   /* Recalculate our side, and recalculate Call ID */
   ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
   build_via(p);

   /* Change the dialog callid. */
   change_callid_pvt(p, NULL);

   AST_SCHED_DEL_UNREF(sched, peer->pokeexpire,
         sip_unref_peer(peer, "removing poke peer ref"));
   
   if (p->relatedpeer)
      p->relatedpeer = sip_unref_peer(p->relatedpeer,"unsetting the relatedpeer field in the dialog, before it is set to something else.");
   p->relatedpeer = sip_ref_peer(peer, "setting the relatedpeer field in the dialog");
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
#ifdef VOCAL_DATA_HACK
   ast_copy_string(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
   xmitres = transmit_invite(p, SIP_INVITE, 0, 2, NULL); /* sinks the p refcount */
#else
   xmitres = transmit_invite(p, SIP_OPTIONS, 0, 2, NULL); /* sinks the p refcount */
#endif
   peer->ps = ast_tvnow();
   if (xmitres == XMIT_ERROR) {
      /* Immediately unreachable, network problems */
      sip_poke_noanswer(sip_ref_peer(peer, "add ref for peerexpire (fake, for sip_poke_noanswer to remove)"));
   } else if (!force) {
      AST_SCHED_REPLACE_UNREF(peer->pokeexpire, sched, peer->maxms * 2, sip_poke_noanswer, peer,
            sip_unref_peer(_data, "removing poke peer ref"),
            sip_unref_peer(peer, "removing poke peer ref"),
            sip_ref_peer(peer, "adding poke peer ref"));
   }
   dialog_unref(p, "unref dialog at end of sip_poke_peer, obtained from sip_alloc, just before it goes out of scope");
   return 0;
}

static int sip_poke_peer_s

(

const void *

data

)

[static]

Poke peer (send qualify to check if peer is alive and well)

Definition at line 15803 of file chan_sip.c.

References ao2_find, OBJ_POINTER, sip_poke_peer(), and sip_unref_peer().

Referenced by handle_response_peerpoke(), reg_source_db(), sip_poke_all_peers(), sip_poke_noanswer(), and sip_show_sched().

{
   struct sip_peer *peer = (struct sip_peer *)data;
   struct sip_peer *foundpeer;

   peer->pokeexpire = -1;

   foundpeer = ao2_find(peers, peer, OBJ_POINTER);
   if (!foundpeer) {
      sip_unref_peer(peer, "removing poke peer ref");
      return 0;
   } else if (foundpeer->name != peer->name) {
      sip_unref_peer(foundpeer, "removing above peer ref");
      sip_unref_peer(peer, "removing poke peer ref");
      return 0;
   }

   sip_unref_peer(foundpeer, "removing above peer ref");
   sip_poke_peer(peer, 0);
   sip_unref_peer(peer, "removing poke peer ref");

   return 0;
}

static int sip_prepare_socket

(

struct sip_pvt *

p

)

[static]

Todo:

Get socket for dialog, prepare if needed, and return file handle

Todo:

Check this... This might be wrong, depending on the proxy configuration If proxy is in "force" mode its correct.

Definition at line 28683 of file chan_sip.c.

References ast_tcptls_session_args::accept_fd, ao2_alloc, ao2_ref, ao2_t_ref, ao2_t_unlink, ast_calloc, ast_copy_string(), ast_debug, ast_pthread_create_detached_background, ast_sockaddr_copy(), ast_strdup, ast_strlen_zero(), ast_tcptls_client_create(), ast_tcptls_close_session_file(), ast_websocket_fd(), ast_tls_config::cafile, ast_tls_config::capath, ast_tls_config::certfile, ast_tls_config::cipher, ast_tcptls_session_instance::fd, ast_tcptls_session_args::hostname, name, ast_tcptls_session_args::name, ast_tls_config::pvtfile, ast_tcptls_session_args::remote_address, sip_real_dst(), sip_tcp_locate(), sip_tcp_worker_fn(), sip_tcptls_client_args_destructor(), sip_threadinfo_create(), sipsock, and ast_tcptls_session_args::tls_cfg.

Referenced by __sip_xmit().

{
   struct sip_socket *s = &p->socket;
   static const char name[] = "SIP socket";
   struct sip_threadinfo *th = NULL;
   struct ast_tcptls_session_instance *tcptls_session;
   struct ast_tcptls_session_args *ca;
   struct ast_sockaddr sa_tmp;
   pthread_t launched;

   /* check to see if a socket is already active */
   if ((s->fd != -1) && (s->type == SIP_TRANSPORT_UDP)) {
      return s->fd;
   }
   if ((s->type & (SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS)) &&
         (s->tcptls_session) &&
         (s->tcptls_session->fd != -1)) {
      return s->tcptls_session->fd;
   }
   if ((s->type & (SIP_TRANSPORT_WS | SIP_TRANSPORT_WSS))) {
      return s->ws_session ? ast_websocket_fd(s->ws_session) : -1;
   }

   /*! \todo Check this... This might be wrong, depending on the proxy configuration
      If proxy is in "force" mode its correct.
    */
   if (p->outboundproxy && p->outboundproxy->transport) {
      s->type = p->outboundproxy->transport;
   }

   if (s->type == SIP_TRANSPORT_UDP) {
      s->fd = sipsock;
      return s->fd;
   }

   /* At this point we are dealing with a TCP/TLS connection
    * 1. We need to check to see if a connection thread exists
    *    for this address, if so use that.
    * 2. If a thread does not exist for this address, but the tcptls_session
    *    exists on the socket, the connection was closed.
    * 3. If no tcptls_session thread exists for the address, and no tcptls_session
    *    already exists on the socket, create a new one and launch a new thread.
    */

   /* 1.  check for existing threads */
   ast_sockaddr_copy(&sa_tmp, sip_real_dst(p));
   if ((tcptls_session = sip_tcp_locate(&sa_tmp))) {
      s->fd = tcptls_session->fd;
      if (s->tcptls_session) {
         ao2_ref(s->tcptls_session, -1);
         s->tcptls_session = NULL;
      }
      s->tcptls_session = tcptls_session;
      return s->fd;
   /* 2.  Thread not found, if tcptls_session already exists, it once had a thread and is now terminated */
   } else if (s->tcptls_session) {
      return s->fd; /* XXX whether reconnection is ever necessary here needs to be investigated further */
   }

   /* 3.  Create a new TCP/TLS client connection */
   /* create new session arguments for the client connection */
   if (!(ca = ao2_alloc(sizeof(*ca), sip_tcptls_client_args_destructor)) ||
      !(ca->name = ast_strdup(name))) {
      goto create_tcptls_session_fail;
   }
   ca->accept_fd = -1;
   ast_sockaddr_copy(&ca->remote_address,sip_real_dst(p));
   /* if type is TLS, we need to create a tls cfg for this session arg */
   if (s->type == SIP_TRANSPORT_TLS) {
      if (!(ca->tls_cfg = ast_calloc(1, sizeof(*ca->tls_cfg)))) {
         goto create_tcptls_session_fail;
      }
      memcpy(ca->tls_cfg, &default_tls_cfg, sizeof(*ca->tls_cfg));

      if (!(ca->tls_cfg->certfile = ast_strdup(default_tls_cfg.certfile)) ||
         !(ca->tls_cfg->pvtfile = ast_strdup(default_tls_cfg.pvtfile)) ||
         !(ca->tls_cfg->cipher = ast_strdup(default_tls_cfg.cipher)) ||
         !(ca->tls_cfg->cafile = ast_strdup(default_tls_cfg.cafile)) ||
         !(ca->tls_cfg->capath = ast_strdup(default_tls_cfg.capath))) {

         goto create_tcptls_session_fail;
      }

      /* this host is used as the common name in ssl/tls */
      if (!ast_strlen_zero(p->tohost)) {
         ast_copy_string(ca->hostname, p->tohost, sizeof(ca->hostname));
      }
   }

   /* Create a client connection for address, this does not start the connection, just sets it up. */
   if (!(s->tcptls_session = ast_tcptls_client_create(ca))) {
      goto create_tcptls_session_fail;
   }

   s->fd = s->tcptls_session->fd;

   /* client connections need to have the sip_threadinfo object created before
    * the thread is detached.  This ensures the alert_pipe is up before it will
    * be used.  Note that this function links the new threadinfo object into the
    * threadt container. */
   if (!(th = sip_threadinfo_create(s->tcptls_session, s->type))) {
      goto create_tcptls_session_fail;
   }

   /* Give the new thread a reference to the tcptls_session */
   ao2_ref(s->tcptls_session, +1);

   if (ast_pthread_create_detached_background(&launched, NULL, sip_tcp_worker_fn, s->tcptls_session)) {
      ast_debug(1, "Unable to launch '%s'.", ca->name);
      ao2_ref(s->tcptls_session, -1); /* take away the thread ref we just gave it */
      goto create_tcptls_session_fail;
   }

   return s->fd;

create_tcptls_session_fail:
   if (ca) {
      ao2_t_ref(ca, -1, "failed to create client, getting rid of client tcptls_session arguments");
   }
   if (s->tcptls_session) {
      ast_tcptls_close_session_file(s->tcptls_session);
      s->fd = -1;
      ao2_ref(s->tcptls_session, -1);
      s->tcptls_session = NULL;
   }
   if (th) {
      ao2_t_unlink(threadt, th, "Removing tcptls thread info object, thread failed to open");
   }

   return -1;
}

static char * sip_prune_realtime	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Remove temporary realtime objects from memory (CLI)

Todo:

XXXX Propably needs an overhaul after removal of the devices

Definition at line 19575 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_find, ao2_t_iterator_next, ao2_t_link, ao2_t_unlink, ao2_unlock, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_cli_complete(), ast_copy_string(), ast_sockaddr_isnull(), ast_test_flag, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_peer(), FALSE, ast_cli_args::fd, ast_cli_args::n, name, OBJ_POINTER, OBJ_UNLINK, ast_cli_args::pos, sip_unref_peer(), TRUE, unlink_marked_peers_from_tables(), ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_peer *peer, *pi;
   int prunepeer = FALSE;
   int multi = FALSE;
   const char *name = NULL;
   regex_t regexbuf;
   int havepattern = 0;
   struct ao2_iterator i;
   static const char * const choices[] = { "all", "like", NULL };
   char *cmplt;
   
   if (cmd == CLI_INIT) {
      e->command = "sip prune realtime [peer|all]";
      e->usage =
         "Usage: sip prune realtime [peer [<name>|all|like <pattern>]|all]\n"
         "       Prunes object(s) from the cache.\n"
         "       Optional regular expression pattern is used to filter the objects.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE) {
      if (a->pos == 4 && !strcasecmp(a->argv[3], "peer")) {
         cmplt = ast_cli_complete(a->word, choices, a->n);
         if (!cmplt)
            cmplt = complete_sip_peer(a->word, a->n - sizeof(choices), SIP_PAGE2_RTCACHEFRIENDS);
         return cmplt;
      }
      if (a->pos == 5 && !strcasecmp(a->argv[4], "like"))
         return complete_sip_peer(a->word, a->n, SIP_PAGE2_RTCACHEFRIENDS);
      return NULL;
   }
   switch (a->argc) {
   case 4:
      name = a->argv[3];
      /* we accept a name in position 3, but keywords are not good. */
      if (!strcasecmp(name, "peer") || !strcasecmp(name, "like"))
         return CLI_SHOWUSAGE;
      prunepeer = TRUE;
      if (!strcasecmp(name, "all")) {
         multi = TRUE;
         name = NULL;
      }
      /* else a single name, already set */
      break;
   case 5:
      /* sip prune realtime {peer|like} name */
      name = a->argv[4];
      if (!strcasecmp(a->argv[3], "peer"))
         prunepeer = TRUE;
      else if (!strcasecmp(a->argv[3], "like")) {
         prunepeer = TRUE;
         multi = TRUE;
      } else
         return CLI_SHOWUSAGE;
      if (!strcasecmp(name, "like"))
         return CLI_SHOWUSAGE;
      if (!multi && !strcasecmp(name, "all")) {
         multi = TRUE;
         name = NULL;
      }
      break;
   case 6:
      name = a->argv[5];
      multi = TRUE;
      /* sip prune realtime {peer} like name */
      if (strcasecmp(a->argv[4], "like"))
         return CLI_SHOWUSAGE;
      if (!strcasecmp(a->argv[3], "peer")) {
         prunepeer = TRUE;
      } else
         return CLI_SHOWUSAGE;
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   if (multi && name) {
      if (regcomp(&regexbuf, name, REG_EXTENDED | REG_NOSUB)) {
         return CLI_SHOWUSAGE;
      }
      havepattern = 1;
   }

   if (multi) {
      if (prunepeer) {
         int pruned = 0;
         
         i = ao2_iterator_init(peers, 0);
         while ((pi = ao2_t_iterator_next(&i, "iterate thru peers table"))) {
            ao2_lock(pi);
            if (name && regexec(&regexbuf, pi->name, 0, NULL, 0)) {
               ao2_unlock(pi);
               sip_unref_peer(pi, "toss iterator peer ptr before continue");
               continue;
            };
            if (ast_test_flag(&pi->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
               pi->the_mark = 1;
               pruned++;
            }
            ao2_unlock(pi);
            sip_unref_peer(pi, "toss iterator peer ptr");
         }
         ao2_iterator_destroy(&i);
         if (pruned) {
            unlink_marked_peers_from_tables();
            ast_cli(a->fd, "%d peers pruned.\n", pruned);
         } else
            ast_cli(a->fd, "No peers found to prune.\n");
      }
   } else {
      if (prunepeer) {
         struct sip_peer tmp;
         ast_copy_string(tmp.name, name, sizeof(tmp.name));
         if ((peer = ao2_t_find(peers, &tmp, OBJ_POINTER | OBJ_UNLINK, "finding to unlink from peers"))) {
            if (!ast_sockaddr_isnull(&peer->addr)) {
               ao2_t_unlink(peers_by_ip, peer, "unlinking peer from peers_by_ip also");
            }
            if (!ast_test_flag(&peer->flags[1], SIP_PAGE2_RTCACHEFRIENDS)) {
               ast_cli(a->fd, "Peer '%s' is not a Realtime peer, cannot be pruned.\n", name);
               /* put it back! */
               ao2_t_link(peers, peer, "link peer into peer table");
               if (!ast_sockaddr_isnull(&peer->addr)) {
                  ao2_t_link(peers_by_ip, peer, "link peer into peers_by_ip table");
               }
            } else
               ast_cli(a->fd, "Peer '%s' pruned.\n", name);
            sip_unref_peer(peer, "sip_prune_realtime: sip_unref_peer: tossing temp peer ptr");
         } else
            ast_cli(a->fd, "Peer '%s' not found.\n", name);
      }
   }

   if (havepattern) {
      regfree(&regexbuf);
   }

   return CLI_SUCCESS;
}

static void sip_pvt_callid_set	(	struct sip_pvt *	pvt,
		struct ast_callid *	callid
	)		[static]

Definition at line 8618 of file chan_sip.c.

References ast_callid_ref, and ast_callid_unref.

Referenced by sip_alloc().

{
   if (pvt->logger_callid) {
      ast_callid_unref(pvt->logger_callid);
   }
   ast_callid_ref(callid);
   pvt->logger_callid = callid;
}

static struct ast_channel * sip_pvt_lock_full

(

struct sip_pvt *

pvt

)

[static, read]

Definition at line 9091 of file chan_sip.c.

References ast_channel_lock, ast_channel_ref, ast_channel_unlock, ast_channel_unref, sip_pvt_lock, and sip_pvt_unlock.

Referenced by __sip_autodestruct(), dialog_unlink_all(), handle_request_do(), reinvite_timeout(), send_provisional_keepalive_full(), and sip_queue_hangup_cause().

{
   struct ast_channel *chan;

   /* Locking is simple when it is done right.  If you see a deadlock resulting
    * in this function, it is not this function's fault, Your problem exists elsewhere.
    * This function is perfect... seriously. */
   for (;;) {
      /* First, get the channel and grab a reference to it */
      sip_pvt_lock(pvt);
      chan = pvt->owner;
      if (chan) {
         /* The channel can not go away while we hold the pvt lock.
          * Give the channel a ref so it will not go away after we let
          * the pvt lock go. */
         ast_channel_ref(chan);
      } else {
         /* no channel, return pvt locked */
         return NULL;
      }

      /* We had to hold the pvt lock while getting a ref to the owner channel
       * but now we have to let this lock go in order to preserve proper
       * locking order when grabbing the channel lock */
      sip_pvt_unlock(pvt);

      /* Look, no deadlock avoidance, hooray! */
      ast_channel_lock(chan);
      sip_pvt_lock(pvt);

      if (pvt->owner == chan) {
         /* done */
         break;
      }

      /* If the owner changed while everything was unlocked, no problem,
       * just start over and everthing will work.  This is rare, do not be
       * confused by this loop and think this it is an expensive operation.
       * The majority of the calls to this function will never involve multiple
       * executions of this loop. */
      ast_channel_unlock(chan);
      ast_channel_unref(chan);
      sip_pvt_unlock(pvt);
   }

   /* If owner exists, it is locked and reffed */
   return pvt->owner;
}

static char * sip_qualify_peer	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Send an OPTIONS packet to a SIP peer.

Definition at line 19946 of file chan_sip.c.

References _sip_qualify_peer(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, complete_sip_show_peer(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, ast_cli_entry::usage, and ast_cli_args::word.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip qualify peer";
      e->usage =
         "Usage: sip qualify peer <name> [load]\n"
         "       Requests a response from one SIP peer and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
   }
   return _sip_qualify_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}

static int sip_queryoption	(	struct ast_channel *	chan,
		int	option,
		void *	data,
		int *	datalen
	)		[static]

Query an option on a SIP dialog.

Definition at line 4909 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_copy_string(), ast_debug, ast_log(), AST_OPTION_DEVICE_NAME, AST_OPTION_DIGIT_DETECT, AST_OPTION_SECURE_MEDIA, AST_OPTION_SECURE_SIGNALING, AST_OPTION_T38_STATE, ast_test_flag, LOG_ERROR, sip_pvt_lock, sip_pvt_unlock, state, T38_STATE_NEGOTIATED, T38_STATE_NEGOTIATING, T38_STATE_REJECTED, T38_STATE_UNAVAILABLE, and T38_STATE_UNKNOWN.

{
   int res = -1;
   enum ast_t38_state state = T38_STATE_UNAVAILABLE;
   struct sip_pvt *p = (struct sip_pvt *) ast_channel_tech_pvt(chan);
   char *cp;

   sip_pvt_lock(p);

   switch (option) {
   case AST_OPTION_T38_STATE:
      /* Make sure we got an ast_t38_state enum passed in */
      if (*datalen != sizeof(enum ast_t38_state)) {
         ast_log(LOG_ERROR, "Invalid datalen for AST_OPTION_T38_STATE option. Expected %d, got %d\n", (int)sizeof(enum ast_t38_state), *datalen);
         break;
      }

      /* Now if T38 support is enabled we need to look and see what the current state is to get what we want to report back */
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT)) {
         switch (p->t38.state) {
         case T38_LOCAL_REINVITE:
         case T38_PEER_REINVITE:
            state = T38_STATE_NEGOTIATING;
            break;
         case T38_ENABLED:
            state = T38_STATE_NEGOTIATED;
            break;
         case T38_REJECTED:
            state = T38_STATE_REJECTED;
            break;
         default:
            state = T38_STATE_UNKNOWN;
         }
      }

      *((enum ast_t38_state *) data) = state;
      res = 0;

      break;
   case AST_OPTION_DIGIT_DETECT:
      cp = (char *) data;
      *cp = p->dsp ? 1 : 0;
      ast_debug(1, "Reporting digit detection %sabled on %s\n", *cp ? "en" : "dis", ast_channel_name(chan));
      break;
   case AST_OPTION_SECURE_SIGNALING:
      *((unsigned int *) data) = p->req_secure_signaling;
      res = 0;
      break;
   case AST_OPTION_SECURE_MEDIA:
      *((unsigned int *) data) = ast_test_flag(&p->flags[1], SIP_PAGE2_USE_SRTP) ? 1 : 0;
      res = 0;
      break;
   case AST_OPTION_DEVICE_NAME:
      if (p && p->outgoing_call) {
         cp = (char *) data;
         ast_copy_string(cp, p->dialstring, *datalen);
         res = 0;
      }
      /* We purposely break with a return of -1 in the
       * implied else case here
       */
      break;
   default:
      break;
   }

   sip_pvt_unlock(p);

   return res;
}

static void sip_queue_hangup_cause	(	struct sip_pvt *	p,
		int	cause
	)		[static]

Definition at line 22676 of file chan_sip.c.

References ast_channel_name(), ast_channel_ref, ast_channel_unlock, ast_channel_unref, ast_queue_hangup(), ast_queue_hangup_with_cause(), ast_set_hangupsource(), name, sip_pvt_lock_full(), and sip_pvt_unlock.

Referenced by handle_request_bye(), handle_request_cancel(), and handle_response_invite().

{
   struct ast_channel *owner = p->owner;
   const char *name = ast_strdupa(ast_channel_name(owner));

   /* Cannot hold any channel/private locks when calling. */
   ast_channel_ref(owner);
   ast_channel_unlock(owner);
   sip_pvt_unlock(p);
   ast_set_hangupsource(owner, name, 0);
   if (cause) {
      ast_queue_hangup_with_cause(owner, cause);
   } else {
      ast_queue_hangup(owner);
   }
   ast_channel_unref(owner);

   /* Relock things. */
   owner = sip_pvt_lock_full(p);
   if (owner) {
      ast_channel_unref(owner);
   }
}

static struct ast_frame * sip_read

(

struct ast_channel *

ast

)

[static, read]

Read SIP RTP from channel.

Definition at line 8429 of file chan_sip.c.

References ast_async_goto(), ast_channel_caller(), ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_macrocontext(), ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_exists_extension(), AST_FRAME_VOICE, ast_frfree, ast_log(), ast_null_frame, AST_STATE_UP, ast_test_flag, ast_verb, FALSE, ast_frame::frametype, LOG_NOTICE, pbx_builtin_setvar_helper(), S_COR, S_OR, sip_pvt_lock, sip_pvt_unlock, and sip_rtp_read().

{
   struct ast_frame *fr;
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int faxdetected = FALSE;

   sip_pvt_lock(p);
   fr = sip_rtp_read(ast, p, &faxdetected);
   p->lastrtprx = time(NULL);

   /* If we detect a CNG tone and fax detection is enabled then send us off to the fax extension */
   if (faxdetected && ast_test_flag(&p->flags[1], SIP_PAGE2_FAX_DETECT_CNG)) {
      if (strcmp(ast_channel_exten(ast), "fax")) {
         const char *target_context = S_OR(ast_channel_macrocontext(ast), ast_channel_context(ast));
         /* We need to unlock 'ast' here because
          * ast_exists_extension has the potential to start and
          * stop an autoservice on the channel. Such action is
          * prone to deadlock if the channel is locked.
          */
         sip_pvt_unlock(p);
         ast_channel_unlock(ast);
         if (ast_exists_extension(ast, target_context, "fax", 1,
            S_COR(ast_channel_caller(ast)->id.number.valid, ast_channel_caller(ast)->id.number.str, NULL))) {
            ast_channel_lock(ast);
            sip_pvt_lock(p);
            ast_verb(2, "Redirecting '%s' to fax extension due to CNG detection\n", ast_channel_name(ast));
            pbx_builtin_setvar_helper(ast, "FAXEXTEN", ast_channel_exten(ast));
            if (ast_async_goto(ast, target_context, "fax", 1)) {
               ast_log(LOG_NOTICE, "Failed to async goto '%s' into fax of '%s'\n", ast_channel_name(ast), target_context);
            }
            ast_frfree(fr);
            fr = &ast_null_frame;
         } else {
            ast_channel_lock(ast);
            sip_pvt_lock(p);
            ast_log(LOG_NOTICE, "FAX CNG detected but no fax extension\n");
         }
      }
   }

   /* Only allow audio through if they sent progress with SDP, or if the channel is actually answered */
   if (fr && fr->frametype == AST_FRAME_VOICE && p->invitestate != INV_EARLY_MEDIA && ast_channel_state(ast) != AST_STATE_UP) {
      ast_frfree(fr);
      fr = &ast_null_frame;
   }

   sip_pvt_unlock(p);

   return fr;
}

static struct ast_sockaddr * sip_real_dst

(

const struct sip_pvt *

p

)

[static, read]

The real destination address for a write.

Definition at line 3703 of file chan_sip.c.

References ast_test_flag.

Referenced by __sip_xmit(), check_via(), retrans_pkt(), send_response(), show_channels_cb(), sip_debug_test_pvt(), and sip_prepare_socket().

{
   if (p->outboundproxy) {
      return &p->outboundproxy->ip;
   }

   return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT) ? &p->recv : &p->sa;
}

static const char* sip_reason_code_to_str

(

enum AST_REDIRECTING_REASON

code

)

[static]

Definition at line 2396 of file chan_sip.c.

References ARRAY_LEN, sip_reason_table, and sip_reasons::text.

Referenced by add_diversion().

{
   if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
      return sip_reason_table[code].text;
   }

   return "unknown";
}

static enum AST_REDIRECTING_REASON sip_reason_str_to_code

(

const char *

text

)

[static]

Definition at line 2381 of file chan_sip.c.

References ARRAY_LEN, AST_REDIRECTING_REASON_UNKNOWN, sip_reasons::code, and sip_reason_table.

Referenced by get_rdnis().

{
   enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
   int i;

   for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
      if (!strcasecmp(text, sip_reason_table[i].text)) {
         ast = sip_reason_table[i].code;
         break;
      }
   }

   return ast;
}

struct sip_peer* sip_ref_peer	(	struct sip_peer *	peer,
		char *	tag
	)		[read]

Definition at line 3284 of file chan_sip.c.

References ao2_t_ref.

Referenced by build_peer(), check_peer_ok(), create_addr(), handle_request_invite(), handle_request_subscribe(), handle_response_peerpoke(), parse_register_contact(), realtime_peer(), reg_source_db(), register_verify(), sip_keepalive_all_peers(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), sip_send_keepalive(), sip_unregister(), and update_call_counter().

{
   ao2_t_ref(peer, 1, tag);
   return peer;
}

static int sip_refer_alloc

(

struct sip_pvt *

p

)

[static]

Allocate SIP refer structure.

Definition at line 15486 of file chan_sip.c.

References ast_calloc_with_stringfields.

Referenced by get_also_info(), handle_request_invite(), handle_request_refer(), and transmit_refer().

{
   p->refer = ast_calloc_with_stringfields(1, struct sip_refer, 512);
   return p->refer ? 1 : 0;
}

static int sip_reg_timeout

(

const void *

data

)

[static]

Registration timeout, register again Registered as a timeout handler during transmit_register(), to retransmit the packet if a reply does not come back. This is called by the scheduler so the event is not pending anymore when we are called.

Definition at line 15133 of file chan_sip.c.

References __sip_pretend_ack(), ast_dnsmgr_refresh(), ast_log(), EVENT_FLAG_SYSTEM, LOG_NOTICE, manager_event, pvt_set_needdestroy(), REG_STATE_UNREGISTERED, registry_unref(), regstate2str(), sip_pvt_lock, sip_pvt_unlock, and transmit_register().

Referenced by sip_show_sched(), and transmit_register().

{

   /* if we are here, our registration timed out, so we'll just do it over */
   struct sip_registry *r = (struct sip_registry *)data; /* the ref count should have been bumped when the sched item was added */
   struct sip_pvt *p;

   /* if we couldn't get a reference to the registry object, punt */
   if (!r) {
      return 0;
   }

   if (r->dnsmgr) {
      /* If the registration has timed out, maybe the IP changed.  Force a refresh. */
      ast_dnsmgr_refresh(r->dnsmgr);
   }

   /* If the initial tranmission failed, we may not have an existing dialog,
    * so it is possible that r->call == NULL.
    * Otherwise destroy it, as we have a timeout so we don't want it.
    */
   if (r->call) {
      /* Unlink us, destroy old call.  Locking is not relevant here because all this happens
         in the single SIP manager thread. */
      p = r->call;
      sip_pvt_lock(p);
      pvt_set_needdestroy(p, "registration timeout");
      /* Pretend to ACK anything just in case */
      __sip_pretend_ack(p);
      sip_pvt_unlock(p);

      /* decouple the two objects */
      /* p->registry == r, so r has 2 refs, and the unref won't take the object away */
      if (p->registry) {
         p->registry = registry_unref(p->registry, "p->registry unreffed");
      }
      r->call = dialog_unref(r->call, "unrefing r->call");
   }
   /* If we have a limit, stop registration and give up */
   r->timeout = -1;
   if (global_regattempts_max && r->regattempts >= global_regattempts_max) {
      /* Ok, enough is enough. Don't try any more */
      /* We could add an external notification here...
         steal it from app_voicemail :-) */
      ast_log(LOG_NOTICE, "   -- Last Registration Attempt #%d failed, Giving up forever trying to register '%s@%s'\n", r->regattempts, r->username, r->hostname);
      r->regstate = REG_STATE_FAILED;
   } else {
      r->regstate = REG_STATE_UNREGISTERED;
      transmit_register(r, SIP_REGISTER, NULL, NULL);
      ast_log(LOG_NOTICE, "   -- Registration for '%s@%s' timed out, trying again (Attempt #%d)\n", r->username, r->hostname, r->regattempts);
   }
   manager_event(EVENT_FLAG_SYSTEM, "Registry", "ChannelType: SIP\r\nUsername: %s\r\nDomain: %s\r\nStatus: %s\r\n", r->username, r->hostname, regstate2str(r->regstate));
   registry_unref(r, "unreffing registry_unref r");
   return 0;
}

static int sip_register	(	const char *	value,
		int	lineno
	)		[static]

create sip_registry object from register=> line in sip.conf and link into reg container

Definition at line 9344 of file chan_sip.c.

References ast_atomic_fetchadd_int(), ast_calloc_with_stringfields, ast_copy_string(), ast_log(), ASTOBJ_CONTAINER_FIND, ASTOBJ_CONTAINER_LINK, ASTOBJ_INIT, default_expiry, LOG_ERROR, registry_unref(), regl, and sip_parse_register_line().

Referenced by build_peer(), and reload_config().

{
   struct sip_registry *reg, *tmp;

   if (!(reg = ast_calloc_with_stringfields(1, struct sip_registry, 256))) {
      ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
      return -1;
   }

   ASTOBJ_INIT(reg);

   ast_copy_string(reg->name, value, sizeof(reg->name));
   if (sip_parse_register_line(reg, default_expiry, value, lineno)) {
      registry_unref(reg, "failure to parse, unref the reg pointer");
      return -1;
   }

   /* set default expiry if necessary */
   if (reg->refresh && !reg->expiry && !reg->configured_expiry) {
      reg->refresh = reg->expiry = reg->configured_expiry = default_expiry;
   }

   /* Add the new registry entry to the list, but only if it isn't already there */
   if ((tmp = ASTOBJ_CONTAINER_FIND(&regl, reg->name))) {
      registry_unref(tmp, "throw away found registry");
   } else {
      ast_atomic_fetchadd_int(&regobjs, 1);
      ASTOBJ_CONTAINER_LINK(&regl, reg);
   }

   /* release the reference given by ASTOBJ_INIT. The container has another reference */
   registry_unref(reg, "unref the reg pointer");

   return 0;
}

static void sip_register_tests

(

void

)

[static]

SIP test registration.

Definition at line 33644 of file chan_sip.c.

References sip_config_parser_register_tests(), sip_dialplan_function_register_tests(), and sip_request_parser_register_tests().

Referenced by load_module().

{
   sip_config_parser_register_tests();
   sip_request_parser_register_tests();
   sip_dialplan_function_register_tests();
}

static void sip_registry_destroy

(

struct sip_registry *

reg

)

[static]

Destroy registry object Objects created with the register= statement in static configuration.

Definition at line 6427 of file chan_sip.c.

References ast_atomic_fetchadd_int(), ast_debug, ast_free, AST_SCHED_DEL, ast_string_field_free_memory, dialog_unlink_all(), and registry_unref().

Referenced by registry_unref(), reload_config(), and unload_module().

{
   /* Really delete */
   ast_debug(3, "Destroying registry entry for %s@%s\n", reg->username, reg->hostname);

   if (reg->call) {
      /* Clear registry before destroying to ensure
         we don't get reentered trying to grab the registry lock */
      reg->call->registry = registry_unref(reg->call->registry, "destroy reg->call->registry");
      ast_debug(3, "Destroying active SIP dialog for registry %s@%s\n", reg->username, reg->hostname);
      dialog_unlink_all(reg->call);
      reg->call = dialog_unref(reg->call, "unref reg->call");
      /* reg->call = sip_destroy(reg->call); */
   }
   AST_SCHED_DEL(sched, reg->expire);
   AST_SCHED_DEL(sched, reg->timeout);

   ast_string_field_free_memory(reg);
   ast_atomic_fetchadd_int(&regobjs, -1);
   ast_free(reg);
}

static int sip_reinvite_retry

(

const void *

data

)

[static]

Reset the NEEDREINVITE flag after waiting when we get 491 on a Re-invite to avoid race conditions between asterisk servers. Called from the scheduler.

Definition at line 22528 of file chan_sip.c.

References ast_channel_trylock, ast_channel_unlock, ast_set_flag, check_pendings(), sip_pvt_lock, and sip_pvt_unlock.

Referenced by handle_response_invite(), and sip_show_sched().

{
   struct sip_pvt *p = (struct sip_pvt *) data;
   struct ast_channel *owner;

   sip_pvt_lock(p); /* called from schedule thread which requires a lock */
   while ((owner = p->owner) && ast_channel_trylock(owner)) {
      sip_pvt_unlock(p);
      usleep(1);
      sip_pvt_lock(p);
   }
   ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
   p->waitid = -1;
   check_pendings(p);
   sip_pvt_unlock(p);
   if (owner) {
      ast_channel_unlock(owner);
   }
   dialog_unref(p, "unref the dialog ptr from sip_reinvite_retry, because it held a dialog ptr");
   return 0;
}

static char * sip_reload	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Force reload of module from cli.

Definition at line 33380 of file chan_sip.c.

References ao2_t_ref, ast_clear_flag, ast_log(), ast_mutex_lock, ast_mutex_unlock, ast_string_field_set, ast_verbose(), bogus_peer, BOGUS_PEER_MD5SECRET, CHANNEL_CLI_RELOAD, CHANNEL_MODULE_RELOAD, CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, LOG_ERROR, restart_monitor(), sip_reload_lock, temp_peer(), TRUE, and ast_cli_entry::usage.

Referenced by reload().

{
   static struct sip_peer *tmp_peer, *new_peer;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip reload";
      e->usage =
         "Usage: sip reload\n"
         "       Reloads SIP configuration from sip.conf\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   ast_mutex_lock(&sip_reload_lock);
   if (sip_reloading) {
      ast_verbose("Previous SIP reload not yet done\n");
   } else {
      sip_reloading = TRUE;
      sip_reloadreason = (a && a->fd) ? CHANNEL_CLI_RELOAD : CHANNEL_MODULE_RELOAD;
   }
   ast_mutex_unlock(&sip_reload_lock);
   restart_monitor();

   tmp_peer = bogus_peer;
   /* Create new bogus peer possibly with new global settings. */
   if ((new_peer = temp_peer("(bogus_peer)"))) {
      ast_string_field_set(new_peer, md5secret, BOGUS_PEER_MD5SECRET);
      ast_clear_flag(&new_peer->flags[0], SIP_INSECURE);
      bogus_peer = new_peer;
      ao2_t_ref(tmp_peer, -1, "unref the old bogus_peer during reload");
   } else {
      ast_log(LOG_ERROR, "Could not update the fake authentication peer.\n");
      /* You probably have bigger (memory?) issues to worry about though.. */
   }

   return CLI_SUCCESS;
}

static int sip_removeheader	(	struct ast_channel *	chan,
		const char *	data
	)		[static]

Remove SIP headers added previously with SipAddHeader application.

Definition at line 33069 of file chan_sip.c.

References ast_channel_lock, ast_channel_unlock, ast_channel_varshead(), ast_debug, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, ast_strlen_zero(), ast_var_delete(), ast_var_name(), ast_var_value(), and inbuf().

Referenced by load_module().

{
   struct ast_var_t *newvariable;
   struct varshead *headp;
   int removeall = 0;
   char *inbuf = (char *) data;

   if (ast_strlen_zero(inbuf)) {
      removeall = 1;
   }
   ast_channel_lock(chan);

   headp=ast_channel_varshead(chan);
   AST_LIST_TRAVERSE_SAFE_BEGIN (headp, newvariable, entries) {
      if (strncasecmp(ast_var_name(newvariable), "SIPADDHEADER", strlen("SIPADDHEADER")) == 0) {
         if (removeall || (!strncasecmp(ast_var_value(newvariable),inbuf,strlen(inbuf)))) {
            if (sipdebug) {
               ast_debug(1,"removing SIP Header \"%s\" as %s\n",
                  ast_var_value(newvariable),
                  ast_var_name(newvariable));
            }
            AST_LIST_REMOVE_CURRENT(entries);
            ast_var_delete(newvariable);
         }
      }
   }
   AST_LIST_TRAVERSE_SAFE_END;

   ast_channel_unlock(chan);
   return 0;
}

static struct ast_channel * sip_request_call	(	const char *	type,
		struct ast_format_cap *	cap,
		const struct ast_channel *	requestor,
		const char *	dest,
		int *	cause
	)		[static, read]

PBX interface function -build SIP pvt structure SIP calls initiated by the PBX arrive here.

 *	SIP Dial string syntax:
 *		SIP/devicename
 *	or	SIP/username@domain (SIP uri)
 *	or	SIP/username[:password[:md5secret[:authname[:transport]]]]@host[:port]
 *	or	SIP/devicename/extension
 *	or	SIP/devicename/extension/IPorHost
 *	or	SIP/username@domain//IPorHost
 *	and there is an optional [!dnid] argument you can append to alter the
 *	To: header.
 * 

Definition at line 29812 of file chan_sip.c.

References __set_address_from_contact(), args, AST_APP_ARG, ast_callid_unref, ast_calloc, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL, AST_CAUSE_CHANNEL_UNACCEPTABLE, AST_CAUSE_SWITCH_CONGESTION, AST_CAUSE_UNREGISTERED, ast_channel_linkedid(), ast_channel_name(), ast_channel_unlock, ast_copy_string(), ast_debug, AST_DECLARE_APP_ARGS, ast_format_cap_append(), ast_format_cap_has_type(), ast_format_cap_joint_copy(), AST_FORMAT_TYPE_AUDIO, ast_getformatname_multiple(), ast_log(), AST_NONSTANDARD_APP_ARGS, ast_read_threadstorage_callid(), ast_sip_ouraddrfor(), AST_STATE_DOWN, ast_string_field_set, ast_strlen_zero(), ast_test_flag, ast_update_use_count(), build_via(), change_callid_pvt(), check_for_nat(), create_addr(), dialog_unlink_all(), do_setnat(), EVENT_FLAG_SYSTEM, ext, exten, LOG_ERROR, LOG_NOTICE, LOG_WARNING, manager_event, proxy_from_config(), restart_monitor(), secret, set_peer_nat(), set_socket_transport(), sip_alloc(), sip_cfg, sip_new(), sip_pvt_lock, sip_pvt_unlock, and TRUE.

{
   struct sip_pvt *p;
   struct ast_channel *tmpc = NULL;
   char *ext = NULL, *host;
   char tmp[256];
   char tmp2[256];
   char *dnid;
   char *secret = NULL;
   char *md5secret = NULL;
   char *authname = NULL;
   char *trans = NULL;
   char dialstring[256];
   char *remote_address;
   enum sip_transport transport = 0;
   struct ast_callid *callid;
   AST_DECLARE_APP_ARGS(args,
      AST_APP_ARG(peerorhost);
      AST_APP_ARG(exten);
      AST_APP_ARG(remote_address);
   );

   /* mask request with some set of allowed formats.
    * XXX this needs to be fixed.
    * The original code uses AST_FORMAT_AUDIO_MASK, but it is
    * unclear what to use here. We have global_capabilities, which is
    * configured from sip.conf, and sip_tech.capabilities, which is
    * hardwired to all audio formats.
    */
   if (!(ast_format_cap_has_type(cap, AST_FORMAT_TYPE_AUDIO))) {
      ast_log(LOG_NOTICE, "Asked to get a channel of unsupported format %s while capability is %s\n",
      ast_getformatname_multiple(tmp, sizeof(tmp), cap),
      ast_getformatname_multiple(tmp2, sizeof(tmp2), sip_cfg.caps));
      *cause = AST_CAUSE_BEARERCAPABILITY_NOTAVAIL;   /* Can't find codec to connect to host */
      return NULL;
   }
   ast_debug(1, "Asked to create a SIP channel with formats: %s\n", ast_getformatname_multiple(tmp, sizeof(tmp), cap));

   if (ast_strlen_zero(dest)) {
      ast_log(LOG_ERROR, "Unable to create channel with empty destination.\n");
      *cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
      return NULL;
   }

   callid = ast_read_threadstorage_callid();
   if (!(p = sip_alloc(NULL, NULL, 0, SIP_INVITE, NULL, callid))) {
      ast_log(LOG_ERROR, "Unable to build sip pvt data for '%s' (Out of memory or socket error)\n", dest);
      *cause = AST_CAUSE_SWITCH_CONGESTION;
      if (callid) {
         ast_callid_unref(callid);
      }
      return NULL;
   }

   p->outgoing_call = TRUE;

   snprintf(dialstring, sizeof(dialstring), "%s/%s", type, dest);
   ast_string_field_set(p, dialstring, dialstring);

   if (!(p->options = ast_calloc(1, sizeof(*p->options)))) {
      dialog_unlink_all(p);
      dialog_unref(p, "unref dialog p from mem fail");
      /* sip_destroy(p); */
      ast_log(LOG_ERROR, "Unable to build option SIP data structure - Out of memory\n");
      *cause = AST_CAUSE_SWITCH_CONGESTION;
      if (callid) {
         ast_callid_unref(callid);
      }
      return NULL;
   }

   /* Save the destination, the SIP dial string */
   ast_copy_string(tmp, dest, sizeof(tmp));

   /* Find DNID and take it away */
   dnid = strchr(tmp, '!');
   if (dnid != NULL) {
      *dnid++ = '\0';
      ast_string_field_set(p, todnid, dnid);
   }

   /* Divvy up the items separated by slashes */
   AST_NONSTANDARD_APP_ARGS(args, tmp, '/');

   /* Find at sign - @ */
   host = strchr(args.peerorhost, '@');
   if (host) {
      *host++ = '\0';
      ext = args.peerorhost;
      secret = strchr(ext, ':');
   }
   if (secret) {
      *secret++ = '\0';
      md5secret = strchr(secret, ':');
   }
   if (md5secret) {
      *md5secret++ = '\0';
      authname = strchr(md5secret, ':');
   }
   if (authname) {
      *authname++ = '\0';
      trans = strchr(authname, ':');
   }
   if (trans) {
      *trans++ = '\0';
      if (!strcasecmp(trans, "tcp"))
         transport = SIP_TRANSPORT_TCP;
      else if (!strcasecmp(trans, "tls"))
         transport = SIP_TRANSPORT_TLS;
      else {
         if (strcasecmp(trans, "udp"))
            ast_log(LOG_WARNING, "'%s' is not a valid transport option to Dial() for SIP calls, using udp by default.\n", trans);
         transport = SIP_TRANSPORT_UDP;
      }
   } else { /* use default */
      transport = SIP_TRANSPORT_UDP;
   }

   if (!host) {
      ext = args.exten;
      host = args.peerorhost;
      remote_address = args.remote_address;
   } else {
      remote_address = args.remote_address;
      if (!ast_strlen_zero(args.exten)) {
         ast_log(LOG_NOTICE, "Conflicting extension values given. Using '%s' and not '%s'\n", ext, args.exten);
      }
   }

   if (!ast_strlen_zero(remote_address)) {
      p->options->outboundproxy = proxy_from_config(remote_address, 0, NULL);
      if (!p->options->outboundproxy) {
         ast_log(LOG_WARNING, "Unable to parse outboundproxy %s. We will not use this remote IP address\n", remote_address);
      }
   }

   set_socket_transport(&p->socket, transport);

   /* We now have
      host = peer name, DNS host name or DNS domain (for SRV)
      ext = extension (user part of URI)
      dnid = destination of the call (applies to the To: header)
   */
   if (create_addr(p, host, NULL, 1)) {
      *cause = AST_CAUSE_UNREGISTERED;
      ast_debug(3, "Cant create SIP call - target device not registered\n");
      dialog_unlink_all(p);
      dialog_unref(p, "unref dialog p UNREGISTERED");
      /* sip_destroy(p); */
      if (callid) {
         ast_callid_unref(callid);
      }
      return NULL;
   }
   if (ast_strlen_zero(p->peername) && ext)
      ast_string_field_set(p, peername, ext);
   /* Recalculate our side, and recalculate Call ID */
   ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
   /* When chan_sip is first loaded or reloaded, we need to check for NAT and set the appropiate flags
      now that we have the auto_* settings. */
   check_for_nat(&p->sa, p);
   /* If there is a peer related to this outgoing call and it hasn't re-registered after
      a reload, we need to set the peer's NAT flags accordingly. */
   if (p->relatedpeer) {

      if (!ast_strlen_zero(p->relatedpeer->fullcontact) && !p->natdetected &&
         (ast_test_flag(&p->flags[2], SIP_PAGE3_NAT_AUTO_RPORT) && !ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT))) {
         /* We need to make an attempt to determine if a peer is behind NAT
            if the peer has the auto_force_rport flag set. */
         struct ast_sockaddr tmpaddr;

         __set_address_from_contact(p->relatedpeer->fullcontact, &tmpaddr, 0);

         check_for_nat(&tmpaddr, p);
      }

      set_peer_nat(p, p->relatedpeer);
   }

   do_setnat(p);

   build_via(p);

   /* Change the dialog callid. */
   change_callid_pvt(p, NULL);

   /* We have an extension to call, don't use the full contact here */
   /* This to enable dialing registered peers with extension dialling,
      like SIP/peername/extension   
      SIP/peername will still use the full contact
    */
   if (ext) {
      ast_string_field_set(p, username, ext);
      ast_string_field_set(p, fullcontact, NULL);
   }
   if (secret && !ast_strlen_zero(secret))
      ast_string_field_set(p, peersecret, secret);

   if (md5secret && !ast_strlen_zero(md5secret))
      ast_string_field_set(p, peermd5secret, md5secret);

   if (authname && !ast_strlen_zero(authname))
      ast_string_field_set(p, authname, authname);
#if 0
   printf("Setting up to call extension '%s' at '%s'\n", ext ? ext : "<none>", host);
#endif
   ast_format_cap_append(p->prefcaps, cap);
   ast_format_cap_joint_copy(cap, p->caps, p->jointcaps);

   sip_pvt_lock(p);

   tmpc = sip_new(p, AST_STATE_DOWN, host, requestor ? ast_channel_linkedid(requestor) : NULL, callid);  /* Place the call */
   if (callid) {
      callid = ast_callid_unref(callid);
   }

   if (sip_cfg.callevents)
      manager_event(EVENT_FLAG_SYSTEM, "ChannelUpdate",
         "Channel: %s\r\nChanneltype: %s\r\nSIPcallid: %s\r\nSIPfullcontact: %s\r\nPeername: %s\r\n",
         p->owner ? ast_channel_name(p->owner) : "", "SIP", p->callid, p->fullcontact, p->peername);
   sip_pvt_unlock(p);
   if (!tmpc) {
      dialog_unlink_all(p);
      /* sip_destroy(p); */
   } else {
      ast_channel_unlock(tmpc);
   }
   dialog_unref(p, "toss pvt ptr at end of sip_request_call");
   ast_update_use_count();
   restart_monitor();
   return tmpc;
}

static int sip_reregister

(

const void *

data

)

[static]

Update registration with SIP Proxy. Called from the scheduler when the previous registration expires, so we don't have to cancel the pending event. We assume the reference so the sip_registry is valid, since it is stored in the scheduled event anyways.

Definition at line 15090 of file chan_sip.c.

References __sip_do_register(), append_history, ast_log(), LOG_NOTICE, and registry_unref().

Referenced by handle_response_register(), sip_send_all_registers(), and sip_show_sched().

{
   /* if we are here, we know that we need to reregister. */
   struct sip_registry *r = (struct sip_registry *) data;

   /* if we couldn't get a reference to the registry object, punt */
   if (!r) {
      return 0;
   }

   if (r->call && r->call->do_history) {
      append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
   }
   /* Since registry's are only added/removed by the the monitor thread, this
      may be overkill to reference/dereference at all here */
   if (sipdebug) {
      ast_log(LOG_NOTICE, "   -- Re-registration for  %s@%s\n", r->username, r->hostname);
   }

   r->expire = -1;
   r->expiry = r->configured_expiry;
   __sip_do_register(r);
   registry_unref(r, "unref the re-register scheduled event");
   return 0;
}

static struct ast_frame* sip_rtp_read	(	struct ast_channel *	ast,
		struct sip_pvt *	p,
		int *	faxdetect
	)		[static, read]

Read RTP from network.

Definition at line 8327 of file chan_sip.c.

References ast_channel_fdno(), ast_channel_name(), ast_channel_nativeformats(), ast_channel_readformat(), ast_channel_writeformat(), ast_debug, ast_dsp_free(), ast_dsp_process(), ast_dsp_set_features(), ast_format_cap_add(), ast_format_cap_iscompatible(), ast_format_cap_remove_bytype(), AST_FORMAT_TYPE_AUDIO, AST_FRAME_DTMF, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_VOICE, ast_free, ast_frfree, ast_getformatname(), ast_null_frame, ast_rtp_instance_read(), ast_set_read_format(), ast_set_write_format(), ast_str_append(), ast_str_buffer(), ast_str_create(), ast_test_flag, ast_udptl_read(), ast_verb, ast_frame::data, ast_frame::datalen, DSP_FEATURE_DIGIT_DETECT, f, ast_frame_subclass::format, ast_frame::frametype, ast_frame_subclass::integer, ast_frame::ptr, and ast_frame::subclass.

Referenced by sip_read().

{
   /* Retrieve audio/etc from channel.  Assumes p->lock is already held. */
   struct ast_frame *f;
   
   if (!p->rtp) {
      /* We have no RTP allocated for this channel */
      return &ast_null_frame;
   }

   switch(ast_channel_fdno(ast)) {
   case 0:
      f = ast_rtp_instance_read(p->rtp, 0);  /* RTP Audio */
      break;
   case 1:
      f = ast_rtp_instance_read(p->rtp, 1);  /* RTCP Control Channel */
      break;
   case 2:
      f = ast_rtp_instance_read(p->vrtp, 0); /* RTP Video */
      break;
   case 3:
      f = ast_rtp_instance_read(p->vrtp, 1); /* RTCP Control Channel for video */
      break;
   case 4:
      f = ast_rtp_instance_read(p->trtp, 0); /* RTP Text */
      if (sipdebug_text) {
         struct ast_str *out = ast_str_create(f->datalen * 4 + 6);
         int i;
         unsigned char* arr = f->data.ptr;
         do {
            if (!out) {
               break;
            }
            for (i = 0; i < f->datalen; i++) {
               ast_str_append(&out, 0, "%c", (arr[i] > ' ' && arr[i] < '}') ? arr[i] : '.');
            }
            ast_str_append(&out, 0, " -> ");
            for (i = 0; i < f->datalen; i++) {
               ast_str_append(&out, 0, "%02X ", arr[i]);
            }
            ast_verb(0, "%s\n", ast_str_buffer(out));
            ast_free(out);
         } while (0);
      }
      break;
   case 5:
      f = ast_udptl_read(p->udptl); /* UDPTL for T.38 */
      break;
   default:
      f = &ast_null_frame;
   }
   /* Don't forward RFC2833 if we're not supposed to */
   if (f && (f->frametype == AST_FRAME_DTMF_BEGIN || f->frametype == AST_FRAME_DTMF_END) &&
       (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833)) {
      ast_debug(1, "Ignoring DTMF (%c) RTP frame because dtmfmode is not RFC2833\n", f->subclass.integer);
      ast_frfree(f);
      return &ast_null_frame;
   }

   /* We already hold the channel lock */
   if (!p->owner || (f && f->frametype != AST_FRAME_VOICE)) {
      return f;
   }

   if (f && !ast_format_cap_iscompatible(ast_channel_nativeformats(p->owner), &f->subclass.format)) {
      if (!ast_format_cap_iscompatible(p->jointcaps, &f->subclass.format)) {
         ast_debug(1, "Bogus frame of format '%s' received from '%s'!\n",
            ast_getformatname(&f->subclass.format), ast_channel_name(p->owner));
         ast_frfree(f);
         return &ast_null_frame;
      }
      ast_debug(1, "Oooh, format changed to %s\n",
         ast_getformatname(&f->subclass.format));
      ast_format_cap_remove_bytype(ast_channel_nativeformats(p->owner), AST_FORMAT_TYPE_AUDIO);
      ast_format_cap_add(ast_channel_nativeformats(p->owner), &f->subclass.format);
      ast_set_read_format(p->owner, ast_channel_readformat(p->owner));
      ast_set_write_format(p->owner, ast_channel_writeformat(p->owner));
   }

   if (f && p->dsp) {
      f = ast_dsp_process(p->owner, p->dsp, f);
      if (f && f->frametype == AST_FRAME_DTMF) {
         if (f->subclass.integer == 'f') {
            ast_debug(1, "Fax CNG detected on %s\n", ast_channel_name(ast));
            *faxdetect = 1;
            /* If we only needed this DSP for fax detection purposes we can just drop it now */
            if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) {
               ast_dsp_set_features(p->dsp, DSP_FEATURE_DIGIT_DETECT);
            } else {
               ast_dsp_free(p->dsp);
               p->dsp = NULL;
            }
         } else {
            ast_debug(1, "* Detected inband DTMF '%c'\n", f->subclass.integer);
         }
      }
   }

   return f;
}

static const char* sip_sanitized_host

(

const char *

host

)

[static]

Definition at line 15189 of file chan_sip.c.

References ast_sockaddr_parse(), ast_sockaddr_stringify_host_remote(), and PARSE_PORT_FORBID.

Referenced by transmit_register().

{
   struct ast_sockaddr addr = { { 0, 0, }, };

   /* peer/sip_pvt->tohost and sip_registry->hostname should never have a port
    * in them, so we use PARSE_PORT_FORBID here. If this lookup fails, we return
    * the original host which is most likely a host name and not an IP. */
   if (!ast_sockaddr_parse(&addr, host, PARSE_PORT_FORBID)) {
      return host;
   }
   return ast_sockaddr_stringify_host_remote(&addr);
}

void sip_scheddestroy	(	struct sip_pvt *	p,
		int	ms
	)

Schedule destruction of SIP dialog.

Definition at line 4456 of file chan_sip.c.

References __sip_autodestruct(), append_history, ast_log(), ast_sched_add(), ast_verbose(), global_t1, global_timer_b, LOG_WARNING, sip_cancel_destroy(), sip_debug_test_pvt(), sip_methods, stop_session_timer(), cfsip_methods::text, and TRUE.

Referenced by __sip_autodestruct(), auto_congest(), check_auth(), check_pendings(), extensionstate_update(), handle_incoming(), handle_invite_replaces(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_register(), handle_request_subscribe(), handle_response_invite(), manager_sipnotify(), receive_message(), sip_cli_notify(), sip_hangup(), sip_msg_send(), sip_scheddestroy_final(), sip_send_mwi_to_peer(), sip_sipredirect(), transmit_fake_auth_response(), and transmit_publish().

{
   if (p->final_destruction_scheduled) {
      return; /* already set final destruction */
   }

   if (ms < 0) {
      if (p->timer_t1 == 0) {
         p->timer_t1 = global_t1;   /* Set timer T1 if not set (RFC 3261) */
      }
      if (p->timer_b == 0) {
         p->timer_b = global_timer_b;  /* Set timer B if not set (RFC 3261) */
      }
      ms = p->timer_t1 * 64;
   }
   if (sip_debug_test_pvt(p)) {
      ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
   }
   if (sip_cancel_destroy(p)) {
      ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");
   }

   if (p->do_history) {
      append_history(p, "SchedDestroy", "%d ms", ms);
   }
   p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, dialog_ref(p, "setting ref as passing into ast_sched_add for __sip_autodestruct"));

   if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > 0) {
      stop_session_timer(p);
   }
}

void sip_scheddestroy_final	(	struct sip_pvt *	p,
		int	ms
	)

Schedule final destruction of SIP dialog. This can not be canceled. This function is used to keep a dialog around for a period of time in order to properly respond to any retransmits.

Definition at line 4443 of file chan_sip.c.

References sip_scheddestroy().

Referenced by handle_request_bye().

{
   if (p->final_destruction_scheduled) {
      return; /* already set final destruction */
   }

   sip_scheddestroy(p, ms);
   if (p->autokillid != -1) {
      p->final_destruction_scheduled = 1;
   }
}

static void sip_send_all_mwi_subscriptions

(

void

)

[static]

Send all MWI subscriptions.

Definition at line 33274 of file chan_sip.c.

References ast_sched_add(), AST_SCHED_DEL, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_REF, ASTOBJ_UNLOCK, ASTOBJ_UNREF, ASTOBJ_WRLOCK, sip_subscribe_mwi_destroy(), sip_subscribe_mwi_do(), and submwil.

Referenced by load_module(), network_change_event_sched_cb(), and sip_do_reload().

{
   ASTOBJ_CONTAINER_TRAVERSE(&submwil, 1, do {
      struct sip_subscription_mwi *saved;
      ASTOBJ_WRLOCK(iterator);
      AST_SCHED_DEL(sched, iterator->resub);
      saved = ASTOBJ_REF(iterator);
      if ((iterator->resub = ast_sched_add(sched, 1, sip_subscribe_mwi_do, saved)) < 0) {
         ASTOBJ_UNREF(saved, sip_subscribe_mwi_destroy);
      }
      ASTOBJ_UNLOCK(iterator);
   } while (0));
}

static void sip_send_all_registers

(

void

)

[static]

Send all known registrations.

Definition at line 33249 of file chan_sip.c.

References AST_SCHED_REPLACE_UNREF, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_UNLOCK, ASTOBJ_WRLOCK, registry_addref(), registry_unref(), regl, regobjs, and sip_reregister().

Referenced by load_module(), network_change_event_sched_cb(), and sip_do_reload().

{
   int ms;
   int regspacing;
   if (!regobjs) {
      return;
   }
   regspacing = default_expiry * 1000/regobjs;
   if (regspacing > 100) {
      regspacing = 100;
   }
   ms = regspacing;
   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_WRLOCK(iterator);
      ms += regspacing;
      AST_SCHED_REPLACE_UNREF(iterator->expire, sched, ms, sip_reregister, iterator,
                        registry_unref(_data, "REPLACE sched del decs the refcount"),
                        registry_unref(iterator, "REPLACE sched add failure decs the refcount"),
                        registry_addref(iterator, "REPLACE sched add incs the refcount"));
      ASTOBJ_UNLOCK(iterator);
   } while (0)
   );
}

static int sip_send_keepalive

(

const void *

data

)

[static]

Send keep alive packet to peer.

Definition at line 29510 of file chan_sip.c.

References ast_log(), AST_SCHED_REPLACE_UNREF, ast_sendto(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), errno, keepalive, LOG_WARNING, sip_ref_peer(), sip_tcptls_write(), and sip_unref_peer().

Referenced by sip_keepalive_all_peers().

{
   struct sip_peer *peer = (struct sip_peer*) data;
   int res = 0;
   const char keepalive[] = "\r\n";

   peer->keepalivesend = -1;

   if (!peer->keepalive || ast_sockaddr_isnull(&peer->addr)) {
      sip_unref_peer(peer, "release keepalive peer ref");
      return 0;
   }

   /* Send the packet out using the proper method for this peer */
   if ((peer->socket.fd != -1) && (peer->socket.type == SIP_TRANSPORT_UDP)) {
      res = ast_sendto(peer->socket.fd, keepalive, sizeof(keepalive), 0, &peer->addr);
   } else if ((peer->socket.type & (SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS)) &&
         (peer->socket.tcptls_session) &&
         (peer->socket.tcptls_session->fd != -1)) {
      res = sip_tcptls_write(peer->socket.tcptls_session, keepalive, sizeof(keepalive));
   } else if (peer->socket.type == SIP_TRANSPORT_UDP) {
      res = ast_sendto(sipsock, keepalive, sizeof(keepalive), 0, &peer->addr);
   }

   if (res == -1) {
      switch (errno) {
      case EBADF:             /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
      case EHOSTUNREACH:      /* Host can't be reached */
      case ENETDOWN:          /* Interface down */
      case ENETUNREACH:       /* Network failure */
      case ECONNREFUSED:      /* ICMP port unreachable */
         res = XMIT_ERROR;       /* Don't bother with trying to transmit again */
      }
   }

   if (res != sizeof(keepalive)) {
      ast_log(LOG_WARNING, "sip_send_keepalive to %s returned %d: %s\n", ast_sockaddr_stringify(&peer->addr), res, strerror(errno));
   }

   AST_SCHED_REPLACE_UNREF(peer->keepalivesend, sched,
            peer->keepalive * 1000, sip_send_keepalive, peer,
            sip_unref_peer(_data, "removing keepalive peer ref"),
            sip_unref_peer(peer, "removing keepalive peer ref"),
            sip_ref_peer(peer, "adding keepalive peer ref"));

   sip_unref_peer(peer, "release keepalive peer ref");

   return 0;
}

static int sip_send_mwi_to_peer	(	struct sip_peer *	peer,
		int	cache_only
	)		[static]

Send message waiting indication to alert peer that they've got voicemail.

Note:

Both peer and associated sip_pvt must be unlocked prior to calling this function

Returns:

-1 on failure, 0 on success

Definition at line 28846 of file chan_sip.c.

References ao2_lock, ao2_unlock, ast_app_inboxcount(), ast_set_flag, ast_sip_ouraddrfor(), ast_sockaddr_isnull(), ast_str_alloca, ast_str_buffer(), ast_str_strlen(), ast_string_field_set, ast_strlen_zero(), ast_test_flag, build_via(), change_callid_pvt(), create_addr_from_peer(), dialog_unlink_all(), get_cached_mwi(), peer_mailboxes_to_str(), set_socket_transport(), sip_alloc(), sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), transmit_notify_with_mwi(), update_peer_lastmsgssent(), and vmexten.

Referenced by build_peer(), handle_request_subscribe(), mwi_event_cb(), and register_verify().

{
   /* Called with peer lock, but releases it */
   struct sip_pvt *p;
   int newmsgs = 0, oldmsgs = 0;
   const char *vmexten = NULL;

   ao2_lock(peer);

   if (peer->vmexten) {
      vmexten = ast_strdupa(peer->vmexten);
   }

   if (ast_test_flag((&peer->flags[1]), SIP_PAGE2_SUBSCRIBEMWIONLY) && !peer->mwipvt) {
      update_peer_lastmsgssent(peer, -1, 1);
      ao2_unlock(peer);
      return -1;
   }

   /* Do we have an IP address? If not, skip this peer */
   if (ast_sockaddr_isnull(&peer->addr) && ast_sockaddr_isnull(&peer->defaddr)) {
      update_peer_lastmsgssent(peer, -1, 1);
      ao2_unlock(peer);
      return -1;
   }

   /* Attempt to use cached mwi to get message counts. */
   if (!get_cached_mwi(peer, &newmsgs, &oldmsgs) && !cache_only) {
      /* Fall back to manually checking the mailbox if not cache_only and get_cached_mwi failed */
      struct ast_str *mailbox_str = ast_str_alloca(512);
      peer_mailboxes_to_str(&mailbox_str, peer);
      /* if there is no mailbox do nothing */
      if (!ast_str_strlen(mailbox_str)) {
         ao2_unlock(peer);
         return -1;
      }
      ao2_unlock(peer);
      /* If there is no mailbox do nothing */
      if (!ast_str_strlen(mailbox_str)) {
         update_peer_lastmsgssent(peer, -1, 0);
         return 0;
      }
      ast_app_inboxcount(ast_str_buffer(mailbox_str), &newmsgs, &oldmsgs);
      ao2_lock(peer);
   }

   if (peer->mwipvt) {
      /* Base message on subscription */
      p = dialog_ref(peer->mwipvt, "sip_send_mwi_to_peer: Setting dialog ptr p from peer->mwipvt");
      ao2_unlock(peer);
   } else {
      ao2_unlock(peer);
      /* Build temporary dialog for this message */
      if (!(p = sip_alloc(NULL, NULL, 0, SIP_NOTIFY, NULL, NULL))) {
         update_peer_lastmsgssent(peer, -1, 0);
         return -1;
      }

      /* If we don't set the socket type to 0, then create_addr_from_peer will fail immediately if the peer
       * uses any transport other than UDP. We set the type to 0 here and then let create_addr_from_peer copy
       * the peer's socket information to the sip_pvt we just allocated
       */
      set_socket_transport(&p->socket, 0);
      if (create_addr_from_peer(p, peer)) {
         /* Maybe they're not registered, etc. */
         dialog_unlink_all(p);
         dialog_unref(p, "unref dialog p just created via sip_alloc");
         update_peer_lastmsgssent(peer, -1, 0);
         return -1;
      }
      /* Recalculate our side, and recalculate Call ID */
      ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
      build_via(p);

      ao2_lock(peer);
      if (!ast_strlen_zero(peer->mwi_from)) {
         ast_string_field_set(p, mwi_from, peer->mwi_from);
      } else if (!ast_strlen_zero(default_mwi_from)) {
         ast_string_field_set(p, mwi_from, default_mwi_from);
      }
      ao2_unlock(peer);

      /* Change the dialog callid. */
      change_callid_pvt(p, NULL);

      /* Destroy this session after 32 secs */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   }

   /* We have multiple threads (mwi events and monitor retransmits) working with this PVT and as we modify the sip history if that's turned on,
      we really need to have a lock on it */
   sip_pvt_lock(p);

   /* Send MWI */
   ast_set_flag(&p->flags[0], SIP_OUTGOING);
   /* the following will decrement the refcount on p as it finishes */
   transmit_notify_with_mwi(p, newmsgs, oldmsgs, vmexten);
   sip_pvt_unlock(p);
   dialog_unref(p, "unref dialog ptr p just before it goes out of scope at the end of sip_send_mwi_to_peer.");

   update_peer_lastmsgssent(peer, ((newmsgs > 0x7fff ? 0x7fff0000 : (newmsgs << 16)) | (oldmsgs > 0xffff ? 0xffff : oldmsgs)), 0);

   return 0;
}

static int sip_senddigit_begin	(	struct ast_channel *	ast,
		char	digit
	)		[static]

Definition at line 7441 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_rtp_instance_dtmf_begin(), ast_test_flag, sip_pvt_lock, and sip_pvt_unlock.

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int res = 0;

   if (!p) {
      ast_debug(1, "Asked to begin DTMF digit on channel %s with no pvt; ignoring\n",
            ast_channel_name(ast));
      return res;
   }

   sip_pvt_lock(p);
   switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
   case SIP_DTMF_INBAND:
      res = -1; /* Tell Asterisk to generate inband indications */
      break;
   case SIP_DTMF_RFC2833:
      if (p->rtp)
         ast_rtp_instance_dtmf_begin(p->rtp, digit);
      break;
   default:
      break;
   }
   sip_pvt_unlock(p);

   return res;
}

static int sip_senddigit_end	(	struct ast_channel *	ast,
		char	digit,
		unsigned int	duration
	)		[static]

Send DTMF character on SIP channel within one call, we're able to transmit in many methods simultaneously.

Definition at line 7471 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_rtp_instance_dtmf_end_with_duration(), ast_test_flag, sip_pvt_lock, sip_pvt_unlock, and transmit_info_with_digit().

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int res = 0;

   if (!p) {
      ast_debug(1, "Asked to end DTMF digit on channel %s with no pvt; ignoring\n",
            ast_channel_name(ast));
      return res;
   }

   sip_pvt_lock(p);
   switch (ast_test_flag(&p->flags[0], SIP_DTMF)) {
   case SIP_DTMF_INFO:
   case SIP_DTMF_SHORTINFO:
      transmit_info_with_digit(p, digit, duration);
      break;
   case SIP_DTMF_RFC2833:
      if (p->rtp)
         ast_rtp_instance_dtmf_end_with_duration(p->rtp, digit, duration);
      break;
   case SIP_DTMF_INBAND:
      res = -1; /* Tell Asterisk to stop inband indications */
      break;
   }
   sip_pvt_unlock(p);

   return res;
}

static int sip_sendhtml	(	struct ast_channel *	chan,
		int	subclass,
		const char *	data,
		int	datalen
	)		[static]

Send message with Access-URL header, if this is an HTML URL only!

Definition at line 4996 of file chan_sip.c.

References ast_channel_tech_pvt(), ast_debug, AST_HTML_URL, ast_log(), ast_set_flag, AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_string_field_build, ast_test_flag, FALSE, LOG_WARNING, sip_debug_test_pvt(), transmit_reinvite_with_sdp(), transmit_response(), and url.

{
   struct sip_pvt *p = ast_channel_tech_pvt(chan);

   if (subclass != AST_HTML_URL)
      return -1;

   ast_string_field_build(p, url, "<%s>;mode=active", data);

   if (sip_debug_test_pvt(p))
      ast_debug(1, "Send URL %s, state = %d!\n", data, ast_channel_state(chan));

   switch (ast_channel_state(chan)) {
   case AST_STATE_RING:
      transmit_response(p, "100 Trying", &p->initreq);
      break;
   case AST_STATE_RINGING:
      transmit_response(p, "180 Ringing", &p->initreq);
      break;
   case AST_STATE_UP:
      if (!p->pendinginvite) {      /* We are up, and have no outstanding invite */
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
      break;
   default:
      ast_log(LOG_WARNING, "Don't know how to send URI when state is %d!\n", ast_channel_state(chan));
   }

   return 0;
}

static int sip_sendtext	(	struct ast_channel *	ast,
		const char *	text
	)		[static]

Definition at line 5040 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_string_field_set, ast_verbose(), debug, destroy_msg_headers(), is_method_allowed(), sip_debug_test_pvt(), sip_pvt_lock, sip_pvt_unlock, and transmit_message().

{
   struct sip_pvt *dialog = ast_channel_tech_pvt(ast);
   int debug;

   if (!dialog) {
      return -1;
   }
   /* NOT ast_strlen_zero, because a zero-length message is specifically
    * allowed by RFC 3428 (See section 10, Examples) */
   if (!text) {
      return 0;
   }
   if(!is_method_allowed(&dialog->allowed_methods, SIP_MESSAGE)) {
      ast_debug(2, "Trying to send MESSAGE to device that does not support it.\n");
      return 0;
   }

   debug = sip_debug_test_pvt(dialog);
   if (debug) {
      ast_verbose("Sending text %s on %s\n", text, ast_channel_name(ast));
   }

   /* Setup to send text message */
   sip_pvt_lock(dialog);
   destroy_msg_headers(dialog);
   ast_string_field_set(dialog, msg_body, text);
   transmit_message(dialog, 0, 0);
   sip_pvt_unlock(dialog);
   return 0;
}

static void sip_set_default_format_capabilities

(

struct ast_format_cap *

cap

)

[static]

Definition at line 31365 of file chan_sip.c.

References AST_FORMAT_ALAW, ast_format_cap_add(), ast_format_cap_remove_all(), AST_FORMAT_GSM, AST_FORMAT_H263, ast_format_set(), AST_FORMAT_TESTLAW, and AST_FORMAT_ULAW.

Referenced by reload_config().

{
   struct ast_format tmp_fmt;

   ast_format_cap_remove_all(cap);
   ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ULAW, 0));
   ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_TESTLAW, 0));
   ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_ALAW, 0));
   ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_GSM, 0));
   ast_format_cap_add(cap, ast_format_set(&tmp_fmt, AST_FORMAT_H263, 0));
}

static char * sip_set_history	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Enable/Disable SIP History logging (CLI)

Definition at line 21763 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, TRUE, and ast_cli_entry::usage.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip set history {on|off}";
      e->usage =
         "Usage: sip set history {on|off}\n"
         "       Enables/Disables recording of SIP dialog history for debugging purposes.\n"
         "       Use 'sip show history' to view the history of a call number.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != e->args)
      return CLI_SHOWUSAGE;

   if (!strncasecmp(a->argv[e->args - 1], "on", 2)) {
      recordhistory = TRUE;
      ast_cli(a->fd, "SIP History Recording Enabled (use 'sip show history')\n");
   } else if (!strncasecmp(a->argv[e->args - 1], "off", 3)) {
      recordhistory = FALSE;
      ast_cli(a->fd, "SIP History Recording Disabled\n");
   } else {
      return CLI_SHOWUSAGE;
   }
   return CLI_SUCCESS;
}

static void sip_set_redirstr	(	struct sip_pvt *	p,
		char *	reason
	)		[static]

Translate referring cause.

Definition at line 17192 of file chan_sip.c.

References ast_string_field_set.

Referenced by get_rdnis().

                                                              {

   if (!strcmp(reason, "unknown")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "user-busy")) {
      ast_string_field_set(p, redircause, "BUSY");
   } else if (!strcmp(reason, "no-answer")) {
      ast_string_field_set(p, redircause, "NOANSWER");
   } else if (!strcmp(reason, "unavailable")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else if (!strcmp(reason, "unconditional")) {
      ast_string_field_set(p, redircause, "UNCONDITIONAL");
   } else if (!strcmp(reason, "time-of-day")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "do-not-disturb")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "deflection")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "follow-me")) {
      ast_string_field_set(p, redircause, "UNKNOWN");
   } else if (!strcmp(reason, "out-of-service")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else if (!strcmp(reason, "away")) {
      ast_string_field_set(p, redircause, "UNREACHABLE");
   } else {
      ast_string_field_set(p, redircause, "UNKNOWN");
   }
}

static int sip_set_rtp_peer	(	struct ast_channel *	chan,
		struct ast_rtp_instance *	instance,
		struct ast_rtp_instance *	vinstance,
		struct ast_rtp_instance *	tinstance,
		const struct ast_format_cap *	cap,
		int	nat_active
	)		[static]

Definition at line 32817 of file chan_sip.c.

References append_history, ast_bridged_channel(), ast_channel_lock, ast_channel_name(), ast_channel_set_fd(), ast_channel_tech_pvt(), ast_channel_unlock, ast_clear_flag, ast_debug, ast_format_cap_copy(), ast_format_cap_identical(), ast_format_cap_is_empty(), ast_rtp_instance_fd(), ast_rtp_instance_get_and_cmp_remote_address(), ast_rtp_instance_set_prop(), AST_RTP_PROPERTY_RTCP, ast_set_flag, ast_sockaddr_isnull(), ast_sockaddr_stringify(), AST_STATE_UP, ast_test_flag, FALSE, sip_cfg, sip_pvt_lock, sip_pvt_unlock, and transmit_reinvite_with_sdp().

Referenced by sip_fixup().

{
   struct sip_pvt *p;
   int changed = 0;

   /* Lock the channel and the private safely. */
   ast_channel_lock(chan);
   p = ast_channel_tech_pvt(chan);
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }
   sip_pvt_lock(p);
   if (p->owner != chan) {
      /* I suppose it could be argued that if this happens it is a bug. */
      ast_debug(1, "The private is not owned by channel %s anymore.\n", ast_channel_name(chan));
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   /* Disable early RTP bridge  */
   if ((instance || vinstance || tinstance) &&
      !ast_bridged_channel(chan) &&
      !sip_cfg.directrtpsetup) {
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   if (p->alreadygone) {
      /* If we're destroyed, don't bother */
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   /* if this peer cannot handle reinvites of the media stream to devices
      that are known to be behind a NAT, then stop the process now
   */
   if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   if (instance) {
      changed |= ast_rtp_instance_get_and_cmp_remote_address(instance, &p->redirip);

      if (p->rtp) {
         /* Prevent audio RTCP reads */
         ast_channel_set_fd(chan, 1, -1);
         /* Silence RTCP while audio RTP is inactive */
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 0);
      }
   } else if (!ast_sockaddr_isnull(&p->redirip)) {
      memset(&p->redirip, 0, sizeof(p->redirip));
      changed = 1;

      if (p->rtp) {
         /* Enable RTCP since it will be inactive if we're coming back
          * from a reinvite */
         ast_rtp_instance_set_prop(p->rtp, AST_RTP_PROPERTY_RTCP, 1);
         /* Enable audio RTCP reads */
         ast_channel_set_fd(chan, 1, ast_rtp_instance_fd(p->rtp, 1));
      }
   }

   if (vinstance) {
      changed |= ast_rtp_instance_get_and_cmp_remote_address(vinstance, &p->vredirip);

      if (p->vrtp) {
         /* Prevent video RTCP reads */
         ast_channel_set_fd(chan, 3, -1);
         /* Silence RTCP while video RTP is inactive */
         ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 0);
      }
   } else if (!ast_sockaddr_isnull(&p->vredirip)) {
      memset(&p->vredirip, 0, sizeof(p->vredirip));
      changed = 1;

      if (p->vrtp) {
         /* Enable RTCP since it will be inactive if we're coming back
          * from a reinvite */
         ast_rtp_instance_set_prop(p->vrtp, AST_RTP_PROPERTY_RTCP, 1);
         /* Enable video RTCP reads */
         ast_channel_set_fd(chan, 3, ast_rtp_instance_fd(p->vrtp, 1));
      }
   }

   if (tinstance) {
      changed |= ast_rtp_instance_get_and_cmp_remote_address(tinstance, &p->tredirip);
   } else if (!ast_sockaddr_isnull(&p->tredirip)) {
      memset(&p->tredirip, 0, sizeof(p->tredirip));
      changed = 1;
   }
   if (cap && !ast_format_cap_is_empty(cap) && !ast_format_cap_identical(p->redircaps, cap)) {
      ast_format_cap_copy(p->redircaps, cap);
      changed = 1;
   }

   if (ast_test_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING) && !p->outgoing_call) {
      /* We only wish to withhold sending the initial direct media reinvite on the incoming dialog.
       * Further direct media reinvites beyond the initial should be sent. In order to allow further
       * direct media reinvites to be sent, we clear this flag.
       */
      ast_clear_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER) && !ast_test_flag(&p->flags[0], SIP_DEFER_BYE_ON_TRANSFER)) {
      if (ast_channel_state(chan) != AST_STATE_UP) {     /* We are in early state */
         if (p->do_history)
            append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
         ast_debug(1, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
      } else if (!p->pendinginvite) {   /* We are up, and have no outstanding invite */
         ast_debug(3, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
         transmit_reinvite_with_sdp(p, FALSE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_debug(3, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_sockaddr_stringify(instance ? &p->redirip : &p->ourip));
         /* We have a pending Invite. Send re-invite when we're done with the invite */
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   }
   /* Reset lastrtprx timer */
   p->lastrtprx = p->lastrtptx = time(NULL);
   sip_pvt_unlock(p);
   ast_channel_unlock(chan);
   return 0;
}

static int sip_set_udptl_peer	(	struct ast_channel *	chan,
		struct ast_udptl *	udptl
	)		[static]

Definition at line 32617 of file chan_sip.c.

References ast_channel_lock, ast_channel_name(), ast_channel_tech_pvt(), ast_channel_unlock, ast_debug, ast_set_flag, ast_sockaddr_stringify(), ast_test_flag, ast_udptl_get_peer(), FALSE, sip_pvt_lock, sip_pvt_unlock, transmit_reinvite_with_sdp(), and TRUE.

{
   struct sip_pvt *p;

   /* Lock the channel and the private safely. */
   ast_channel_lock(chan);
   p = ast_channel_tech_pvt(chan);
   if (!p) {
      ast_channel_unlock(chan);
      return -1;
   }
   sip_pvt_lock(p);
   if (p->owner != chan) {
      /* I suppose it could be argued that if this happens it is a bug. */
      ast_debug(1, "The private is not owned by channel %s anymore.\n", ast_channel_name(chan));
      sip_pvt_unlock(p);
      ast_channel_unlock(chan);
      return 0;
   }

   if (udptl) {
      ast_udptl_get_peer(udptl, &p->udptlredirip);
   } else {
      memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
   }
   if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
      if (!p->pendinginvite) {
         ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s\n",
               p->callid, ast_sockaddr_stringify(udptl ? &p->udptlredirip : &p->ourip));
         transmit_reinvite_with_sdp(p, TRUE, FALSE);
      } else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
         ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s\n",
               p->callid, ast_sockaddr_stringify(udptl ? &p->udptlredirip : &p->ourip));
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   }
   /* Reset lastrtprx timer */
   p->lastrtprx = p->lastrtptx = time(NULL);
   sip_pvt_unlock(p);
   ast_channel_unlock(chan);
   return 0;
}

static int sip_setoption	(	struct ast_channel *	chan,
		int	option,
		void *	data,
		int	datalen
	)		[static]

Set an option on a SIP dialog.

Definition at line 4849 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, ast_log(), AST_OPTION_DIGIT_DETECT, AST_OPTION_FORMAT_READ, AST_OPTION_FORMAT_WRITE, AST_OPTION_MAKE_COMPATIBLE, AST_OPTION_SECURE_MEDIA, AST_OPTION_SECURE_SIGNALING, ast_rtp_instance_make_compatible(), ast_rtp_instance_set_read_format(), ast_rtp_instance_set_write_format(), ast_set2_flag, ast_test_flag, disable_dsp_detect(), enable_dsp_detect(), LOG_ERROR, sip_pvt_lock, and sip_pvt_unlock.

{
   int res = -1;
   struct sip_pvt *p = ast_channel_tech_pvt(chan);

        if (!p) {
      ast_log(LOG_ERROR, "Attempt to Ref a null pointer.  sip private structure is gone!\n");
      return -1;
        }

   sip_pvt_lock(p);

   switch (option) {
   case AST_OPTION_FORMAT_READ:
      if (p->rtp) {
         res = ast_rtp_instance_set_read_format(p->rtp, (struct ast_format *) data);
      }
      break;
   case AST_OPTION_FORMAT_WRITE:
      if (p->rtp) {
         res = ast_rtp_instance_set_write_format(p->rtp, (struct ast_format *) data);
      }
      break;
   case AST_OPTION_MAKE_COMPATIBLE:
      if (p->rtp) {
         res = ast_rtp_instance_make_compatible(chan, p->rtp, (struct ast_channel *) data);
      }
      break;
   case AST_OPTION_DIGIT_DETECT:
      if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) ||
          (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO)) {
         char *cp = (char *) data;

         ast_debug(1, "%sabling digit detection on %s\n", *cp ? "En" : "Dis", ast_channel_name(chan));
         if (*cp) {
            enable_dsp_detect(p);
         } else {
            disable_dsp_detect(p);
         }
         res = 0;
      }
      break;
   case AST_OPTION_SECURE_SIGNALING:
      p->req_secure_signaling = *(unsigned int *) data;
      res = 0;
      break;
   case AST_OPTION_SECURE_MEDIA:
      ast_set2_flag(&p->flags[1], *(unsigned int *) data, SIP_PAGE2_USE_SRTP);
      res = 0;
      break;
   default:
      break;
   }

   sip_pvt_unlock(p);

   return res;
}

static char * sip_show_channel	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show details of one active dialog.

Definition at line 21179 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_cli_args::argc, ast_cli_args::argv, ARRAY_LEN, ast_channel_name(), ast_channel_nativeformats(), ast_cli(), AST_CLI_YESNO, ast_getformatname_multiple(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_strlen_zero(), ast_test_flag, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sipch(), dtmfmode2str(), ast_cli_args::fd, first, force_rport_string(), len(), ast_cli_args::line, ast_cli_args::n, NONE, ast_cli_args::pos, sip_pvt_lock, sip_pvt_unlock, stmode2str(), strefresher2str(), strefresherparam2str(), subscription_type2str(), transfermode2str(), TRUE, ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_pvt *cur;
   size_t len;
   int found = 0;
   struct ao2_iterator i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show channel";
      e->usage =
         "Usage: sip show channel <call-id>\n"
         "       Provides detailed status on a given SIP dialog (identified by SIP call-id).\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sipch(a->line, a->word, a->pos, a->n);
   }

   if (a->argc != 4)
      return CLI_SHOWUSAGE;
   len = strlen(a->argv[3]);
   
   i = ao2_iterator_init(dialogs, 0);
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);

      if (!strncasecmp(cur->callid, a->argv[3], len)) {
         char formatbuf[SIPBUFSIZE/2];
         ast_cli(a->fd, "\n");
         if (cur->subscribed != NONE) {
            ast_cli(a->fd, "  * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
         } else {
            ast_cli(a->fd, "  * SIP Call\n");
         }
         ast_cli(a->fd, "  Curr. trans. direction:  %s\n", ast_test_flag(&cur->flags[0], SIP_OUTGOING) ? "Outgoing" : "Incoming");
         ast_cli(a->fd, "  Call-ID:                %s\n", cur->callid);
         ast_cli(a->fd, "  Owner channel ID:       %s\n", cur->owner ? ast_channel_name(cur->owner) : "<none>");
         ast_cli(a->fd, "  Our Codec Capability:   %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->caps));
         ast_cli(a->fd, "  Non-Codec Capability (DTMF):   %d\n", cur->noncodeccapability);
         ast_cli(a->fd, "  Their Codec Capability:   %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->peercaps));
         ast_cli(a->fd, "  Joint Codec Capability:   %s\n", ast_getformatname_multiple(formatbuf, sizeof(formatbuf), cur->jointcaps));
         ast_cli(a->fd, "  Format:                 %s\n", cur->owner ? ast_getformatname_multiple(formatbuf, sizeof(formatbuf), ast_channel_nativeformats(cur->owner)) : "(nothing)" );
         ast_cli(a->fd, "  T.38 support            %s\n", AST_CLI_YESNO(cur->udptl != NULL));
         ast_cli(a->fd, "  Video support           %s\n", AST_CLI_YESNO(cur->vrtp != NULL));
         ast_cli(a->fd, "  MaxCallBR:              %d kbps\n", cur->maxcallbitrate);
         ast_cli(a->fd, "  Theoretical Address:    %s\n", ast_sockaddr_stringify(&cur->sa));
         ast_cli(a->fd, "  Received Address:       %s\n", ast_sockaddr_stringify(&cur->recv));
         ast_cli(a->fd, "  SIP Transfer mode:      %s\n", transfermode2str(cur->allowtransfer));
         ast_cli(a->fd, "  Force rport:            %s\n", force_rport_string(cur->flags));
         if (ast_sockaddr_isnull(&cur->redirip)) {
            ast_cli(a->fd,
               "  Audio IP:               %s (local)\n",
               ast_sockaddr_stringify_addr(&cur->ourip));
         } else {
            ast_cli(a->fd,
               "  Audio IP:               %s (Outside bridge)\n",
               ast_sockaddr_stringify_addr(&cur->redirip));
         }
         ast_cli(a->fd, "  Our Tag:                %s\n", cur->tag);
         ast_cli(a->fd, "  Their Tag:              %s\n", cur->theirtag);
         ast_cli(a->fd, "  SIP User agent:         %s\n", cur->useragent);
         if (!ast_strlen_zero(cur->username)) {
            ast_cli(a->fd, "  Username:               %s\n", cur->username);
         }
         if (!ast_strlen_zero(cur->peername)) {
            ast_cli(a->fd, "  Peername:               %s\n", cur->peername);
         }
         if (!ast_strlen_zero(cur->uri)) {
            ast_cli(a->fd, "  Original uri:           %s\n", cur->uri);
         }
         if (!ast_strlen_zero(cur->cid_num)) {
            ast_cli(a->fd, "  Caller-ID:              %s\n", cur->cid_num);
         }
         ast_cli(a->fd, "  Need Destroy:           %s\n", AST_CLI_YESNO(cur->needdestroy));
         ast_cli(a->fd, "  Last Message:           %s\n", cur->lastmsg);
         ast_cli(a->fd, "  Promiscuous Redir:      %s\n", AST_CLI_YESNO(ast_test_flag(&cur->flags[0], SIP_PROMISCREDIR)));
         ast_cli(a->fd, "  Route:                  ");
         if (cur->route) {
            struct sip_route *route;
            int first = 1;

            for (route = cur->route; route; route = route->next) {
               ast_cli(a->fd, "%s<%s>", first ? "" : ", ", route->hop);
               first = 0;
            }
            ast_cli(a->fd, "\n");
         } else {
            ast_cli(a->fd, "N/A\n");
         }
         ast_cli(a->fd, "  DTMF Mode:              %s\n", dtmfmode2str(ast_test_flag(&cur->flags[0], SIP_DTMF)));
         ast_cli(a->fd, "  SIP Options:            ");
         if (cur->sipoptions) {
            int x;
            for (x = 0 ; x < ARRAY_LEN(sip_options); x++) {
               if (cur->sipoptions & sip_options[x].id)
                  ast_cli(a->fd, "%s ", sip_options[x].text);
            }
            ast_cli(a->fd, "\n");
         } else {
            ast_cli(a->fd, "(none)\n");
         }

         if (!cur->stimer) {
            ast_cli(a->fd, "  Session-Timer:          Uninitiallized\n");
         } else {
            ast_cli(a->fd, "  Session-Timer:          %s\n", cur->stimer->st_active ? "Active" : "Inactive");
            if (cur->stimer->st_active == TRUE) {
               ast_cli(a->fd, "  S-Timer Interval:       %d\n", cur->stimer->st_interval);
               ast_cli(a->fd, "  S-Timer Refresher:      %s\n", strefresher2str(cur->stimer->st_ref));
               ast_cli(a->fd, "  S-Timer Sched Id:       %d\n", cur->stimer->st_schedid);
               ast_cli(a->fd, "  S-Timer Peer Sts:       %s\n", cur->stimer->st_active_peer_ua ? "Active" : "Inactive");
               ast_cli(a->fd, "  S-Timer Cached Min-SE:  %d\n", cur->stimer->st_cached_min_se);
               ast_cli(a->fd, "  S-Timer Cached SE:      %d\n", cur->stimer->st_cached_max_se);
               ast_cli(a->fd, "  S-Timer Cached Ref:     %s\n", strefresherparam2str(cur->stimer->st_cached_ref));
               ast_cli(a->fd, "  S-Timer Cached Mode:    %s\n", stmode2str(cur->stimer->st_cached_mode));
            }
         }

         ast_cli(a->fd, "\n\n");

         found++;
      }

      sip_pvt_unlock(cur);

      ao2_t_ref(cur, -1, "toss dialog ptr set by iterator_next");
   }
   ao2_iterator_destroy(&i);

   if (!found) {
      ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
   }

   return CLI_SUCCESS;
}

static char* sip_show_channels	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

CLI for show channels or subscriptions. This is a new-style CLI handler so a single function contains the prototype for the function, the 'generator' to produce multiple entries in case it is required, and the actual handler for the command.

Definition at line 21000 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli_entry::args, ast_cli_args::argv, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ESS, ast_cli_args::fd, FORMAT2, FORMAT3, OBJ_NODATA, show_channels_cb(), and ast_cli_entry::usage.

{
   struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };


   if (cmd == CLI_INIT) {
      e->command = "sip show {channels|subscriptions}";
      e->usage =
         "Usage: sip show channels\n"
         "       Lists all currently active SIP calls (dialogs).\n"
         "Usage: sip show subscriptions\n"
         "       Lists active SIP subscriptions.\n";
      return NULL;
   } else if (cmd == CLI_GENERATE)
      return NULL;

   if (a->argc != e->args)
      return CLI_SHOWUSAGE;
   arg.subscriptions = !strcasecmp(a->argv[e->args - 1], "subscriptions");
   if (!arg.subscriptions)
      ast_cli(arg.fd, FORMAT2, "Peer", "User/ANR", "Call ID", "Format", "Hold", "Last Message", "Expiry", "Peer");
   else
      ast_cli(arg.fd, FORMAT3, "Peer", "User", "Call ID", "Extension", "Last state", "Type", "Mailbox", "Expiry");

   /* iterate on the container and invoke the callback on each item */
   ao2_t_callback(dialogs, OBJ_NODATA, show_channels_cb, &arg, "callback to show channels");
   
   /* print summary information */
   ast_cli(arg.fd, "%d active SIP %s%s\n", arg.numchans,
      (arg.subscriptions ? "subscription" : "dialog"),
      ESS(arg.numchans));  /* ESS(n) returns an "s" if n>1 */
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
#undef FORMAT3
}

static char * sip_show_channelstats	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

SIP show channelstats CLI (main function)

Definition at line 20612 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, FORMAT2, OBJ_NODATA, show_chanstats_cb(), and ast_cli_entry::usage.

{
   struct __show_chan_arg arg = { .fd = a->fd, .numchans = 0 };

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show channelstats";
      e->usage =
         "Usage: sip show channelstats\n"
         "       Lists all currently active SIP channel's RTCP statistics.\n"
         "       Note that calls in the much optimized RTP P2P bridge mode will not show any packets here.";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;

   ast_cli(a->fd, FORMAT2, "Peer", "Call ID", "Duration", "Recv: Pack", "Lost", "Jitter", "Send: Pack", "Lost", "Jitter");
   /* iterate on the container and invoke the callback on each item */
   ao2_t_callback(dialogs, OBJ_NODATA, show_chanstats_cb, &arg, "callback to sip show chanstats");
   ast_cli(a->fd, "%d active SIP channel%s\n", arg.numchans, (arg.numchans != 1) ? "s" : "");
   return CLI_SUCCESS;
}

static char * sip_show_domains	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

CLI command to list local domains.

Definition at line 19746 of file chan_sip.c.

References ast_cli(), AST_LIST_EMPTY, AST_LIST_LOCK, AST_LIST_TRAVERSE, AST_LIST_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, domain_mode_to_text(), ast_cli_args::fd, FORMAT, S_OR, and ast_cli_entry::usage.

{
   struct domain *d;
#define FORMAT "%-40.40s %-20.20s %-16.16s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show domains";
      e->usage =
         "Usage: sip show domains\n"
         "       Lists all configured SIP local domains.\n"
         "       Asterisk only responds to SIP messages to local domains.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (AST_LIST_EMPTY(&domain_list)) {
      ast_cli(a->fd, "SIP Domain support not enabled.\n\n");
      return CLI_SUCCESS;
   } else {
      ast_cli(a->fd, FORMAT, "Our local SIP domains:", "Context", "Set by");
      AST_LIST_LOCK(&domain_list);
      AST_LIST_TRAVERSE(&domain_list, d, list)
         ast_cli(a->fd, FORMAT, d->domain, S_OR(d->context, "(default)"),
            domain_mode_to_text(d->mode));
      AST_LIST_UNLOCK(&domain_list);
      ast_cli(a->fd, "\n");
      return CLI_SUCCESS;
   }
}

static char * sip_show_history	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show history details of one dialog.

Definition at line 21316 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), AST_LIST_TRAVERSE, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_show_history(), ast_cli_args::fd, len(), ast_cli_args::line, ast_cli_args::n, NONE, ast_cli_args::pos, sip_pvt_lock, sip_pvt_unlock, ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_pvt *cur;
   size_t len;
   int found = 0;
   struct ao2_iterator i;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show history";
      e->usage =
         "Usage: sip show history <call-id>\n"
         "       Provides detailed dialog history on a given SIP call (specified by call-id).\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_history(a->line, a->word, a->pos, a->n);
   }

   if (a->argc != 4) {
      return CLI_SHOWUSAGE;
   }

   if (!recordhistory) {
      ast_cli(a->fd, "\n***Note: History recording is currently DISABLED.  Use 'sip set history on' to ENABLE.\n");
   }

   len = strlen(a->argv[3]);

   i = ao2_iterator_init(dialogs, 0);
   while ((cur = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      sip_pvt_lock(cur);
      if (!strncasecmp(cur->callid, a->argv[3], len)) {
         struct sip_history *hist;
         int x = 0;

         ast_cli(a->fd, "\n");
         if (cur->subscribed != NONE) {
            ast_cli(a->fd, "  * Subscription\n");
         } else {
            ast_cli(a->fd, "  * SIP Call\n");
         }
         if (cur->history) {
            AST_LIST_TRAVERSE(cur->history, hist, list)
               ast_cli(a->fd, "%d. %s\n", ++x, hist->event);
         }
         if (x == 0) {
            ast_cli(a->fd, "Call '%s' has no history\n", cur->callid);
         }
         found++;
      }
      sip_pvt_unlock(cur);
      ao2_t_ref(cur, -1, "toss dialog ptr from iterator_next");
   }
   ao2_iterator_destroy(&i);

   if (!found) {
      ast_cli(a->fd, "No such SIP Call ID starting with '%s'\n", a->argv[3]);
   }

   return CLI_SUCCESS;
}

static char * sip_show_inuse	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

CLI Command to show calls within limits set by call_limit.

Definition at line 18751 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_cli_args::argc, ast_cli_args::argv, ast_cli(), ast_copy_string(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, FORMAT, FORMAT2, sip_unref_peer(), TRUE, and ast_cli_entry::usage.

{
#define FORMAT "%-25.25s %-15.15s %-15.15s \n"
#define FORMAT2 "%-25.25s %-15.15s %-15.15s \n"
   char ilimits[40];
   char iused[40];
   int showall = FALSE;
   struct ao2_iterator i;
   struct sip_peer *peer;
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show inuse";
      e->usage =
         "Usage: sip show inuse [all]\n"
         "       List all SIP devices usage counters and limits.\n"
         "       Add option \"all\" to show all devices, not only those with a limit.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc < 3)
      return CLI_SHOWUSAGE;

   if (a->argc == 4 && !strcmp(a->argv[3], "all"))
      showall = TRUE;
   
   ast_cli(a->fd, FORMAT, "* Peer name", "In use", "Limit");

   i = ao2_iterator_init(peers, 0);
   while ((peer = ao2_t_iterator_next(&i, "iterate thru peer table"))) {
      ao2_lock(peer);
      if (peer->call_limit)
         snprintf(ilimits, sizeof(ilimits), "%d", peer->call_limit);
      else
         ast_copy_string(ilimits, "N/A", sizeof(ilimits));
      snprintf(iused, sizeof(iused), "%d/%d/%d", peer->inuse, peer->ringing, peer->onhold);
      if (showall || peer->call_limit)
         ast_cli(a->fd, FORMAT2, peer->name, iused, ilimits);
      ao2_unlock(peer);
      sip_unref_peer(peer, "toss iterator pointer");
   }
   ao2_iterator_destroy(&i);

   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char * sip_show_mwi	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Definition at line 20879 of file chan_sip.c.

References ast_cli(), AST_CLI_YESNO, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, FORMAT, submwil, and ast_cli_entry::usage.

{
#define FORMAT  "%-30.30s  %-12.12s  %-10.10s  %-10.10s\n"
   char host[80];
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show mwi";
      e->usage =
         "Usage: sip show mwi\n"
         "       Provides a list of MWI subscriptions and status.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }
   
   ast_cli(a->fd, FORMAT, "Host", "Username", "Mailbox", "Subscribed");
   
   ASTOBJ_CONTAINER_TRAVERSE(&submwil, 1, do {
      ASTOBJ_RDLOCK(iterator);
      snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
      ast_cli(a->fd, FORMAT, host, iterator->username, iterator->mailbox, AST_CLI_YESNO(iterator->subscribed));
      ASTOBJ_UNLOCK(iterator);
   } while(0));

   return CLI_SUCCESS;
#undef FORMAT
}

static char * sip_show_objects	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

List all allocated SIP Objects (realtime or static)

Definition at line 19364 of file chan_sip.c.

References ao2_t_callback, ast_cli_args::argc, ast_cli(), ASTOBJ_CONTAINER_DUMP, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, dialog_dump_func(), ast_cli_args::fd, OBJ_NODATA, peer_dump_func(), regl, and ast_cli_entry::usage.

{
   char tmp[256];
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show objects";
      e->usage =
         "Usage: sip show objects\n"
         "       Lists status of known SIP objects\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }  

   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   ast_cli(a->fd, "-= Peer objects: %d static, %d realtime, %d autocreate =-\n\n", speerobjs, rpeerobjs, apeerobjs);
   ao2_t_callback(peers, OBJ_NODATA, peer_dump_func, a, "initiate ao2_callback to dump peers");
   ast_cli(a->fd, "-= Peer objects by IP =-\n\n"); 
   ao2_t_callback(peers_by_ip, OBJ_NODATA, peer_dump_func, a, "initiate ao2_callback to dump peers_by_ip");
   ast_cli(a->fd, "-= Registry objects: %d =-\n\n", regobjs);
   ASTOBJ_CONTAINER_DUMP(a->fd, tmp, sizeof(tmp), &regl);
   ast_cli(a->fd, "-= Dialog objects:\n\n");
   ao2_t_callback(dialogs, OBJ_NODATA, dialog_dump_func, a, "initiate ao2_callback to dump dialogs");
   return CLI_SUCCESS;
}

static char * sip_show_peer	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show one peer in detail.

Definition at line 19801 of file chan_sip.c.

References _sip_show_peer(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, complete_sip_show_peer(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, ast_cli_entry::usage, and ast_cli_args::word.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show peer";
      e->usage =
         "Usage: sip show peer <name> [load]\n"
         "       Shows all details on one SIP peer and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_peer(a->line, a->word, a->pos, a->n);
   }
   return _sip_show_peer(0, a->fd, NULL, NULL, a->argc, (const char **) a->argv);
}

static char * sip_show_peers	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

CLI Show Peers command.

Definition at line 19087 of file chan_sip.c.

References _sip_show_peers(), ast_cli_args::argc, ast_cli_args::argv, CLI_GENERATE, CLI_INIT, ast_cli_entry::command, ast_cli_args::fd, and ast_cli_entry::usage.

{
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show peers";
      e->usage =
         "Usage: sip show peers [like <pattern>]\n"
         "       Lists all known SIP peers.\n"
         "       Optional regular expression pattern is used to filter the peer list.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   return _sip_show_peers(a->fd, NULL, NULL, NULL, a->argc, (const char **) a->argv);
}

static char * sip_show_registry	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show SIP Registry (registrations with other SIP proxies.

Definition at line 20450 of file chan_sip.c.

References ast_cli_args::argc, ast_cli(), ast_localtime(), ast_strftime(), ast_strlen_zero(), ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_RDLOCK, ASTOBJ_UNLOCK, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, FORMAT, FORMAT2, regl, regstate2str(), and ast_cli_entry::usage.

{
#define FORMAT2 "%-39.39s %-6.6s %-12.12s  %8.8s %-20.20s %-25.25s\n"
#define FORMAT  "%-39.39s %-6.6s %-12.12s  %8d %-20.20s %-25.25s\n"
   char host[80];
   char user[80];
   char tmpdat[256];
   struct ast_tm tm;
   int counter = 0;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show registry";
      e->usage =
         "Usage: sip show registry\n"
         "       Lists all registration requests and status.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   ast_cli(a->fd, FORMAT2, "Host", "dnsmgr", "Username", "Refresh", "State", "Reg.Time");
   
   ASTOBJ_CONTAINER_TRAVERSE(&regl, 1, do {
      ASTOBJ_RDLOCK(iterator);
      snprintf(host, sizeof(host), "%s:%d", iterator->hostname, iterator->portno ? iterator->portno : STANDARD_SIP_PORT);
      snprintf(user, sizeof(user), "%s", iterator->username);
      if (!ast_strlen_zero(iterator->regdomain)) {
         snprintf(tmpdat, sizeof(tmpdat), "%s", user);
         snprintf(user, sizeof(user), "%s@%s", tmpdat, iterator->regdomain);}
      if (iterator->regdomainport) {
         snprintf(tmpdat, sizeof(tmpdat), "%s", user);
         snprintf(user, sizeof(user), "%s:%d", tmpdat, iterator->regdomainport);}
      if (iterator->regtime.tv_sec) {
         ast_localtime(&iterator->regtime, &tm, NULL);
         ast_strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T", &tm);
      } else
         tmpdat[0] = '\0';
      ast_cli(a->fd, FORMAT, host, (iterator->dnsmgr) ? "Y" : "N", user, iterator->refresh, regstate2str(iterator->regstate), tmpdat);
      ASTOBJ_UNLOCK(iterator);
      counter++;
   } while(0));
   ast_cli(a->fd, "%d SIP registrations.\n", counter);
   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char * sip_show_sched	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Definition at line 20407 of file chan_sip.c.

References __sip_autodestruct(), ast_cli(), ast_sched_report(), ast_str_alloca, ast_str_buffer(), auto_congest(), CLI_GENERATE, CLI_INIT, CLI_SUCCESS, ast_cli_entry::command, expire_register(), ast_cli_args::fd, retrans_pkt(), sip_poke_noanswer(), sip_poke_peer_s(), sip_reg_timeout(), sip_reinvite_retry(), sip_reregister(), and ast_cli_entry::usage.

{
   struct ast_str *cbuf;
   struct ast_cb_names cbnames = {9, { "retrans_pkt",
                                        "__sip_autodestruct",
                                        "expire_register",
                                        "auto_congest",
                                        "sip_reg_timeout",
                                        "sip_poke_peer_s",
                                        "sip_poke_noanswer",
                                        "sip_reregister",
                                        "sip_reinvite_retry"},
                           { retrans_pkt,
                                     __sip_autodestruct,
                                     expire_register,
                                     auto_congest,
                                     sip_reg_timeout,
                                     sip_poke_peer_s,
                                     sip_poke_noanswer,
                                     sip_reregister,
                                     sip_reinvite_retry}};
   
   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show sched";
      e->usage =
         "Usage: sip show sched\n"
         "       Shows stats on what's in the sched queue at the moment\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   cbuf = ast_str_alloca(2048);

   ast_cli(a->fd, "\n");
   ast_sched_report(sched, &cbuf, &cbnames);
   ast_cli(a->fd, "%s", ast_str_buffer(cbuf));

   return CLI_SUCCESS;
}

static char * sip_show_settings	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

List global settings for the SIP channel.

Definition at line 20641 of file chan_sip.c.

References ast_ha::addr, allowoverlap2str(), ao2_t_ref, ast_cli_args::argc, ast_check_realtime(), ast_cli(), AST_CLI_ONOFF, AST_CLI_YESNO, ast_getformatname_multiple(), AST_JB_ENABLED, AST_JB_FORCED, AST_JB_LOG, AST_LIST_EMPTY, AST_LIST_TRAVERSE, ast_mutex_lock, ast_mutex_unlock, ast_sockaddr_is_any(), ast_sockaddr_is_ipv6(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_sockaddr_stringify_addr(), ast_strlen_zero(), ast_test_flag, ast_tos2str(), authl, authl_lock, autocreatepeer2str(), bindaddr, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, comedia_string(), ast_cli_entry::command, default_prefs, dtmfmode2str(), ast_tls_config::enabled, externaddr, FALSE, faxec2str(), ast_cli_args::fd, force_rport_string(), get_transport_list(), global_jbconf, ast_jb_conf::impl, ast_tcptls_session_args::local_address, ast_jb_conf::max_size, ast_ha::netmask, ast_ha::next, prefix, print_codec_to_cli(), ast_jb_conf::resync_threshold, S_OR, sip_cfg, sip_get_transport(), stmode2str(), strefresherparam2str(), ast_jb_conf::target_extra, transfermode2str(), and ast_cli_entry::usage.

{
   int realtimepeers;
   int realtimeregs;
   char codec_buf[SIPBUFSIZE];
   const char *msg;  /* temporary msg pointer */
   struct sip_auth_container *credentials;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show settings";
      e->usage =
         "Usage: sip show settings\n"
         "       Provides detailed list of the configuration of the SIP channel.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;

   realtimepeers = ast_check_realtime("sippeers");
   realtimeregs = ast_check_realtime("sipregs");

   ast_mutex_lock(&authl_lock);
   credentials = authl;
   if (credentials) {
      ao2_t_ref(credentials, +1, "Ref global auth for show");
   }
   ast_mutex_unlock(&authl_lock);

   ast_cli(a->fd, "\n\nGlobal Settings:\n");
   ast_cli(a->fd, "----------------\n");
   ast_cli(a->fd, "  UDP Bindaddress:        %s\n", ast_sockaddr_stringify(&bindaddr));
   if (ast_sockaddr_is_ipv6(&bindaddr) && ast_sockaddr_is_any(&bindaddr)) {
      ast_cli(a->fd, "  ** Additional Info:\n");
      ast_cli(a->fd, "     [::] may include IPv4 in addition to IPv6, if such a feature is enabled in the OS.\n");
   }
   ast_cli(a->fd, "  TCP SIP Bindaddress:    %s\n",
      sip_cfg.tcp_enabled != FALSE ?
            ast_sockaddr_stringify(&sip_tcp_desc.local_address) :
            "Disabled");
   ast_cli(a->fd, "  TLS SIP Bindaddress:    %s\n",
      default_tls_cfg.enabled != FALSE ?
            ast_sockaddr_stringify(&sip_tls_desc.local_address) :
            "Disabled");
   ast_cli(a->fd, "  Videosupport:           %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_VIDEOSUPPORT)));
   ast_cli(a->fd, "  Textsupport:            %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_TEXTSUPPORT)));
   ast_cli(a->fd, "  Ignore SDP sess. ver.:  %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_IGNORESDPVERSION)));
   ast_cli(a->fd, "  AutoCreate Peer:        %s\n", autocreatepeer2str(sip_cfg.autocreatepeer));
   ast_cli(a->fd, "  Match Auth Username:    %s\n", AST_CLI_YESNO(global_match_auth_username));
   ast_cli(a->fd, "  Allow unknown access:   %s\n", AST_CLI_YESNO(sip_cfg.allowguest));
   ast_cli(a->fd, "  Allow subscriptions:    %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWSUBSCRIBE)));
   ast_cli(a->fd, "  Allow overlap dialing:  %s\n", allowoverlap2str(ast_test_flag(&global_flags[1], SIP_PAGE2_ALLOWOVERLAP)));
   ast_cli(a->fd, "  Allow promisc. redir:   %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROMISCREDIR)));
   ast_cli(a->fd, "  Enable call counters:   %s\n", AST_CLI_YESNO(global_callcounter));
   ast_cli(a->fd, "  SIP domain support:     %s\n", AST_CLI_YESNO(!AST_LIST_EMPTY(&domain_list)));
   ast_cli(a->fd, "  Realm. auth:            %s\n", AST_CLI_YESNO(credentials != NULL));
   if (credentials) {
      struct sip_auth *auth;

      AST_LIST_TRAVERSE(&credentials->list, auth, node) {
         ast_cli(a->fd, "  Realm. auth entry:      Realm %-15.15s User %-10.20s %s\n",
            auth->realm,
            auth->username,
            !ast_strlen_zero(auth->secret)
               ? "<Secret set>"
               : (!ast_strlen_zero(auth->md5secret)
                  ? "<MD5secret set>" : "<Not set>"));
      }
      ao2_t_ref(credentials, -1, "Unref global auth for show");
   }
   ast_cli(a->fd, "  Our auth realm          %s\n", sip_cfg.realm);
   ast_cli(a->fd, "  Use domains as realms:  %s\n", AST_CLI_YESNO(sip_cfg.domainsasrealm));
   ast_cli(a->fd, "  Call to non-local dom.: %s\n", AST_CLI_YESNO(sip_cfg.allow_external_domains));
   ast_cli(a->fd, "  URI user is phone no:   %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USEREQPHONE)));
   ast_cli(a->fd, "  Always auth rejects:    %s\n", AST_CLI_YESNO(sip_cfg.alwaysauthreject));
   ast_cli(a->fd, "  Direct RTP setup:       %s\n", AST_CLI_YESNO(sip_cfg.directrtpsetup));
   ast_cli(a->fd, "  User Agent:             %s\n", global_useragent);
   ast_cli(a->fd, "  SDP Session Name:       %s\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
   ast_cli(a->fd, "  SDP Owner Name:         %s\n", ast_strlen_zero(global_sdpowner) ? "-" : global_sdpowner);
   ast_cli(a->fd, "  Reg. context:           %s\n", S_OR(sip_cfg.regcontext, "(not set)"));
   ast_cli(a->fd, "  Regexten on Qualify:    %s\n", AST_CLI_YESNO(sip_cfg.regextenonqualify));
   ast_cli(a->fd, "  Trust RPID:             %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_TRUSTRPID)));
   ast_cli(a->fd, "  Send RPID:              %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_SENDRPID)));
   ast_cli(a->fd, "  Legacy userfield parse: %s\n", AST_CLI_YESNO(sip_cfg.legacy_useroption_parsing));
   ast_cli(a->fd, "  Send Diversion:         %s\n", AST_CLI_YESNO(sip_cfg.send_diversion));
   ast_cli(a->fd, "  Caller ID:              %s\n", default_callerid);
   if ((default_fromdomainport) && (default_fromdomainport != STANDARD_SIP_PORT)) {
      ast_cli(a->fd, "  From: Domain:           %s:%d\n", default_fromdomain, default_fromdomainport);
   } else {
      ast_cli(a->fd, "  From: Domain:           %s\n", default_fromdomain);
   }
   ast_cli(a->fd, "  Record SIP history:     %s\n", AST_CLI_ONOFF(recordhistory));
   ast_cli(a->fd, "  Call Events:            %s\n", AST_CLI_ONOFF(sip_cfg.callevents));
   ast_cli(a->fd, "  Auth. Failure Events:   %s\n", AST_CLI_ONOFF(global_authfailureevents));

   ast_cli(a->fd, "  T.38 support:           %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
   ast_cli(a->fd, "  T.38 EC mode:           %s\n", faxec2str(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
   ast_cli(a->fd, "  T.38 MaxDtgrm:          %d\n", global_t38_maxdatagram);
   if (!realtimepeers && !realtimeregs)
      ast_cli(a->fd, "  SIP realtime:           Disabled\n" );
   else
      ast_cli(a->fd, "  SIP realtime:           Enabled\n" );
   ast_cli(a->fd, "  Qualify Freq :          %d ms\n", global_qualifyfreq);
   ast_cli(a->fd, "  Q.850 Reason header:    %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_Q850_REASON)));
   ast_cli(a->fd, "  Store SIP_CAUSE:        %s\n", AST_CLI_YESNO(global_store_sip_cause));
   ast_cli(a->fd, "\nNetwork QoS Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   ast_cli(a->fd, "  IP ToS SIP:             %s\n", ast_tos2str(global_tos_sip));
   ast_cli(a->fd, "  IP ToS RTP audio:       %s\n", ast_tos2str(global_tos_audio));
   ast_cli(a->fd, "  IP ToS RTP video:       %s\n", ast_tos2str(global_tos_video));
   ast_cli(a->fd, "  IP ToS RTP text:        %s\n", ast_tos2str(global_tos_text));
   ast_cli(a->fd, "  802.1p CoS SIP:         %d\n", global_cos_sip);
   ast_cli(a->fd, "  802.1p CoS RTP audio:   %d\n", global_cos_audio);
   ast_cli(a->fd, "  802.1p CoS RTP video:   %d\n", global_cos_video);
   ast_cli(a->fd, "  802.1p CoS RTP text:    %d\n", global_cos_text);
   ast_cli(a->fd, "  Jitterbuffer enabled:   %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
   if (ast_test_flag(&global_jbconf, AST_JB_ENABLED)) {
      ast_cli(a->fd, "  Jitterbuffer forced:    %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
      ast_cli(a->fd, "  Jitterbuffer max size:  %ld\n", global_jbconf.max_size);
      ast_cli(a->fd, "  Jitterbuffer resync:    %ld\n", global_jbconf.resync_threshold);
      ast_cli(a->fd, "  Jitterbuffer impl:      %s\n", global_jbconf.impl);
      if (!strcasecmp(global_jbconf.impl, "adaptive")) {
         ast_cli(a->fd, "  Jitterbuffer tgt extra: %ld\n", global_jbconf.target_extra);
      }
      ast_cli(a->fd, "  Jitterbuffer log:       %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_LOG)));
   }

   ast_cli(a->fd, "\nNetwork Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   /* determine if/how SIP address can be remapped */
   if (localaddr == NULL)
      msg = "Disabled, no localnet list";
   else if (ast_sockaddr_isnull(&externaddr))
      msg = "Disabled";
   else if (!ast_strlen_zero(externhost))
      msg = "Enabled using externhost";
   else
      msg = "Enabled using externaddr";
   ast_cli(a->fd, "  SIP address remapping:  %s\n", msg);
   ast_cli(a->fd, "  Externhost:             %s\n", S_OR(externhost, "<none>"));
   ast_cli(a->fd, "  Externaddr:             %s\n", ast_sockaddr_stringify(&externaddr));
   ast_cli(a->fd, "  Externrefresh:          %d\n", externrefresh);
   {
      struct ast_ha *d;
      const char *prefix = "Localnet:";

      for (d = localaddr; d ; prefix = "", d = d->next) {
         const char *addr = ast_strdupa(ast_sockaddr_stringify_addr(&d->addr));
         const char *mask = ast_strdupa(ast_sockaddr_stringify_addr(&d->netmask));
         ast_cli(a->fd, "  %-24s%s/%s\n", prefix, addr, mask);
      }
   }
   ast_cli(a->fd, "\nGlobal Signalling Settings:\n");
   ast_cli(a->fd, "---------------------------\n");
   ast_cli(a->fd, "  Codecs:                 ");
   ast_getformatname_multiple(codec_buf, sizeof(codec_buf) -1, sip_cfg.caps);
   ast_cli(a->fd, "%s\n", codec_buf);
   ast_cli(a->fd, "  Codec Order:            ");
   print_codec_to_cli(a->fd, &default_prefs);
   ast_cli(a->fd, "\n");
   ast_cli(a->fd, "  Relax DTMF:             %s\n", AST_CLI_YESNO(global_relaxdtmf));
   ast_cli(a->fd, "  RFC2833 Compensation:   %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_RFC2833_COMPENSATE)));
   ast_cli(a->fd, "  Symmetric RTP:          %s\n", comedia_string(global_flags));
   ast_cli(a->fd, "  Compact SIP headers:    %s\n", AST_CLI_YESNO(sip_cfg.compactheaders));
   ast_cli(a->fd, "  RTP Keepalive:          %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
   ast_cli(a->fd, "  RTP Timeout:            %d %s\n", global_rtptimeout, global_rtptimeout ? "" : "(Disabled)" );
   ast_cli(a->fd, "  RTP Hold Timeout:       %d %s\n", global_rtpholdtimeout, global_rtpholdtimeout ? "" : "(Disabled)");
   ast_cli(a->fd, "  MWI NOTIFY mime type:   %s\n", default_notifymime);
   ast_cli(a->fd, "  DNS SRV lookup:         %s\n", AST_CLI_YESNO(sip_cfg.srvlookup));
   ast_cli(a->fd, "  Pedantic SIP support:   %s\n", AST_CLI_YESNO(sip_cfg.pedanticsipchecking));
   ast_cli(a->fd, "  Reg. min duration       %d secs\n", min_expiry);
   ast_cli(a->fd, "  Reg. max duration:      %d secs\n", max_expiry);
   ast_cli(a->fd, "  Reg. default duration:  %d secs\n", default_expiry);
   ast_cli(a->fd, "  Sub. min duration       %d secs\n", min_subexpiry);
   ast_cli(a->fd, "  Sub. max duration:      %d secs\n", max_subexpiry);
   ast_cli(a->fd, "  Outbound reg. timeout:  %d secs\n", global_reg_timeout);
   ast_cli(a->fd, "  Outbound reg. attempts: %d\n", global_regattempts_max);
   ast_cli(a->fd, "  Outbound reg. retry 403:%d\n", global_reg_retry_403);
   ast_cli(a->fd, "  Notify ringing state:   %s\n", AST_CLI_YESNO(sip_cfg.notifyringing));
   if (sip_cfg.notifyringing) {
      ast_cli(a->fd, "    Include CID:          %s%s\n",
            AST_CLI_YESNO(sip_cfg.notifycid),
            sip_cfg.notifycid == IGNORE_CONTEXT ? " (Ignoring context)" : "");
   }
   ast_cli(a->fd, "  Notify hold state:      %s\n", AST_CLI_YESNO(sip_cfg.notifyhold));
   ast_cli(a->fd, "  SIP Transfer mode:      %s\n", transfermode2str(sip_cfg.allowtransfer));
   ast_cli(a->fd, "  Max Call Bitrate:       %d kbps\n", default_maxcallbitrate);
   ast_cli(a->fd, "  Auto-Framing:           %s\n", AST_CLI_YESNO(global_autoframing));
   ast_cli(a->fd, "  Outb. proxy:            %s %s\n", ast_strlen_zero(sip_cfg.outboundproxy.name) ? "<not set>" : sip_cfg.outboundproxy.name,
                     sip_cfg.outboundproxy.force ? "(forced)" : "");
   ast_cli(a->fd, "  Session Timers:         %s\n", stmode2str(global_st_mode));
   ast_cli(a->fd, "  Session Refresher:      %s\n", strefresherparam2str(global_st_refresher));
   ast_cli(a->fd, "  Session Expires:        %d secs\n", global_max_se);
   ast_cli(a->fd, "  Session Min-SE:         %d secs\n", global_min_se);
   ast_cli(a->fd, "  Timer T1:               %d\n", global_t1);
   ast_cli(a->fd, "  Timer T1 minimum:       %d\n", global_t1min);
   ast_cli(a->fd, "  Timer B:                %d\n", global_timer_b);
   ast_cli(a->fd, "  No premature media:     %s\n", AST_CLI_YESNO(global_prematuremediafilter));
   ast_cli(a->fd, "  Max forwards:           %d\n", sip_cfg.default_max_forwards);

   ast_cli(a->fd, "\nDefault Settings:\n");
   ast_cli(a->fd, "-----------------\n");
   ast_cli(a->fd, "  Allowed transports:     %s\n", get_transport_list(default_transports));
   ast_cli(a->fd, "  Outbound transport:    %s\n", sip_get_transport(default_primary_transport));
   ast_cli(a->fd, "  Context:                %s\n", sip_cfg.default_context);
   ast_cli(a->fd, "  Record on feature:      %s\n", sip_cfg.default_record_on_feature);
   ast_cli(a->fd, "  Record off feature:     %s\n", sip_cfg.default_record_off_feature);
   ast_cli(a->fd, "  Force rport:            %s\n", force_rport_string(global_flags));
   ast_cli(a->fd, "  DTMF:                   %s\n", dtmfmode2str(ast_test_flag(&global_flags[0], SIP_DTMF)));
   ast_cli(a->fd, "  Qualify:                %d\n", default_qualify);
   ast_cli(a->fd, "  Keepalive:              %d\n", default_keepalive);
   ast_cli(a->fd, "  Use ClientCode:         %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_USECLIENTCODE)));
   ast_cli(a->fd, "  Progress inband:        %s\n", (ast_test_flag(&global_flags[0], SIP_PROG_INBAND) == SIP_PROG_INBAND_NEVER) ? "Never" : (AST_CLI_YESNO(ast_test_flag(&global_flags[0], SIP_PROG_INBAND) != SIP_PROG_INBAND_NO)));
   ast_cli(a->fd, "  Language:               %s\n", default_language);
   ast_cli(a->fd, "  Tone zone:              %s\n", default_zone[0] != '\0' ? default_zone : "<Not set>");
   ast_cli(a->fd, "  MOH Interpret:          %s\n", default_mohinterpret);
   ast_cli(a->fd, "  MOH Suggest:            %s\n", default_mohsuggest);
   ast_cli(a->fd, "  Voice Mail Extension:   %s\n", default_vmexten);

   
   if (realtimepeers || realtimeregs) {
      ast_cli(a->fd, "\nRealtime SIP Settings:\n");
      ast_cli(a->fd, "----------------------\n");
      ast_cli(a->fd, "  Realtime Peers:         %s\n", AST_CLI_YESNO(realtimepeers));
      ast_cli(a->fd, "  Realtime Regs:          %s\n", AST_CLI_YESNO(realtimeregs));
      ast_cli(a->fd, "  Cache Friends:          %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_RTCACHEFRIENDS)));
      ast_cli(a->fd, "  Update:                 %s\n", AST_CLI_YESNO(sip_cfg.peer_rtupdate));
      ast_cli(a->fd, "  Ignore Reg. Expire:     %s\n", AST_CLI_YESNO(sip_cfg.ignore_regexpire));
      ast_cli(a->fd, "  Save sys. name:         %s\n", AST_CLI_YESNO(sip_cfg.rtsave_sysname));
      ast_cli(a->fd, "  Auto Clear:             %d (%s)\n", sip_cfg.rtautoclear, ast_test_flag(&global_flags[1], SIP_PAGE2_RTAUTOCLEAR) ? "Enabled" : "Disabled");
   }
   ast_cli(a->fd, "\n----\n");
   return CLI_SUCCESS;
}

static char* sip_show_tcp	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show active TCP connections.

Definition at line 18902 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ast_cli_args::argc, ast_cli(), ast_sockaddr_stringify(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, ast_cli_args::fd, FORMAT, FORMAT2, sip_get_transport(), and ast_cli_entry::usage.

{
   struct sip_threadinfo *th;
   struct ao2_iterator i;

#define FORMAT2 "%-47.47s %9.9s %6.6s\n"
#define FORMAT  "%-47.47s %-9.9s %-6.6s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show tcp";
      e->usage =
         "Usage: sip show tcp\n"
         "       Lists all active TCP/TLS sessions.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   if (a->argc != 3)
      return CLI_SHOWUSAGE;

   ast_cli(a->fd, FORMAT2, "Address", "Transport", "Type");

   i = ao2_iterator_init(threadt, 0);
   while ((th = ao2_t_iterator_next(&i, "iterate through tcp threads for 'sip show tcp'"))) {
      ast_cli(a->fd, FORMAT,
         ast_sockaddr_stringify(&th->tcptls_session->remote_address),
         sip_get_transport(th->type),
         (th->tcptls_session->client ? "Client" : "Server"));
      ao2_t_ref(th, -1, "decrement ref from iterator");
   }
   ao2_iterator_destroy(&i);

   return CLI_SUCCESS;
#undef FORMAT
#undef FORMAT2
}

static char* sip_show_user	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Show one user in detail.

Definition at line 20326 of file chan_sip.c.

References ao2_lock, ao2_unlock, ast_cli_args::argc, ast_cli_args::argv, ast_acl_list_is_empty(), ast_callerid_merge(), ast_cdr_flags2str(), ast_cli(), AST_CLI_YESNO, ast_describe_caller_presentation(), ast_strlen_zero(), CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_show_user(), FALSE, ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_variable::name, ast_variable::next, ast_cli_args::pos, print_codec_to_cli(), print_group(), print_named_groups(), sip_find_peer(), sip_unref_peer(), stmode2str(), strefresherparam2str(), transfermode2str(), TRUE, ast_cli_entry::usage, user, ast_variable::value, and ast_cli_args::word.

{
   char cbuf[256];
   struct sip_peer *user;
   struct ast_variable *v;
   int load_realtime;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show user";
      e->usage =
         "Usage: sip show user <name> [load]\n"
         "       Shows all details on one SIP user and the current status.\n"
         "       Option \"load\" forces lookup of peer in realtime storage.\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_show_user(a->line, a->word, a->pos, a->n);
   }

   if (a->argc < 4)
      return CLI_SHOWUSAGE;

   /* Load from realtime storage? */
   load_realtime = (a->argc == 5 && !strcmp(a->argv[4], "load")) ? TRUE : FALSE;

   if ((user = sip_find_peer(a->argv[3], NULL, load_realtime, FINDUSERS, FALSE, 0))) {
      ao2_lock(user);
      ast_cli(a->fd, "\n\n");
      ast_cli(a->fd, "  * Name       : %s\n", user->name);
      ast_cli(a->fd, "  Secret       : %s\n", ast_strlen_zero(user->secret)?"<Not set>":"<Set>");
      ast_cli(a->fd, "  MD5Secret    : %s\n", ast_strlen_zero(user->md5secret)?"<Not set>":"<Set>");
      ast_cli(a->fd, "  Context      : %s\n", user->context);
      ast_cli(a->fd, "  Language     : %s\n", user->language);
      if (!ast_strlen_zero(user->accountcode))
         ast_cli(a->fd, "  Accountcode  : %s\n", user->accountcode);
      ast_cli(a->fd, "  AMA flags    : %s\n", ast_cdr_flags2str(user->amaflags));
      ast_cli(a->fd, "  Tonezone     : %s\n", user->zone[0] != '\0' ? user->zone : "<Not set>");
      ast_cli(a->fd, "  Transfer mode: %s\n", transfermode2str(user->allowtransfer));
      ast_cli(a->fd, "  MaxCallBR    : %d kbps\n", user->maxcallbitrate);
      ast_cli(a->fd, "  CallingPres  : %s\n", ast_describe_caller_presentation(user->callingpres));
      ast_cli(a->fd, "  Call limit   : %d\n", user->call_limit);
      ast_cli(a->fd, "  Callgroup    : ");
      print_group(a->fd, user->callgroup, 0);
      ast_cli(a->fd, "  Pickupgroup  : ");
      print_group(a->fd, user->pickupgroup, 0);
      ast_cli(a->fd, "  Named Callgr : ");
      print_named_groups(a->fd, user->named_callgroups, 0);
      ast_cli(a->fd, "  Nam. Pickupgr: ");
      print_named_groups(a->fd, user->named_pickupgroups, 0);
      ast_cli(a->fd, "  Callerid     : %s\n", ast_callerid_merge(cbuf, sizeof(cbuf), user->cid_name, user->cid_num, "<unspecified>"));
      ast_cli(a->fd, "  ACL          : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(user->acl) == 0));
      ast_cli(a->fd, "  Sess-Timers  : %s\n", stmode2str(user->stimer.st_mode_oper));
      ast_cli(a->fd, "  Sess-Refresh : %s\n", strefresherparam2str(user->stimer.st_ref));
      ast_cli(a->fd, "  Sess-Expires : %d secs\n", user->stimer.st_max_se);
      ast_cli(a->fd, "  Sess-Min-SE  : %d secs\n", user->stimer.st_min_se);
      ast_cli(a->fd, "  RTP Engine   : %s\n", user->engine);

      ast_cli(a->fd, "  Codec Order  : (");
      print_codec_to_cli(a->fd, &user->prefs);
      ast_cli(a->fd, ")\n");

      ast_cli(a->fd, "  Auto-Framing:  %s \n", AST_CLI_YESNO(user->autoframing));
      if (user->chanvars) {
         ast_cli(a->fd, "  Variables    :\n");
         for (v = user->chanvars ; v ; v = v->next)
            ast_cli(a->fd, "                 %s = %s\n", v->name, v->value);
      }

      ast_cli(a->fd, "\n");

      ao2_unlock(user);
      sip_unref_peer(user, "sip show user");
   } else {
      ast_cli(a->fd, "User %s not found.\n", a->argv[3]);
      ast_cli(a->fd, "\n");
   }

   return CLI_SUCCESS;
}

static char* sip_show_users	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

CLI Command 'SIP Show Users'.

Definition at line 18942 of file chan_sip.c.

References ao2_iterator_destroy(), ao2_iterator_init(), ao2_lock, ao2_t_iterator_next, ao2_unlock, ast_cli_args::argc, ast_cli_args::argv, ast_acl_list_is_empty(), ast_cli(), AST_CLI_YESNO, ast_test_flag, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, FALSE, ast_cli_args::fd, FORMAT, sip_unref_peer(), TRUE, ast_cli_entry::usage, and user.

{
   regex_t regexbuf;
   int havepattern = FALSE;
   struct ao2_iterator user_iter;
   struct sip_peer *user;

#define FORMAT  "%-25.25s  %-15.15s  %-15.15s  %-15.15s  %-5.5s%-10.10s\n"

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip show users";
      e->usage =
         "Usage: sip show users [like <pattern>]\n"
         "       Lists all known SIP users.\n"
         "       Optional regular expression pattern is used to filter the user list.\n";
      return NULL;
   case CLI_GENERATE:
      return NULL;
   }

   switch (a->argc) {
   case 5:
      if (!strcasecmp(a->argv[3], "like")) {
         if (regcomp(&regexbuf, a->argv[4], REG_EXTENDED | REG_NOSUB))
            return CLI_SHOWUSAGE;
         havepattern = TRUE;
      } else
         return CLI_SHOWUSAGE;
   case 3:
      break;
   default:
      return CLI_SHOWUSAGE;
   }

   ast_cli(a->fd, FORMAT, "Username", "Secret", "Accountcode", "Def.Context", "ACL", "Forcerport");

   user_iter = ao2_iterator_init(peers, 0);
   while ((user = ao2_t_iterator_next(&user_iter, "iterate thru peers table"))) {
      ao2_lock(user);
      if (!(user->type & SIP_TYPE_USER)) {
         ao2_unlock(user);
         sip_unref_peer(user, "sip show users");
         continue;
      }

      if (havepattern && regexec(&regexbuf, user->name, 0, NULL, 0)) {
         ao2_unlock(user);
         sip_unref_peer(user, "sip show users");
         continue;
      }

      ast_cli(a->fd, FORMAT, user->name,
         user->secret,
         user->accountcode,
         user->context,
         AST_CLI_YESNO(ast_acl_list_is_empty(user->acl) == 0),
         AST_CLI_YESNO(ast_test_flag(&user->flags[0], SIP_NAT_FORCE_RPORT)));
      ao2_unlock(user);
      sip_unref_peer(user, "sip show users");
   }
   ao2_iterator_destroy(&user_iter);

   if (havepattern)
      regfree(&regexbuf);

   return CLI_SUCCESS;
#undef FORMAT
}

static int sip_sipredirect	(	struct sip_pvt *	p,
		const char *	dest
	)		[static]

Transfer call before connect with a 302 redirect.

Note:

Called by the transfer() dialplan application through the sip_transfer() pbx interface function if the call is in ringing state

Todo:

Fix this function so that we wait for reply to the REFER and react to errors, denials or other issues the other end might have.

Definition at line 33129 of file chan_sip.c.

References AST_CONTROL_TRANSFER, ast_copy_string(), ast_log(), ast_queue_control_data(), ast_string_field_build, ast_strlen_zero(), AST_TRANSFER_SUCCESS, LOG_ERROR, sip_alreadygone(), sip_get_header(), sip_scheddestroy(), and transmit_response_reliable().

Referenced by sip_transfer().

{
   char *cdest;
   char *extension, *domain;

   cdest = ast_strdupa(dest);

   extension = strsep(&cdest, "@");
   domain = cdest;
   if (ast_strlen_zero(extension)) {
      ast_log(LOG_ERROR, "Missing mandatory argument: extension\n");
      return 0;
   }

   /* we'll issue the redirect message here */
   if (!domain) {
      char *local_to_header;
      char to_header[256];

      ast_copy_string(to_header, sip_get_header(&p->initreq, "To"), sizeof(to_header));
      if (ast_strlen_zero(to_header)) {
         ast_log(LOG_ERROR, "Cannot retrieve the 'To' header from the original SIP request!\n");
         return 0;
      }
      if (((local_to_header = strcasestr(to_header, "sip:")) || (local_to_header = strcasestr(to_header, "sips:")))
         && (local_to_header = strchr(local_to_header, '@'))) {
         char ldomain[256];

         memset(ldomain, 0, sizeof(ldomain));
         local_to_header++;
         /* This is okey because lhost and lport are as big as tmp */
         sscanf(local_to_header, "%256[^<>; ]", ldomain);
         if (ast_strlen_zero(ldomain)) {
            ast_log(LOG_ERROR, "Can't find the host address\n");
            return 0;
         }
         domain = ast_strdupa(ldomain);
      }
   }

   ast_string_field_build(p, our_contact, "Transfer <sip:%s@%s>", extension, domain);
   transmit_response_reliable(p, "302 Moved Temporarily", &p->initreq);

   sip_scheddestroy(p, SIP_TRANS_TIMEOUT);   /* Make sure we stop send this reply. */
   sip_alreadygone(p);

   if (p->owner) {
      enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
      ast_queue_control_data(p->owner, AST_CONTROL_TRANSFER, &message, sizeof(message));
   }
   /* hangup here */
   return 0;
}

static struct sip_st_dlg * sip_st_alloc

(

struct sip_pvt *const

p

)

[static, read]

Allocate Session-Timers struct w/in dialog.

Definition at line 8599 of file chan_sip.c.

References ast_calloc, ast_log(), and LOG_ERROR.

Referenced by handle_request_invite_st(), and st_get_mode().

{
   struct sip_st_dlg *stp;

   if (p->stimer) {
      ast_log(LOG_ERROR, "Session-Timer struct already allocated\n");
      return p->stimer;
   }

   if (!(stp = ast_calloc(1, sizeof(struct sip_st_dlg))))
      return NULL;

   p->stimer = stp;

   stp->st_schedid = -1;           /* Session-Timers ast_sched scheduler id */

   return p->stimer;
}

static int sip_standard_port	(	enum sip_transport	type,
		int	port
	)		[static]

Returns the port to use for this socket.

Parameters:

type	The type of transport used
port	Port we are checking to see if it's the standard port.

Note:

port is expected in host byte order

Definition at line 28620 of file chan_sip.c.

Referenced by initreqprep(), and transmit_notify_with_mwi().

{
   if (type & SIP_TRANSPORT_TLS)
      return port == STANDARD_TLS_PORT;
   else
      return port == STANDARD_SIP_PORT;
}

static int sip_subscribe_mwi	(	const char *	value,
		int	lineno
	)		[static]

Parse mwi=> line in sip.conf and add to list.

--- SIP MWI Subscription support

Definition at line 9381 of file chan_sip.c.

References ast_calloc_with_stringfields, ast_copy_string(), ast_log(), ast_string_field_set, ast_strlen_zero(), ASTOBJ_CONTAINER_LINK, ASTOBJ_INIT, ASTOBJ_UNREF, hostname, LOG_WARNING, mailbox, secret, sip_subscribe_mwi_destroy(), and submwil.

Referenced by reload_config().

{
   struct sip_subscription_mwi *mwi;
   int portnum = 0;
   enum sip_transport transport = SIP_TRANSPORT_UDP;
   char buf[256] = "";
   char *username = NULL, *hostname = NULL, *secret = NULL, *authuser = NULL, *porta = NULL, *mailbox = NULL;

   if (!value) {
      return -1;
   }

   ast_copy_string(buf, value, sizeof(buf));

   username = buf;

   if ((hostname = strrchr(buf, '@'))) {
      *hostname++ = '\0';
   } else {
      return -1;
   }

   if ((secret = strchr(username, ':'))) {
      *secret++ = '\0';
      if ((authuser = strchr(secret, ':'))) {
         *authuser++ = '\0';
      }
   }

   if ((mailbox = strchr(hostname, '/'))) {
      *mailbox++ = '\0';
   }

   if (ast_strlen_zero(username) || ast_strlen_zero(hostname) || ast_strlen_zero(mailbox)) {
      ast_log(LOG_WARNING, "Format for MWI subscription is user[:secret[:authuser]]@host[:port]/mailbox at line %d\n", lineno);
      return -1;
   }

   if ((porta = strchr(hostname, ':'))) {
      *porta++ = '\0';
      if (!(portnum = atoi(porta))) {
         ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
         return -1;
      }
   }

   if (!(mwi = ast_calloc_with_stringfields(1, struct sip_subscription_mwi, 256))) {
      return -1;
   }

   ASTOBJ_INIT(mwi);
   ast_string_field_set(mwi, username, username);
   if (secret) {
      ast_string_field_set(mwi, secret, secret);
   }
   if (authuser) {
      ast_string_field_set(mwi, authuser, authuser);
   }
   ast_string_field_set(mwi, hostname, hostname);
   ast_string_field_set(mwi, mailbox, mailbox);
   mwi->resub = -1;
   mwi->portno = portnum;
   mwi->transport = transport;

   ASTOBJ_CONTAINER_LINK(&submwil, mwi);
   ASTOBJ_UNREF(mwi, sip_subscribe_mwi_destroy);

   return 0;
}

static void sip_subscribe_mwi_destroy

(

struct sip_subscription_mwi *

mwi

)

[static]

Destroy MWI subscription object.

Definition at line 6450 of file chan_sip.c.

References ast_free, AST_SCHED_DEL, ast_string_field_free_memory, and sip_destroy().

Referenced by __sip_subscribe_mwi_do(), handle_response_subscribe(), sip_send_all_mwi_subscriptions(), sip_subscribe_mwi(), sip_subscribe_mwi_do(), and unload_module().

{
   if (mwi->call) {
      mwi->call->mwi = NULL;
      sip_destroy(mwi->call);
   }
   
   AST_SCHED_DEL(sched, mwi->resub);
   ast_string_field_free_memory(mwi);
   ast_free(mwi);
}

static int sip_subscribe_mwi_do

(

const void *

data

)

[static]

Send a subscription or resubscription for MWI.

Definition at line 14260 of file chan_sip.c.

References __sip_subscribe_mwi_do(), ASTOBJ_UNREF, and sip_subscribe_mwi_destroy().

Referenced by handle_response_subscribe(), and sip_send_all_mwi_subscriptions().

{
   struct sip_subscription_mwi *mwi = (struct sip_subscription_mwi*)data;
   
   if (!mwi) {
      return -1;
   }
   
   mwi->resub = -1;
   __sip_subscribe_mwi_do(mwi);
   ASTOBJ_UNREF(mwi, sip_subscribe_mwi_destroy);
   
   return 0;
}

static int sip_t38_abort

(

const void *

data

)

[static]

Called to deny a T38 reinvite if the core does not respond to our request.

Definition at line 25016 of file chan_sip.c.

References change_t38_state(), sip_pvt_lock, sip_pvt_unlock, and transmit_response_reliable().

Referenced by handle_request_invite().

{
   struct sip_pvt *p = (struct sip_pvt *) data;

   sip_pvt_lock(p);
   /* an application may have taken ownership of the T.38 negotiation on this
    * channel while we were waiting to grab the lock... if it did, the scheduler
    * id will have been reset to -1, which is our indication that we do *not*
    * want to abort the negotiation process
    */
   if (p->t38id != -1) {
      change_t38_state(p, T38_REJECTED);
      transmit_response_reliable(p, "488 Not acceptable here", &p->initreq);
      p->t38id = -1;
      dialog_unref(p, "unref the dialog ptr from sip_t38_abort, because it held a dialog ptr");
   }
   sip_pvt_unlock(p);
   return 0;
}

static struct ast_tcptls_session_instance* sip_tcp_locate

(

struct ast_sockaddr *

s

)

[static, read]

Find thread for TCP/TLS session (based on IP/Port.

Note:

This function returns an astobj2 reference

Definition at line 28645 of file chan_sip.c.

References ao2_callback, ao2_ref, ao2_t_ref, and threadinfo_locate_cb().

Referenced by sip_prepare_socket().

{
   struct sip_threadinfo *th;
   struct ast_tcptls_session_instance *tcptls_instance = NULL;

   if ((th = ao2_callback(threadt, 0, threadinfo_locate_cb, s))) {
      tcptls_instance = (ao2_ref(th->tcptls_session, +1), th->tcptls_session);
      ao2_t_ref(th, -1, "decrement ref from callback");
   }

   return tcptls_instance;
}

static int sip_tcp_read	(	struct sip_request *	req,
		struct ast_tcptls_session_instance *	tcptls_session,
		int	authenticated,
		time_t	start
	)		[static]

Read SIP request or response from a TCP connection.

Parameters:

req	The request structure to be filled in
tcptls_session	The TCP connection from which to read

Return values:

-1	Failed to read data
0	Successfully read data

Definition at line 2970 of file chan_sip.c.

References ast_debug, ast_log(), ast_sockaddr_stringify(), ast_str_append(), ast_str_buffer(), ast_str_reset(), ast_str_strlen(), ast_wait_for_input(), check_message_integrity(), ast_tcptls_session_instance::client, ast_tcptls_session_instance::fd, LOG_WARNING, MESSAGE_FRAGMENT, ast_tcptls_session_instance::overflow_buf, ast_tcptls_session_instance::remote_address, and sip_check_authtimeout().

Referenced by _sip_tcp_helper_thread().

{
   enum message_integrity message_integrity = MESSAGE_FRAGMENT;

   while (message_integrity == MESSAGE_FRAGMENT) {
      size_t datalen;

      if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
         char readbuf[4097];
         int timeout;
         int res;
         if (!tcptls_session->client && !authenticated) {
            if ((timeout = sip_check_authtimeout(start)) < 0) {
               return -1;
            }

            if (timeout == 0) {
               ast_debug(2, "SIP TCP server timed out\n");
               return -1;
            }
         } else {
            timeout = -1;
         }
         res = ast_wait_for_input(tcptls_session->fd, timeout);
         if (res < 0) {
            ast_debug(2, "SIP TCP server :: ast_wait_for_input returned %d\n", res);
            return -1;
         } else if (res == 0) {
            ast_debug(2, "SIP TCP server timed out\n");
            return -1;
         }

         res = recv(tcptls_session->fd, readbuf, sizeof(readbuf) - 1, 0);
         if (res < 0) {
            ast_debug(2, "SIP TCP server error when receiving data\n");
            return -1;
         } else if (res == 0) {
            ast_debug(2, "SIP TCP server has shut down\n");
            return -1;
         }
         readbuf[res] = '\0';
         ast_str_append(&req->data, 0, "%s", readbuf);
      } else {
         ast_str_append(&req->data, 0, "%s", ast_str_buffer(tcptls_session->overflow_buf));
         ast_str_reset(tcptls_session->overflow_buf);
      }
      
      datalen = ast_str_strlen(req->data);
      if (datalen > SIP_MAX_PACKET_SIZE) {
         ast_log(LOG_WARNING, "Rejecting TCP packet from '%s' because way too large: %zu\n",
            ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
         return -1;
      }

      message_integrity = check_message_integrity(&req->data, &tcptls_session->overflow_buf);
   }

   return 0;
}

static void * sip_tcp_worker_fn

(

void *

data

)

[static]

SIP TCP connection handler.

Definition at line 2580 of file chan_sip.c.

References _sip_tcp_helper_thread().

Referenced by sip_prepare_socket().

{
   struct ast_tcptls_session_instance *tcptls_session = data;

   return _sip_tcp_helper_thread(tcptls_session);
}

static void sip_tcptls_client_args_destructor

(

void *

obj

)

[static]

Definition at line 2459 of file chan_sip.c.

References args, ast_free, ast_ssl_teardown(), ast_tls_config::cafile, ast_tls_config::capath, ast_tls_config::certfile, ast_tls_config::cipher, ast_tcptls_session_args::name, ast_tls_config::pvtfile, and ast_tcptls_session_args::tls_cfg.

Referenced by sip_prepare_socket().

{
   struct ast_tcptls_session_args *args = obj;
   if (args->tls_cfg) {
      ast_free(args->tls_cfg->certfile);
      ast_free(args->tls_cfg->pvtfile);
      ast_free(args->tls_cfg->cipher);
      ast_free(args->tls_cfg->cafile);
      ast_free(args->tls_cfg->capath);

      ast_ssl_teardown(args->tls_cfg);
   }
   ast_free(args->tls_cfg);
   ast_free((char *) args->name);
}

static int sip_tcptls_write	(	struct ast_tcptls_session_instance *	tcptls_session,
		const void *	buf,
		size_t	len
	)		[static]

used to indicate to a tcptls thread that data is ready to be written

Definition at line 2522 of file chan_sip.c.

References ao2_alloc, ao2_lock, ao2_t_find, ao2_t_ref, ao2_unlock, AST_LIST_INSERT_TAIL, ast_log(), ast_str_create(), ast_str_set(), errno, ast_tcptls_session_instance::fd, len(), LOG_ERROR, OBJ_POINTER, and tcptls_packet_destructor().

Referenced by __sip_xmit(), and sip_send_keepalive().

{
   int res = len;
   struct sip_threadinfo *th = NULL;
   struct tcptls_packet *packet = NULL;
   struct sip_threadinfo tmp = {
      .tcptls_session = tcptls_session,
   };
   enum sip_tcptls_alert alert = TCPTLS_ALERT_DATA;

   if (!tcptls_session) {
      return XMIT_ERROR;
   }

   ao2_lock(tcptls_session);

   if ((tcptls_session->fd == -1) ||
      !(th = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread")) ||
      !(packet = ao2_alloc(sizeof(*packet), tcptls_packet_destructor)) ||
      !(packet->data = ast_str_create(len))) {
      goto tcptls_write_setup_error;
   }

   /* goto tcptls_write_error should _NOT_ be used beyond this point */
   ast_str_set(&packet->data, 0, "%s", (char *) buf);
   packet->len = len;

   /* alert tcptls thread handler that there is a packet to be sent.
    * must lock the thread info object to guarantee control of the
    * packet queue */
   ao2_lock(th);
   if (write(th->alert_pipe[1], &alert, sizeof(alert)) == -1) {
      ast_log(LOG_ERROR, "write() to alert pipe failed: %s\n", strerror(errno));
      ao2_t_ref(packet, -1, "could not write to alert pipe, remove packet");
      packet = NULL;
      res = XMIT_ERROR;
   } else { /* it is safe to queue the frame after issuing the alert when we hold the threadinfo lock */
      AST_LIST_INSERT_TAIL(&th->packet_q, packet, entry);
   }
   ao2_unlock(th);

   ao2_unlock(tcptls_session);
   ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo object after finding it");
   return res;

tcptls_write_setup_error:
   if (th) {
      ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo obj, could not create packet");
   }
   if (packet) {
      ao2_t_ref(packet, -1, "could not allocate packet's data");
   }
   ao2_unlock(tcptls_session);

   return XMIT_ERROR;
}

static struct sip_threadinfo* sip_threadinfo_create	(	struct ast_tcptls_session_instance *	tcptls_session,
		int	transport
	)		[static, read]

creates a sip_threadinfo object and links it into the threadt table.

Definition at line 2498 of file chan_sip.c.

References ao2_alloc, ao2_t_link, ao2_t_ref, ast_log(), errno, LOG_ERROR, sip_threadinfo_destructor(), and ast_tcptls_session_instance::ssl.

Referenced by _sip_tcp_helper_thread(), and sip_prepare_socket().

{
   struct sip_threadinfo *th;

   if (!tcptls_session || !(th = ao2_alloc(sizeof(*th), sip_threadinfo_destructor))) {
      return NULL;
   }

   th->alert_pipe[0] = th->alert_pipe[1] = -1;

   if (pipe(th->alert_pipe) == -1) {
      ao2_t_ref(th, -1, "Failed to open alert pipe on sip_threadinfo");
      ast_log(LOG_ERROR, "Could not create sip alert pipe in tcptls thread, error %s\n", strerror(errno));
      return NULL;
   }
   ao2_t_ref(tcptls_session, +1, "tcptls_session ref for sip_threadinfo object");
   th->tcptls_session = tcptls_session;
   th->type = transport ? transport : (tcptls_session->ssl ? SIP_TRANSPORT_TLS: SIP_TRANSPORT_TCP);
   ao2_t_link(threadt, th, "Adding new tcptls helper thread");
   ao2_t_ref(th, -1, "Decrementing threadinfo ref from alloc, only table ref remains");
   return th;
}

static void sip_threadinfo_destructor

(

void *

obj

)

[static]

Definition at line 2475 of file chan_sip.c.

References ao2_t_ref, and AST_LIST_REMOVE_HEAD.

Referenced by sip_threadinfo_create().

{
   struct sip_threadinfo *th = obj;
   struct tcptls_packet *packet;

   if (th->alert_pipe[1] > -1) {
      close(th->alert_pipe[0]);
   }
   if (th->alert_pipe[1] > -1) {
      close(th->alert_pipe[1]);
   }
   th->alert_pipe[0] = th->alert_pipe[1] = -1;

   while ((packet = AST_LIST_REMOVE_HEAD(&th->packet_q, entry))) {
      ao2_t_ref(packet, -1, "thread destruction, removing packet from frame queue");
   }

   if (th->tcptls_session) {
      ao2_t_ref(th->tcptls_session, -1, "remove tcptls_session for sip_threadinfo object");
   }
}

static int sip_tls_read	(	struct sip_request *	req,
		struct sip_request *	reqcpy,
		struct ast_tcptls_session_instance *	tcptls_session,
		int	authenticated,
		time_t	start,
		struct sip_threadinfo *	me
	)		[static]

Read a SIP request or response from a TLS connection.

Because TLS operations are hidden from view via a FILE handle, the logic for reading data is a bit complex, and we have to make periodic checks to be sure we aren't taking too long to perform the necessary action.

Todo:

XXX This should be altered in the future not to use a FILE pointer

Parameters:

req	The request structure to fill in
tcptls_session	The TLS connection on which the data is being received
authenticated	A flag indicating whether authentication has occurred yet. This is only relevant in a server role.
start	The time at which we started attempting to read data. Used in determining if there has been a timeout.
me	Thread info. Used as a means of determining if the session needs to be stoppped.

Return values:

-1	Failed to read data
0	Succeeded in reading data

Todo:

XXX If there's no Content-Length or if the content-length and what we receive is not the same - we should generate an error

Definition at line 2680 of file chan_sip.c.

References ao2_lock, ao2_unlock, ast_debug, ast_log(), ast_sockaddr_stringify(), ast_str_append(), ast_str_strlen(), ast_wait_for_input(), ast_tcptls_session_instance::client, copy_request(), ast_tcptls_session_instance::f, ast_tcptls_session_instance::fd, LOG_WARNING, MIN, parse_request(), ast_tcptls_session_instance::remote_address, sip_check_authtimeout(), and sip_get_header().

Referenced by _sip_tcp_helper_thread().

{
   int res, content_length, after_poll = 1, need_poll = 1;
   size_t datalen = ast_str_strlen(req->data);
   char buf[1024] = "";
   int timeout = -1;
 
   /* Read in headers one line at a time */
   while (datalen < 4 || strncmp(REQ_OFFSET_TO_STR(req, data->used - 4), "\r\n\r\n", 4)) {
      if (!tcptls_session->client && !authenticated) {
         if ((timeout = sip_check_authtimeout(start)) < 0) {
            ast_debug(2, "SIP TLS server failed to determine authentication timeout\n");
            return -1;
         }

         if (timeout == 0) {
            ast_debug(2, "SIP TLS server timed out\n");
            return -1;
         }
      } else {
         timeout = -1;
      }

      /* special polling behavior is required for TLS
       * sockets because of the buffering done in the
       * TLS layer */
      if (need_poll) {
         need_poll = 0;
         after_poll = 1;
         res = ast_wait_for_input(tcptls_session->fd, timeout);
         if (res < 0) {
            ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
            return -1;
         } else if (res == 0) {
            /* timeout */
            ast_debug(2, "SIP TLS server timed out\n");
            return -1;
         }
      }

      ao2_lock(tcptls_session);
      if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
         ao2_unlock(tcptls_session);
         if (after_poll) {
            return -1;
         } else {
            need_poll = 1;
            continue;
         }
      }
      ao2_unlock(tcptls_session);
      after_poll = 0;
      if (me->stop) {
         return -1;
      }
      ast_str_append(&req->data, 0, "%s", buf);

      datalen = ast_str_strlen(req->data);
      if (datalen > SIP_MAX_PACKET_SIZE) {
         ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
            ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
         return -1;
      }
   }
   copy_request(reqcpy, req);
   parse_request(reqcpy);
   /* In order to know how much to read, we need the content-length header */
   if (sscanf(sip_get_header(reqcpy, "Content-Length"), "%30d", &content_length)) {
      while (content_length > 0) {
         size_t bytes_read;
         if (!tcptls_session->client && !authenticated) {
            if ((timeout = sip_check_authtimeout(start)) < 0) {
               return -1;
            }

            if (timeout == 0) {
               ast_debug(2, "SIP TLS server timed out\n");
               return -1;
            }
         } else {
            timeout = -1;
         }

         if (need_poll) {
            need_poll = 0;
            after_poll = 1;
            res = ast_wait_for_input(tcptls_session->fd, timeout);
            if (res < 0) {
               ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
               return -1;
            } else if (res == 0) {
               /* timeout */
               ast_debug(2, "SIP TLS server timed out\n");
               return -1;
            }
         }

         ao2_lock(tcptls_session);
         if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, content_length), tcptls_session->f))) {
            ao2_unlock(tcptls_session);
            if (after_poll) {
               return -1;
            } else {
               need_poll = 1;
               continue;
            }
         }
         buf[bytes_read] = '\0';
         ao2_unlock(tcptls_session);
         after_poll = 0;
         if (me->stop) {
            return -1;
         }
         content_length -= strlen(buf);
         ast_str_append(&req->data, 0, "%s", buf);
      
         datalen = ast_str_strlen(req->data);
         if (datalen > SIP_MAX_PACKET_SIZE) {
            ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
               ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
            return -1;
         }
      }
   }
   /*! \todo XXX If there's no Content-Length or if the content-length and what
               we receive is not the same - we should generate an error */
   return 0;
}

static int sip_transfer	(	struct ast_channel *	ast,
		const char *	dest
	)		[static]

Transfer SIP call.

Definition at line 7502 of file chan_sip.c.

References ast_channel_name(), ast_channel_tech_pvt(), ast_debug, AST_STATE_RING, sip_pvt_lock, sip_pvt_unlock, sip_sipredirect(), and transmit_refer().

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int res;

   if (!p) {
      ast_debug(1, "Asked to transfer channel %s with no pvt; ignoring\n",
            ast_channel_name(ast));
      return -1;
   }

   if (dest == NULL) /* functions below do not take a NULL */
      dest = "";
   sip_pvt_lock(p);
   if (ast_channel_state(ast) == AST_STATE_RING)
      res = sip_sipredirect(p, dest);
   else
      res = transmit_refer(p, dest);
   sip_pvt_unlock(p);
   return res;
}

void* sip_unref_peer	(	struct sip_peer *	peer,
		char *	tag
	)

helper functions to unreference various types of objects. By handling them this way, we don't have to declare the destructor on each call, which removes the chance of errors.

Definition at line 3278 of file chan_sip.c.

References ao2_t_ref.

Referenced by __sip_destroy(), _sip_qualify_peer(), _sip_show_peer(), _sip_show_peers(), _sip_show_peers_one(), build_peer(), check_peer_ok(), complete_sip_peer(), complete_sip_registered_peer(), complete_sip_user(), create_addr(), expire_register(), function_sippeer(), handle_request_invite(), handle_request_notify(), handle_request_subscribe(), handle_response_peerpoke(), manager_sip_peer_status(), match_and_cleanup_peer_sched(), parse_register_contact(), peer_sched_cleanup(), realtime_peer(), receive_message(), reg_source_db(), register_realtime_peers_with_callbackextens(), register_verify(), reload_config(), sip_devicestate(), sip_do_debug_peer(), sip_find_peer_full(), sip_keepalive_all_peers(), sip_msg_send(), sip_poke_all_peers(), sip_poke_noanswer(), sip_poke_peer(), sip_poke_peer_s(), sip_prune_realtime(), sip_report_security_event(), sip_send_keepalive(), sip_show_inuse(), sip_show_user(), sip_show_users(), sip_unregister(), transmit_register(), and update_call_counter().

{
   ao2_t_ref(peer, -1, tag);
   return NULL;
}

static char * sip_unregister	(	struct ast_cli_entry *	e,
		int	cmd,
		struct ast_cli_args *	a
	)		[static]

Unregister (force expiration) a SIP peer in the registry via CLI.

Note:

This function does not tell the SIP device what's going on, so use it with great care.

Definition at line 20504 of file chan_sip.c.

References ast_cli_args::argc, ast_cli_args::argv, ast_cli(), AST_SCHED_DEL_UNREF, CLI_GENERATE, CLI_INIT, CLI_SHOWUSAGE, CLI_SUCCESS, ast_cli_entry::command, complete_sip_unregister(), expire_register(), ast_cli_args::fd, ast_cli_args::line, ast_cli_args::n, ast_cli_args::pos, sip_find_peer(), sip_ref_peer(), sip_unref_peer(), TRUE, ast_cli_entry::usage, and ast_cli_args::word.

{
   struct sip_peer *peer;
   int load_realtime = 0;

   switch (cmd) {
   case CLI_INIT:
      e->command = "sip unregister";
      e->usage =
         "Usage: sip unregister <peer>\n"
         "       Unregister (force expiration) a SIP peer from the registry\n";
      return NULL;
   case CLI_GENERATE:
      return complete_sip_unregister(a->line, a->word, a->pos, a->n);
   }
   
   if (a->argc != 3)
      return CLI_SHOWUSAGE;
   
   if ((peer = sip_find_peer(a->argv[2], NULL, load_realtime, FINDPEERS, TRUE, 0))) {
      if (peer->expire > 0) {
         AST_SCHED_DEL_UNREF(sched, peer->expire,
            sip_unref_peer(peer, "remove register expire ref"));
         expire_register(sip_ref_peer(peer, "ref for expire_register"));
         ast_cli(a->fd, "Unregistered peer \'%s\'\n\n", a->argv[2]);
      } else {
         ast_cli(a->fd, "Peer %s not registered\n", a->argv[2]);
      }
      sip_unref_peer(peer, "sip_unregister: sip_unref_peer via sip_unregister: done with peer from sip_find_peer call");
   } else {
      ast_cli(a->fd, "Peer unknown: \'%s\'. Not unregistered.\n", a->argv[2]);
   }
   
   return CLI_SUCCESS;
}

static void sip_unregister_tests

(

void

)

[static]

SIP test registration.

Definition at line 33652 of file chan_sip.c.

References sip_config_parser_unregister_tests(), sip_dialplan_function_unregister_tests(), and sip_request_parser_unregister_tests().

Referenced by unload_module().

{
   sip_config_parser_unregister_tests();
   sip_request_parser_unregister_tests();
   sip_dialplan_function_unregister_tests();
}

static void sip_websocket_callback	(	struct ast_websocket *	session,
		struct ast_variable *	parameters,
		struct ast_variable *	headers
	)		[static]

SIP WebSocket connection handler.

Definition at line 2588 of file chan_sip.c.

References AST_DYNSTR_BUILD_FAILED, ast_str_create(), ast_str_set(), ast_wait_for_input(), ast_websocket_fd(), ast_websocket_is_secure(), AST_WEBSOCKET_OPCODE_BINARY, AST_WEBSOCKET_OPCODE_CLOSE, AST_WEBSOCKET_OPCODE_TEXT, ast_websocket_read(), ast_websocket_remote_address(), ast_websocket_set_nonblock(), ast_websocket_unref(), deinit_req(), handle_request_do(), and set_socket_transport().

Referenced by load_module(), and unload_module().

{
   int res;

   if (ast_websocket_set_nonblock(session)) {
      goto end;
   }

   while ((res = ast_wait_for_input(ast_websocket_fd(session), -1)) > 0) {
      char *payload;
      uint64_t payload_len;
      enum ast_websocket_opcode opcode;
      int fragmented;

      if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
         /* We err on the side of caution and terminate the session if any error occurs */
         break;
      }

      if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
         struct sip_request req = { 0, };

         if (!(req.data = ast_str_create(payload_len + 1))) {
            goto end;
         }

         if (ast_str_set(&req.data, -1, "%s", payload) == AST_DYNSTR_BUILD_FAILED) {
            deinit_req(&req);
            goto end;
         }

         req.socket.fd = ast_websocket_fd(session);
         set_socket_transport(&req.socket, ast_websocket_is_secure(session) ? SIP_TRANSPORT_WSS : SIP_TRANSPORT_WS);
         req.socket.ws_session = session;

         handle_request_do(&req, ast_websocket_remote_address(session));
         deinit_req(&req);

      } else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
         break;
      }
   }

end:
   ast_websocket_unref(session);
}

static int sip_write	(	struct ast_channel *	ast,
		struct ast_frame *	frame
	)		[static]

Send frame to media channel (rtp)

Definition at line 7292 of file chan_sip.c.

References ast_channel_nativeformats(), ast_channel_readformat(), ast_channel_tech_pvt(), ast_channel_writeformat(), ast_format_cap_iscompatible(), AST_FRAME_IMAGE, AST_FRAME_MODEM, AST_FRAME_TEXT, AST_FRAME_VIDEO, AST_FRAME_VOICE, ast_getformatname(), ast_getformatname_multiple(), ast_log(), ast_rtp_instance_update_source(), ast_rtp_instance_write(), ast_rtp_red_buffer(), ast_set_flag, AST_STATE_UP, ast_test_flag, ast_udptl_write(), ast_frame_subclass::format, ast_frame::frametype, LOG_WARNING, sip_pvt_lock, sip_pvt_unlock, ast_frame::subclass, transmit_provisional_response(), and TRUE.

{
   struct sip_pvt *p = ast_channel_tech_pvt(ast);
   int res = 0;

   switch (frame->frametype) {
   case AST_FRAME_VOICE:
      if (!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast), &frame->subclass.format))) {
         char s1[512];
         ast_log(LOG_WARNING, "Asked to transmit frame type %s, while native formats is %s read/write = %s/%s\n",
            ast_getformatname(&frame->subclass.format),
            ast_getformatname_multiple(s1, sizeof(s1), ast_channel_nativeformats(ast)),
            ast_getformatname(ast_channel_readformat(ast)),
            ast_getformatname(ast_channel_writeformat(ast)));
         return 0;
      }
      if (p) {
         sip_pvt_lock(p);
         if (p->t38.state == T38_ENABLED) {
            /* drop frame, can't sent VOICE frames while in T.38 mode */
            sip_pvt_unlock(p);
            break;
         } else if (p->rtp) {
            /* If channel is not up, activate early media session */
            if ((ast_channel_state(ast) != AST_STATE_UP) &&
                !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
               ast_rtp_instance_update_source(p->rtp);
               if (!global_prematuremediafilter) {
                  p->invitestate = INV_EARLY_MEDIA;
                  transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
                  ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
               }
            }
            p->lastrtptx = time(NULL);
            res = ast_rtp_instance_write(p->rtp, frame);
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_VIDEO:
      if (p) {
         sip_pvt_lock(p);
         if (p->vrtp) {
            /* Activate video early media */
            if ((ast_channel_state(ast) != AST_STATE_UP) &&
                !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
               p->invitestate = INV_EARLY_MEDIA;
               transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
               ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
            }
            p->lastrtptx = time(NULL);
            res = ast_rtp_instance_write(p->vrtp, frame);
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_TEXT:
      if (p) {
         sip_pvt_lock(p);
         if (p->red) {
            ast_rtp_red_buffer(p->trtp, frame);
         } else {
            if (p->trtp) {
               /* Activate text early media */
               if ((ast_channel_state(ast) != AST_STATE_UP) &&
                   !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT) &&
                   !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
                  p->invitestate = INV_EARLY_MEDIA;
                  transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
                  ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
               }
               p->lastrtptx = time(NULL);
               res = ast_rtp_instance_write(p->trtp, frame);
            }
         }
         sip_pvt_unlock(p);
      }
      break;
   case AST_FRAME_IMAGE:
      return 0;
      break;
   case AST_FRAME_MODEM:
      if (p) {
         sip_pvt_lock(p);
         /* UDPTL requires two-way communication, so early media is not needed here.
            we simply forget the frames if we get modem frames before the bridge is up.
            Fax will re-transmit.
         */
         if ((ast_channel_state(ast) == AST_STATE_UP) &&
             p->udptl &&
             (p->t38.state == T38_ENABLED)) {
            res = ast_udptl_write(p->udptl, frame);
         }
         sip_pvt_unlock(p);
      }
      break;
   default:
      ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
      return 0;
   }

   return res;
}

static int sipinfo_send	(	struct ast_channel *	chan,
		struct ast_variable *	headers,
		const char *	content_type,
		const char *	content,
		const char *	useragent_filter
	)		[static]

Definition at line 7654 of file chan_sip.c.

References add_content(), add_header(), ast_channel_lock, ast_channel_name(), ast_channel_tech(), ast_channel_tech_pvt(), ast_channel_unlock, ast_log(), ast_strlen_zero(), cleanup(), LOG_WARNING, match(), ast_variable::name, ast_variable::next, reqprep(), send_request(), sip_pvt_lock, sip_pvt_unlock, ast_variable::value, and var.

{
   struct sip_pvt *p;
   struct ast_variable *var;
   struct sip_request req;
   int res = -1;

   ast_channel_lock(chan);

   if (ast_channel_tech(chan) != &sip_tech) {
      ast_log(LOG_WARNING, "Attempted to send a custom INFO on a non-SIP channel %s\n", ast_channel_name(chan));
      ast_channel_unlock(chan);
      return res;
   }

   p = ast_channel_tech_pvt(chan);
   sip_pvt_lock(p);

   if (!(ast_strlen_zero(useragent_filter))) {
      int match = (strstr(p->useragent, useragent_filter)) ? 1 : 0;
      if (!match) {
         goto cleanup;
      }
   }

   reqprep(&req, p, SIP_INFO, 0, 1);
   for (var = headers; var; var = var->next) {
      add_header(&req, var->name, var->value);
   }
   if (!ast_strlen_zero(content) && !ast_strlen_zero(content_type)) {
      add_header(&req, "Content-Type", content_type);
      add_content(&req, content);
   }

   res = send_request(p, &req, XMIT_RELIABLE, p->ocseq);

cleanup:
   sip_pvt_unlock(p);
   ast_channel_unlock(chan);
   return res;
}

static int sipsock_read	(	int *	id,
		int	fd,
		short	events,
		void *	ignore
	)		[static]

Read data from SIP UDP socket.

Note:

sipsock_read locks the owner channel while we are processing the SIP message

Returns:

1 on error, 0 on success

Note:

Successful messages is connected to SIP call and forwarded to handle_incoming()

Definition at line 28484 of file chan_sip.c.

References AST_DYNSTR_BUILD_FAILED, ast_log(), ast_recvfrom(), ast_sockaddr_port, ast_str_create(), ast_str_set(), bindaddr, deinit_req(), errno, handle_request_do(), LOG_NOTICE, LOG_WARNING, set_socket_transport(), and sipsock.

Referenced by do_monitor().

{
   struct sip_request req;
   struct ast_sockaddr addr;
   int res;
   static char readbuf[65535];

   memset(&req, 0, sizeof(req));
   res = ast_recvfrom(fd, readbuf, sizeof(readbuf) - 1, 0, &addr);
   if (res < 0) {
#if !defined(__FreeBSD__)
      if (errno == EAGAIN)
         ast_log(LOG_NOTICE, "SIP: Received packet with bad UDP checksum\n");
      else
#endif
      if (errno != ECONNREFUSED)
         ast_log(LOG_WARNING, "Recv error: %s\n", strerror(errno));
      return 1;
   }

   readbuf[res] = '\0';

   if (!(req.data = ast_str_create(SIP_MIN_PACKET))) {
      return 1;
   }

   if (ast_str_set(&req.data, 0, "%s", readbuf) == AST_DYNSTR_BUILD_FAILED) {
      return -1;
   }

   req.socket.fd = sipsock;
   set_socket_transport(&req.socket, SIP_TRANSPORT_UDP);
   req.socket.tcptls_session  = NULL;
   req.socket.port = htons(ast_sockaddr_port(&bindaddr));

   handle_request_do(&req, &addr);
   deinit_req(&req);

   return 1;
}

static int sockaddr_is_null_or_any

(

const struct ast_sockaddr *

addr

)

[static]

Definition at line 9886 of file chan_sip.c.

References ast_sockaddr_is_any(), and ast_sockaddr_isnull().

Referenced by process_sdp().

{
   return ast_sockaddr_isnull(addr) || ast_sockaddr_is_any(addr);
}

enum st_mode st_get_mode	(	struct sip_pvt *	p,
		int	no_cached
	)		[static]

Get the session-timer mode.

Parameters:

p	pointer to the SIP dialog
no_cached,set	this to true in order to force a peername lookup on the session timer mode.

Definition at line 29488 of file chan_sip.c.

References global_st_mode, and sip_st_alloc().

Referenced by add_supported(), handle_request_invite_st(), handle_response_invite(), and transmit_invite().

{
   if (!p->stimer) {
      sip_st_alloc(p);
      if (!p->stimer) {
         return SESSION_TIMER_MODE_INVALID;
      }
   }

   if (!no_cached && p->stimer->st_cached_mode != SESSION_TIMER_MODE_INVALID)
      return p->stimer->st_cached_mode;

   if (p->relatedpeer) {
      p->stimer->st_cached_mode = p->relatedpeer->stimer.st_mode_oper;
      return p->stimer->st_cached_mode;
   }

   p->stimer->st_cached_mode = global_st_mode;
   return global_st_mode;
}

enum st_refresher st_get_refresher

(

struct sip_pvt *

p

)

[static]

Get the entity (UAC or UAS) that's acting as the session-timer refresher.

Note:

This is only called when processing an INVITE, so in that case Asterisk is always currently the UAS. If this is ever used to process responses, the function will have to be changed.

Parameters:

p	pointer to the SIP dialog

Definition at line 29466 of file chan_sip.c.

Referenced by handle_request_invite_st().

{
   if (p->stimer->st_cached_ref != SESSION_TIMER_REFRESHER_AUTO) {
      return p->stimer->st_cached_ref;
   }

   if (p->relatedpeer) {
      p->stimer->st_cached_ref = (p->relatedpeer->stimer.st_ref == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
      return p->stimer->st_cached_ref;
   }
   
   p->stimer->st_cached_ref = (global_st_refresher == SESSION_TIMER_REFRESHER_PARAM_UAC) ? SESSION_TIMER_REFRESHER_THEM : SESSION_TIMER_REFRESHER_US;
   return p->stimer->st_cached_ref;
}

int st_get_se	(	struct sip_pvt *	p,
		int	max
	)		[static]

Get Max or Min SE (session timer expiry)

Parameters:

p	pointer to the SIP dialog
max	if true, get max se, otherwise min se

Definition at line 29434 of file chan_sip.c.

References global_max_se, global_min_se, and TRUE.

Referenced by handle_request_invite_st(), handle_response_invite(), reqprep(), and transmit_invite().

{
   if (max == TRUE) {
      if (p->stimer->st_cached_max_se) {
         return  p->stimer->st_cached_max_se;
      }
      if (p->relatedpeer) {
         p->stimer->st_cached_max_se = p->relatedpeer->stimer.st_max_se;
         return (p->stimer->st_cached_max_se);
      }
      p->stimer->st_cached_max_se = global_max_se;
      return (p->stimer->st_cached_max_se);
   } 
   /* Find Min SE timer */
   if (p->stimer->st_cached_min_se) {
      return p->stimer->st_cached_min_se;
   } 
   if (p->relatedpeer) {
      p->stimer->st_cached_min_se = p->relatedpeer->stimer.st_min_se;
      return (p->stimer->st_cached_min_se);
   }
   p->stimer->st_cached_min_se = global_min_se;
   return (p->stimer->st_cached_min_se);
}

static void start_ice

(

struct ast_rtp_instance *

instance

)

[static]

Start ICE negotiation on an RTP instance.

Definition at line 12697 of file chan_sip.c.

References ast_rtp_instance_get_ice(), and ast_rtp_engine_ice::start.

Referenced by process_sdp().

{
   struct ast_rtp_engine_ice *ice = ast_rtp_instance_get_ice(instance);

   if (!ice) {
      return;
   }

   ice->start(instance);
}

static void start_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Start session timer.

Definition at line 29213 of file chan_sip.c.

References ast_debug, ast_log(), ast_sched_add(), AST_SCHED_DEL_UNREF, LOG_ERROR, LOG_WARNING, MIN, proc_session_timer(), and TRUE.

Referenced by handle_request_invite(), handle_response_invite(), and restart_session_timer().

{
   unsigned int timeout_ms;

   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in start_session_timer - %s\n", p->callid);
      return;
   }

   if (p->stimer->st_schedid > -1) {
      /* in the event a timer is already going, stop it */
      ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
      AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
         dialog_unref(p, "unref stimer->st_schedid from dialog"));
   }

   /*
    * RFC 4028 Section 10
    * If the side not performing refreshes does not receive a
    * session refresh request before the session expiration, it SHOULD send
    * a BYE to terminate the session, slightly before the session
    * expiration.  The minimum of 32 seconds and one third of the session
    * interval is RECOMMENDED.
    */

   timeout_ms = (1000 * p->stimer->st_interval);
   if (p->stimer->st_ref == SESSION_TIMER_REFRESHER_US) {
      timeout_ms /= 2;
   } else {
      timeout_ms -= MIN(timeout_ms / 3, 32000);
   }

   p->stimer->st_schedid = ast_sched_add(sched, timeout_ms, proc_session_timer,
         dialog_ref(p, "adding session timer ref"));

   if (p->stimer->st_schedid < 0) {
      dialog_unref(p, "removing session timer ref");
      ast_log(LOG_ERROR, "ast_sched_add failed - %s\n", p->callid);
   } else {
      p->stimer->st_active = TRUE;
      ast_debug(2, "Session timer started: %d - %s %ums\n", p->stimer->st_schedid, p->callid, timeout_ms);
   }
}

static void state_notify_build_xml	(	struct state_notify_data *	data,
		int	full,
		const char *	exten,
		const char *	context,
		struct ast_str **	tmp,
		struct sip_pvt *	p,
		int	subscribed,
		const char *	mfrom,
		const char *	mto
	)		[static]

Builds XML portion of NOTIFY messages for presence or dialog updates.

Definition at line 14460 of file chan_sip.c.

References allow_notify_user_presence(), ast_alloca, ast_channel_caller(), ast_channel_connected(), ast_channel_lock, ast_channel_unlock, ast_channel_unref, AST_DEVICE_UNAVAILABLE, AST_EXTENSION_BUSY, AST_EXTENSION_INUSE, AST_EXTENSION_NOT_INUSE, AST_EXTENSION_ONHOLD, AST_EXTENSION_RINGING, AST_EXTENSION_UNAVAILABLE, ast_get_hint(), AST_MAX_EXTENSION, AST_PRES_RESTRICTED, AST_PRES_RESTRICTION, AST_PRESENCE_INVALID, AST_PRESENCE_NOT_SET, ast_presence_state2str(), ast_str_append(), ast_test_suite_event_notify, ast_xml_escape(), cid_num, state_notify_data::device_state_info, find_ringing_channel(), ast_party_caller::id, ast_party_connected_line::id, name, NONE, ast_party_id::number, state_notify_data::presence_message, state_notify_data::presence_state, state_notify_data::presence_subtype, ast_party_number::presentation, S_COR, S_OR, sip_cfg, and state_notify_data::state.

Referenced by transmit_state_notify().

{
   enum state { NOTIFY_OPEN, NOTIFY_INUSE, NOTIFY_CLOSED } local_state = NOTIFY_OPEN;
   const char *statestring = "terminated";
   const char *pidfstate = "--";
   const char *pidfnote ="Ready";
   char hint[AST_MAX_EXTENSION];

   switch (data->state) {
   case (AST_EXTENSION_RINGING | AST_EXTENSION_INUSE):
      statestring = (sip_cfg.notifyringing) ? "early" : "confirmed";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "Ringing";
      break;
   case AST_EXTENSION_RINGING:
      statestring = "early";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "Ringing";
      break;
   case AST_EXTENSION_INUSE:
      statestring = "confirmed";
      local_state = NOTIFY_INUSE;
      pidfstate = "busy";
      pidfnote = "On the phone";
      break;
   case AST_EXTENSION_BUSY:
      statestring = "confirmed";
      local_state = NOTIFY_CLOSED;
      pidfstate = "busy";
      pidfnote = "On the phone";
      break;
   case AST_EXTENSION_UNAVAILABLE:
      statestring = "terminated";
      local_state = NOTIFY_CLOSED;
      pidfstate = "away";
      pidfnote = "Unavailable";
      break;
   case AST_EXTENSION_ONHOLD:
      statestring = "confirmed";
      local_state = NOTIFY_CLOSED;
      pidfstate = "busy";
      pidfnote = "On hold";
      break;
   case AST_EXTENSION_NOT_INUSE:
   default:
      /* Default setting */
      break;
   }

   /* Check which device/devices we are watching  and if they are registered */
   if (ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, context, exten)) {
      char *hint2;
      char *individual_hint = NULL;
      int hint_count = 0, unavailable_count = 0;

      /* strip off any possible PRESENCE providers from hint */
      if ((hint2 = strrchr(hint, ','))) {
         *hint2 = '\0';
      }
      hint2 = hint;

      while ((individual_hint = strsep(&hint2, "&"))) {
         hint_count++;

         if (ast_device_state(individual_hint) == AST_DEVICE_UNAVAILABLE)
            unavailable_count++;
      }

      /* If none of the hinted devices are registered, we will
       * override notification and show no availability.
       */
      if (hint_count > 0 && hint_count == unavailable_count) {
         local_state = NOTIFY_CLOSED;
         pidfstate = "away";
         pidfnote = "Not online";
      }
   }

   switch (subscribed) {
   case XPIDF_XML:
   case CPIM_PIDF_XML:
      ast_str_append(tmp, 0,
         "<?xml version=\"1.0\"?>\n"
         "<!DOCTYPE presence PUBLIC \"-//IETF//DTD RFCxxxx XPIDF 1.0//EN\" \"xpidf.dtd\">\n"
         "<presence>\n");
      ast_str_append(tmp, 0, "<presentity uri=\"%s;method=SUBSCRIBE\" />\n", mfrom);
      ast_str_append(tmp, 0, "<atom id=\"%s\">\n", exten);
      ast_str_append(tmp, 0, "<address uri=\"%s;user=ip\" priority=\"0.800000\">\n", mto);
      ast_str_append(tmp, 0, "<status status=\"%s\" />\n", (local_state ==  NOTIFY_OPEN) ? "open" : (local_state == NOTIFY_INUSE) ? "inuse" : "closed");
      ast_str_append(tmp, 0, "<msnsubstatus substatus=\"%s\" />\n", (local_state == NOTIFY_OPEN) ? "online" : (local_state == NOTIFY_INUSE) ? "onthephone" : "offline");
      ast_str_append(tmp, 0, "</address>\n</atom>\n</presence>\n");
      break;
   case PIDF_XML: /* Eyebeam supports this format */
      ast_str_append(tmp, 0,
         "<?xml version=\"1.0\" encoding=\"ISO-8859-1\"?>\n"
         "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" \nxmlns:pp=\"urn:ietf:params:xml:ns:pidf:person\"\nxmlns:es=\"urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status\"\nxmlns:ep=\"urn:ietf:params:xml:ns:pidf:rpid:rpid-person\"\nentity=\"%s\">\n", mfrom);
      ast_str_append(tmp, 0, "<pp:person><status>\n");
      if (pidfstate[0] != '-') {
         ast_str_append(tmp, 0, "<ep:activities><ep:%s/></ep:activities>\n", pidfstate);
      }
      ast_str_append(tmp, 0, "</status></pp:person>\n");
      ast_str_append(tmp, 0, "<note>%s</note>\n", pidfnote); /* Note */
      ast_str_append(tmp, 0, "<tuple id=\"%s\">\n", exten); /* Tuple start */
      ast_str_append(tmp, 0, "<contact priority=\"1\">%s</contact>\n", mto);
      if (pidfstate[0] == 'b') /* Busy? Still open ... */
         ast_str_append(tmp, 0, "<status><basic>open</basic></status>\n");
      else
         ast_str_append(tmp, 0, "<status><basic>%s</basic></status>\n", (local_state != NOTIFY_CLOSED) ? "open" : "closed");

      if (allow_notify_user_presence(p) && (data->presence_state != AST_PRESENCE_INVALID)
            && (data->presence_state != AST_PRESENCE_NOT_SET)) {
         ast_str_append(tmp, 0, "</tuple>\n");
         ast_str_append(tmp, 0, "<tuple id=\"digium-presence\">\n");
         ast_str_append(tmp, 0, "<status>\n");
         ast_str_append(tmp, 0, "<digium_presence type=\"%s\" subtype=\"%s\">%s</digium_presence>\n",
            ast_presence_state2str(data->presence_state),
            S_OR(data->presence_subtype, ""),
            S_OR(data->presence_message, ""));
         ast_str_append(tmp, 0, "</status>\n");
         ast_test_suite_event_notify("DIGIUM_PRESENCE_SENT",
               "PresenceState: %s\r\n"
               "Subtype: %s\r\n"
               "Message: %s",
               ast_presence_state2str(data->presence_state),
               S_OR(data->presence_subtype, ""),
               S_OR(data->presence_message, ""));
      }
      ast_str_append(tmp, 0, "</tuple>\n</presence>\n");
      break;
   case DIALOG_INFO_XML: /* SNOM subscribes in this format */
      ast_str_append(tmp, 0, "<?xml version=\"1.0\"?>\n");
      ast_str_append(tmp, 0, "<dialog-info xmlns=\"urn:ietf:params:xml:ns:dialog-info\" version=\"%u\" state=\"%s\" entity=\"%s\">\n", p->dialogver, full ? "full" : "partial", mto);
      if (data->state > 0 && (data->state & AST_EXTENSION_RINGING) && sip_cfg.notifyringing) {
         /* Twice the extension length should be enough for XML encoding */
         char local_display[AST_MAX_EXTENSION * 2];
         char remote_display[AST_MAX_EXTENSION * 2];
         char *local_target = ast_strdupa(mto);
         /* It may seem odd to base the remote_target on the To header here,
          * but testing by reporters on issue ASTERISK-16735 found that basing
          * on the From header would cause ringing state hints to not work
          * properly on certain SNOM devices. If you are using notifycid properly
          * (i.e. in the same extension and context as the dialed call) then this
          * should not be an issue since the data will be overwritten shortly
          * with channel caller ID
          */
         char *remote_target = ast_strdupa(mto);

         ast_xml_escape(exten, local_display, sizeof(local_display));
         ast_xml_escape(exten, remote_display, sizeof(remote_display));

         /* There are some limitations to how this works.  The primary one is that the
            callee must be dialing the same extension that is being monitored.  Simply dialing
            the hint'd device is not sufficient. */
         if (sip_cfg.notifycid) {
            struct ast_channel *callee;

            callee = find_ringing_channel(data->device_state_info, p);
            if (callee) {
               static char *anonymous = "anonymous";
               static char *invalid = "anonymous.invalid";
               char *cid_num;
               char *connected_num;
               int need;
               int cid_num_restricted, connected_num_restricted;

               ast_channel_lock(callee);

               cid_num_restricted = (ast_channel_caller(callee)->id.number.presentation &
                           AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
               cid_num = S_COR(ast_channel_caller(callee)->id.number.valid,
                     S_COR(cid_num_restricted, anonymous,
                           ast_channel_caller(callee)->id.number.str), "");

               need = strlen(cid_num) + (cid_num_restricted ? strlen(invalid) :
                          strlen(p->fromdomain)) + sizeof("sip:@");
               local_target = ast_alloca(need);

               snprintf(local_target, need, "sip:%s@%s", cid_num,
                   cid_num_restricted ? invalid : p->fromdomain);

               ast_xml_escape(S_COR(ast_channel_caller(callee)->id.name.valid,
                          S_COR((ast_channel_caller(callee)->id.name.presentation &
                             AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
                           ast_channel_caller(callee)->id.name.str), ""),
                         local_display, sizeof(local_display));

               connected_num_restricted = (ast_channel_connected(callee)->id.number.presentation &
                            AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED;
               connected_num = S_COR(ast_channel_connected(callee)->id.number.valid,
                           S_COR(connected_num_restricted, anonymous,
                            ast_channel_connected(callee)->id.number.str), "");

               need = strlen(connected_num) + (connected_num_restricted ? strlen(invalid) :
                           strlen(p->fromdomain)) + sizeof("sip:@");
               remote_target = ast_alloca(need);

               snprintf(remote_target, need, "sip:%s@%s", connected_num,
                   connected_num_restricted ? invalid : p->fromdomain);

               ast_xml_escape(S_COR(ast_channel_connected(callee)->id.name.valid,
                          S_COR((ast_channel_connected(callee)->id.name.presentation &
                             AST_PRES_RESTRICTION) == AST_PRES_RESTRICTED, anonymous,
                            ast_channel_connected(callee)->id.name.str), ""),
                         remote_display, sizeof(remote_display));

               ast_channel_unlock(callee);
               callee = ast_channel_unref(callee);
            }

            /* We create a fake call-id which the phone will send back in an INVITE
               Replaces header which we can grab and do some magic with. */
            if (sip_cfg.pedanticsipchecking) {
               ast_str_append(tmp, 0, "<dialog id=\"%s\" call-id=\"pickup-%s\" local-tag=\"%s\" remote-tag=\"%s\" direction=\"recipient\">\n",
                  exten, p->callid, p->theirtag, p->tag);
            } else {
               ast_str_append(tmp, 0, "<dialog id=\"%s\" call-id=\"pickup-%s\" direction=\"recipient\">\n",
                  exten, p->callid);
            }
            ast_str_append(tmp, 0,
                  "<remote>\n"
                  /* See the limitations of this above.  Luckily the phone seems to still be
                     happy when these values are not correct. */
                  "<identity display=\"%s\">%s</identity>\n"
                  "<target uri=\"%s\"/>\n"
                  "</remote>\n"
                  "<local>\n"
                  "<identity display=\"%s\">%s</identity>\n"
                  "<target uri=\"%s\"/>\n"
                  "</local>\n",
                  remote_display, remote_target, remote_target, local_display, local_target, local_target);
         } else {
            ast_str_append(tmp, 0, "<dialog id=\"%s\" direction=\"recipient\">\n", exten);
         }

      } else {
         ast_str_append(tmp, 0, "<dialog id=\"%s\">\n", exten);
      }
      ast_str_append(tmp, 0, "<state>%s</state>\n", statestring);
      if (data->state == AST_EXTENSION_ONHOLD) {
            ast_str_append(tmp, 0, "<local>\n<target uri=\"%s\">\n"
                                             "<param pname=\"+sip.rendering\" pvalue=\"no\"/>\n"
                                             "</target>\n</local>\n", mto);
      }
      ast_str_append(tmp, 0, "</dialog>\n</dialog-info>\n");
      break;
   case NONE:
   default:
      break;
   }
}

static const char* stmode2str

(

enum st_mode

m

)

[static]

Definition at line 18825 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   return map_x_s(stmodes, m, "Unknown");
}

static void stop_media_flows

(

struct sip_pvt *

p

)

[static]

Immediately stop RTP, VRTP and UDPTL as applicable.

Definition at line 23795 of file chan_sip.c.

References ast_rtp_instance_stop(), and ast_udptl_stop().

Referenced by __sip_autodestruct(), handle_request_bye(), handle_request_cancel(), handle_response(), and sip_hangup().

{
   /* Immediately stop RTP, VRTP and UDPTL as applicable */
   if (p->rtp)
      ast_rtp_instance_stop(p->rtp);
   if (p->vrtp)
      ast_rtp_instance_stop(p->vrtp);
   if (p->trtp)
      ast_rtp_instance_stop(p->trtp);
   if (p->udptl)
      ast_udptl_stop(p->udptl);
}

static void stop_session_timer

(

struct sip_pvt *

p

)

[static]

Session-Timers: Stop session timer.

Definition at line 29196 of file chan_sip.c.

References ast_debug, ast_log(), AST_SCHED_DEL_UNREF, FALSE, LOG_WARNING, and TRUE.

Referenced by __sip_destroy(), dialog_unlink_all(), handle_request_bye(), proc_session_timer(), sip_hangup(), and sip_scheddestroy().

{
   if (!p->stimer) {
      ast_log(LOG_WARNING, "Null stimer in stop_session_timer - %s\n", p->callid);
      return;
   }

   if (p->stimer->st_active == TRUE) {
      p->stimer->st_active = FALSE;
      ast_debug(2, "Session timer stopped: %d - %s\n", p->stimer->st_schedid, p->callid);
      AST_SCHED_DEL_UNREF(sched, p->stimer->st_schedid,
            dialog_unref(p, "removing session timer ref"));
   }
}

static int str2dtmfmode

(

const char *

str

)

[static]

maps a string to dtmfmode, returns -1 on error

Definition at line 19425 of file chan_sip.c.

References map_s_x().

{
   return map_s_x(dtmfstr, str, -1);
}

static enum st_mode str2stmode

(

const char *

s

)

[static]

Definition at line 18830 of file chan_sip.c.

References map_s_x().

Referenced by build_peer(), and reload_config().

{
   return map_s_x(stmodes, s, -1);
}

static enum st_refresher str2strefresherparam

(

const char *

s

)

[static]

Definition at line 18855 of file chan_sip.c.

References map_s_x().

Referenced by build_peer(), and reload_config().

{
   return map_s_x(strefresher_params, s, -1);
}

static const char* strefresher2str

(

enum st_refresher

r

)

[static]

Definition at line 18868 of file chan_sip.c.

References map_x_s().

Referenced by sip_show_channel().

{
   return map_x_s(strefreshers, r, "Unknown");
}

static const char * strefresherparam2str

(

enum st_refresher

r

)

[static]

Definition at line 18850 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   return map_x_s(strefresher_params, r, "Unknown");
}

static const char * subscription_type2str

(

enum subscriptiontype

subtype

)

[static]

Show subscription type in string format.

Definition at line 20910 of file chan_sip.c.

References ARRAY_LEN, subscription_types, cfsubscription_types::text, and type.

Referenced by show_channels_cb(), and sip_show_channel().

{
   int i;

   for (i = 1; i < ARRAY_LEN(subscription_types); i++) {
      if (subscription_types[i].type == subtype) {
         return subscription_types[i].text;
      }
   }
   return subscription_types[0].text;
}

static unsigned int t38_get_rate

(

enum ast_control_t38_rate

rate

)

[static]

Get Max T.38 Transmission rate from T38 capabilities.

Definition at line 12881 of file chan_sip.c.

References AST_T38_RATE_12000, AST_T38_RATE_14400, AST_T38_RATE_2400, AST_T38_RATE_4800, AST_T38_RATE_7200, and AST_T38_RATE_9600.

Referenced by add_sdp().

{
   switch (rate) {
   case AST_T38_RATE_2400:
      return 2400;
   case AST_T38_RATE_4800:
      return 4800;
   case AST_T38_RATE_7200:
      return 7200;
   case AST_T38_RATE_9600:
      return 9600;
   case AST_T38_RATE_12000:
      return 12000;
   case AST_T38_RATE_14400:
      return 14400;
   default:
      return 0;
   }
}

static void tcptls_packet_destructor

(

void *

obj

)

[static]

Definition at line 2452 of file chan_sip.c.

References ast_free.

Referenced by sip_tcptls_write().

{
   struct tcptls_packet *packet = obj;

   ast_free(packet->data);
}

static struct sip_peer * temp_peer

(

const char *

name

)

[static, read]

Create temporary peer (used in autocreatepeer mode)

Definition at line 30542 of file chan_sip.c.

References ao2_t_alloc, ao2_t_ref, ast_atomic_fetchadd_int(), ast_cc_config_params_init, ast_copy_string(), ast_format_cap_alloc_nolock(), ast_string_field_init, default_prefs, reg_source_db(), set_peer_defaults(), sip_destroy_peer_fn(), and TRUE.

Referenced by load_module(), register_verify(), and sip_reload().

{
   struct sip_peer *peer;

   if (!(peer = ao2_t_alloc(sizeof(*peer), sip_destroy_peer_fn, "allocate a peer struct")))
      return NULL;

   if (ast_string_field_init(peer, 512)) {
      ao2_t_ref(peer, -1, "failed to string_field_init, drop peer");
      return NULL;
   }
   
   if (!(peer->cc_params = ast_cc_config_params_init())) {
      ao2_t_ref(peer, -1, "failed to allocate cc_params for peer");
      return NULL;
   }

   if (!(peer->caps = ast_format_cap_alloc_nolock())) {
      ao2_t_ref(peer, -1, "failed to allocate format capabilities, drop peer");
      return NULL;
   }

   ast_atomic_fetchadd_int(&apeerobjs, 1);
   set_peer_defaults(peer);

   ast_copy_string(peer->name, name, sizeof(peer->name));

   peer->selfdestruct = TRUE;
   peer->host_dynamic = TRUE;
   peer->prefs = default_prefs;
   reg_source_db(peer);

   return peer;
}

static void temp_pvt_cleanup

(

void *

data

)

[static]

Definition at line 12105 of file chan_sip.c.

References ast_free, and ast_string_field_free_memory.

{
   struct sip_pvt *p = data;

   ast_string_field_free_memory(p);

   ast_free(data);
}

static int temp_pvt_init

(

void *

data

)

[static]

Definition at line 12097 of file chan_sip.c.

References ast_string_field_init.

{
   struct sip_pvt *p = data;

   p->do_history = 0;   /* XXX do we need it ? isn't already all 0 ? */
   return ast_string_field_init(p, 512);
}

static char* terminate_uri

(

char *

uri

)

[static]

Terminate the uri at the first ';' or space. Technically we should ignore escaped space per RFC3261 (19.1.1 etc) but don't do it for the time being. Remember the uri format is: (User-parameters was added after RFC 3261)

 *
 *	sip:user:password;user-parameters@host:port;uri-parameters?headers
 *	sips:user:password;user-parameters@host:port;uri-parameters?headers
 *
 *

Todo:

As this function does not support user-parameters, it's considered broken and needs fixing.

Definition at line 16857 of file chan_sip.c.

Referenced by check_user_full(), and register_verify().

{
   char *t = uri;
   while (*t && *t > ' ' && *t != ';') {
      t++;
   }
   *t = '\0';
   return uri;
}

static int threadinfo_locate_cb	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 28628 of file chan_sip.c.

References ast_sockaddr_cmp(), CMP_MATCH, and CMP_STOP.

Referenced by sip_tcp_locate().

{
   struct sip_threadinfo *th = obj;
   struct ast_sockaddr *s = arg;

   if (!ast_sockaddr_cmp(s, &th->tcptls_session->remote_address)) {
      return CMP_MATCH | CMP_STOP;
   }

   return 0;
}

static int threadt_cmp_cb	(	void *	obj,
		void *	arg,
		int	flags
	)		[static]

Definition at line 33579 of file chan_sip.c.

References CMP_MATCH, and CMP_STOP.

Referenced by load_module().

{
   struct sip_threadinfo *th = obj, *th2 = arg;

   return (th->tcptls_session == th2->tcptls_session) ? CMP_MATCH | CMP_STOP : 0;
}

static int threadt_hash_cb	(	const void *	obj,
		const int	flags
	)		[static]

Definition at line 33572 of file chan_sip.c.

References ast_sockaddr_hash().

Referenced by load_module().

{
   const struct sip_threadinfo *th = obj;

   return ast_sockaddr_hash(&th->tcptls_session->remote_address);
}

static char * transfermode2str

(

enum transfermodes

mode

)

[static]

Convert transfer mode to text string.

Definition at line 18803 of file chan_sip.c.

Referenced by _sip_show_peer(), peers_data_provider_get(), sip_show_channel(), sip_show_settings(), and sip_show_user().

{
   if (mode == TRANSFER_OPENFORALL)
      return "open";
   else if (mode == TRANSFER_CLOSED)
      return "closed";
   return "strict";
}

static int transmit_cc_notify	(	struct ast_cc_agent *	agent,
		struct sip_pvt *	subscription,
		enum sip_cc_notify_state	state
	)		[static]

Definition at line 14713 of file chan_sip.c.

References add_content(), add_header(), ast_log(), generate_uri(), LOG_WARNING, ast_cc_agent::private_data, reqprep(), send_request(), sip_cc_notify_state_map, state_string, and TRUE.

Referenced by sip_cc_agent_recall(), and sip_cc_agent_respond().

{
   struct sip_request req;
   struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
   char uri[SIPBUFSIZE];
   char state_str[64];
   char subscription_state_hdr[64];

   if (state < CC_QUEUED || state > CC_READY) {
      ast_log(LOG_WARNING, "Invalid state provided for transmit_cc_notify (%d)\n", state);
      return -1;
   }

   reqprep(&req, subscription, SIP_NOTIFY, 0, TRUE);
   snprintf(state_str, sizeof(state_str), "%s\r\n", sip_cc_notify_state_map[state].state_string);
   add_header(&req, "Event", "call-completion");
   add_header(&req, "Content-Type", "application/call-completion");
   snprintf(subscription_state_hdr, sizeof(subscription_state_hdr), "active;expires=%d", subscription->expiry);
   add_header(&req, "Subscription-State", subscription_state_hdr);
   if (state == CC_READY) {
      generate_uri(subscription, agent_pvt->notify_uri, sizeof(agent_pvt->notify_uri));
      snprintf(uri, sizeof(uri) - 1, "cc-URI: %s\r\n", agent_pvt->notify_uri);
   }
   add_content(&req, state_str);
   if (state == CC_READY) {
      add_content(&req, uri);
   }
   return send_request(subscription, &req, XMIT_RELIABLE, subscription->ocseq);
}

static void transmit_fake_auth_response	(	struct sip_pvt *	p,
		struct sip_request *	req,
		enum xmittype	reliable
	)		[static]

Send a fake 401 Unauthorized response when the administrator wants to hide the names of local devices from fishers.

Definition at line 16756 of file chan_sip.c.

References __transmit_response(), AST_DYNSTR_BUILD_FAILED, ast_skip_blanks(), ast_str_set(), ast_str_thread_get(), ast_strlen_zero(), build_nonce(), FALSE, sip_get_header(), sip_scheddestroy(), and transmit_response_with_auth().

Referenced by register_verify().

{
   /* We have to emulate EXACTLY what we'd get with a good peer
    * and a bad password, or else we leak information. */
   const char *response = "401 Unauthorized";
   const char *reqheader = "Authorization";
   const char *respheader = "WWW-Authenticate";
   const char *authtoken;
   struct ast_str *buf;
   char *c;

   /* table of recognised keywords, and their value in the digest */
   enum keys { K_NONCE, K_LAST };
   struct x {
      const char *key;
      const char *s;
   } *i, keys[] = {
      [K_NONCE] = { "nonce=", "" },
      [K_LAST] = { NULL, NULL}
   };

   authtoken = sip_get_header(req, reqheader);
   if (req->ignore && !ast_strlen_zero(p->nonce) && ast_strlen_zero(authtoken)) {
      /* This is a retransmitted invite/register/etc, don't reconstruct authentication
       * information */
      transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
      /* Schedule auto destroy in 32 seconds (according to RFC 3261) */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   } else if (ast_strlen_zero(p->nonce) || ast_strlen_zero(authtoken)) {
      /* We have no auth, so issue challenge and request authentication */
      build_nonce(p, 1);
      transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, 0);
      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
      return;
   }

   if (!(buf = ast_str_thread_get(&check_auth_buf, CHECK_AUTH_BUF_INITLEN))) {
      __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
      return;
   }

   /* Make a copy of the response and parse it */
   if (ast_str_set(&buf, 0, "%s", authtoken) == AST_DYNSTR_BUILD_FAILED) {
      __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
      return;
   }

   c = buf->str;

   while (c && *(c = ast_skip_blanks(c))) { /* lookup for keys */
      for (i = keys; i->key != NULL; i++) {
         const char *separator = ",";  /* default */

         if (strncasecmp(c, i->key, strlen(i->key)) != 0) {
            continue;
         }
         /* Found. Skip keyword, take text in quotes or up to the separator. */
         c += strlen(i->key);
         if (*c == '"') { /* in quotes. Skip first and look for last */
            c++;
            separator = "\"";
         }
         i->s = c;
         strsep(&c, separator);
         break;
      }
      if (i->key == NULL) { /* not found, jump after space or comma */
         strsep(&c, " ,");
      }
   }

   /* Verify nonce from request matches our nonce.  If not, send 401 with new nonce */
   if (strcasecmp(p->nonce, keys[K_NONCE].s)) {
      if (!req->ignore) {
         build_nonce(p, 1);
      }
      transmit_response_with_auth(p, response, req, p->nonce, reliable, respheader, FALSE);

      /* Schedule auto destroy in 32 seconds */
      sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
   } else {
      __transmit_response(p, "403 Forbidden", &p->initreq, reliable);
   }
}

static int transmit_info_with_aoc	(	struct sip_pvt *	p,
		struct ast_aoc_decoded *	decoded
	)		[static]

Send SIP INFO advice of charge message.

Definition at line 15567 of file chan_sip.c.

References add_header(), ast_aoc_unit_entry::amount, AST_AOC_CHARGE_CURRENCY, AST_AOC_CHARGE_FREE, AST_AOC_CHARGE_UNIT, AST_AOC_D, AST_AOC_E, ast_aoc_get_charge_type(), ast_aoc_get_currency_amount(), ast_aoc_get_currency_multiplier_decimal(), ast_aoc_get_currency_name(), ast_aoc_get_msg_type(), ast_aoc_get_unit_info(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_strlen_zero(), reqprep(), send_request(), and str.

Referenced by sip_indicate().

{
   struct sip_request req;
   struct ast_str *str = ast_str_alloca(512);
   const struct ast_aoc_unit_entry *unit_entry = ast_aoc_get_unit_info(decoded, 0);
   enum ast_aoc_charge_type charging = ast_aoc_get_charge_type(decoded);

   reqprep(&req, p, SIP_INFO, 0, 1);

   if (ast_aoc_get_msg_type(decoded) == AST_AOC_D) {
      ast_str_append(&str, 0, "type=active;");
   } else if (ast_aoc_get_msg_type(decoded) == AST_AOC_E) {
      ast_str_append(&str, 0, "type=terminated;");
   } else {
      /* unsupported message type */
      return -1;
   }

   switch (charging) {
   case AST_AOC_CHARGE_FREE:
      ast_str_append(&str, 0, "free-of-charge;");
      break;
   case AST_AOC_CHARGE_CURRENCY:
      ast_str_append(&str, 0, "charging;");
      ast_str_append(&str, 0, "charging-info=currency;");
      ast_str_append(&str, 0, "amount=%u;", ast_aoc_get_currency_amount(decoded));
      ast_str_append(&str, 0, "multiplier=%s;", ast_aoc_get_currency_multiplier_decimal(decoded));
      if (!ast_strlen_zero(ast_aoc_get_currency_name(decoded))) {
         ast_str_append(&str, 0, "currency=%s;", ast_aoc_get_currency_name(decoded));
      }
      break;
   case AST_AOC_CHARGE_UNIT:
      ast_str_append(&str, 0, "charging;");
      ast_str_append(&str, 0, "charging-info=pulse;");
      if (unit_entry) {
         ast_str_append(&str, 0, "recorded-units=%u;", unit_entry->amount);
      }
      break;
   default:
      ast_str_append(&str, 0, "not-available;");
   };

   add_header(&req, "AOC", ast_str_buffer(str));

   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_info_with_digit	(	struct sip_pvt *	p,
		const char	digit,
		unsigned int	duration
	)		[static]

Send SIP INFO dtmf message, see Cisco documentation on cisco.com.

Definition at line 15615 of file chan_sip.c.

References add_digit(), ast_test_flag, reqprep(), and send_request().

Referenced by sip_senddigit_end().

{
   struct sip_request req;
   
   reqprep(&req, p, SIP_INFO, 0, 1);
   add_digit(&req, digit, duration, (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_SHORTINFO));
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_info_with_vidupdate

(

struct sip_pvt *

p

)

[static]

Send SIP INFO with video update request.

Definition at line 15625 of file chan_sip.c.

References add_vidupdate(), reqprep(), and send_request().

Referenced by sip_indicate().

{
   struct sip_request req;
   
   reqprep(&req, p, SIP_INFO, 0, 1);
   add_vidupdate(&req);
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_invite	(	struct sip_pvt *	p,
		int	sipmethod,
		int	sdp,
		int	init,
		const char *const	explicit_uri
	)		[static]

Build REFER/INVITE/OPTIONS/SUBSCRIBE message and transmit it.

Parameters:

p	sip_pvt structure
sipmethod
sdp	unknown
init	0 = Prepare request within dialog, 1= prepare request, new branch, 2= prepare new request and new dialog. do_proxy_auth calls this with init!=2
explicit_uri

Definition at line 14098 of file chan_sip.c.

References add_content(), add_date(), add_diversion(), add_expires(), add_header(), add_rpid(), add_sdp(), add_supported(), ast_channel_lock, ast_channel_name(), ast_channel_unlock, ast_channel_varshead(), ast_debug, AST_LIST_TRAVERSE, ast_log(), ast_random(), ast_skip_blanks(), ast_str_buffer(), ast_str_strlen(), ast_strlen_zero(), ast_test_flag, ast_var_name(), ast_var_value(), build_via(), FALSE, initialize_initreq(), initreqprep(), LOG_WARNING, ast_variable::name, ast_variable::next, offered_media_list_destroy(), reqprep(), send_request(), st_get_mode(), st_get_se(), TRUE, try_suggested_sip_codec(), ast_variable::value, and var.

Referenced by __sip_subscribe_mwi_do(), cc_handle_publish_error(), do_proxy_auth(), manager_sipnotify(), proc_422_rsp(), sip_call(), sip_cc_monitor_request_cc(), sip_cli_notify(), sip_monitor_instance_destructor(), sip_poke_peer(), transmit_publish(), and transmit_refer().

{
   struct sip_request req;
   struct ast_variable *var;

   if (init) {/* Bump branch even on initial requests */
      p->branch ^= ast_random();
      p->invite_branch = p->branch;
      build_via(p);
   }
   if (init > 1) {
      initreqprep(&req, p, sipmethod, explicit_uri);
   } else {
      /* If init=1, we should not generate a new branch. If it's 0, we need a new branch. */
      reqprep(&req, p, sipmethod, 0, init ? 0 : 1);
   }

   if (p->options && p->options->auth) {
      add_header(&req, p->options->authheader, p->options->auth);
   }
   add_date(&req);
   if (sipmethod == SIP_REFER && p->refer) { /* Call transfer */
      if (!ast_strlen_zero(p->refer->refer_to)) {
         add_header(&req, "Refer-To", p->refer->refer_to);
      }
      if (!ast_strlen_zero(p->refer->referred_by)) {
         add_header(&req, "Referred-By", p->refer->referred_by);
      }
   } else if (sipmethod == SIP_SUBSCRIBE) {
      if (p->subscribed == MWI_NOTIFICATION) {
         add_header(&req, "Event", "message-summary");
         add_header(&req, "Accept", "application/simple-message-summary");
      } else if (p->subscribed == CALL_COMPLETION) {
         add_header(&req, "Event", "call-completion");
         add_header(&req, "Accept", "application/call-completion");
      }
      add_expires(&req, p->expiry);
   }

   /* This new INVITE is part of an attended transfer. Make sure that the
   other end knows and replace the current call with this new call */
   if (p->options && !ast_strlen_zero(p->options->replaces)) {
      add_header(&req, "Replaces", p->options->replaces);
      add_header(&req, "Require", "replaces");
   }

   /* Add Session-Timers related headers */
   if (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE
      || (st_get_mode(p, 0) == SESSION_TIMER_MODE_ACCEPT
         && st_get_se(p, FALSE) != DEFAULT_MIN_SE)) {
      char i2astr[10];

      if (!p->stimer->st_interval) {
         p->stimer->st_interval = st_get_se(p, TRUE);
      }

      p->stimer->st_active = TRUE;
      if (st_get_mode(p, 0) == SESSION_TIMER_MODE_ORIGINATE) { 
         snprintf(i2astr, sizeof(i2astr), "%d", p->stimer->st_interval);
         add_header(&req, "Session-Expires", i2astr);
      }

      snprintf(i2astr, sizeof(i2astr), "%d", st_get_se(p, FALSE));
      add_header(&req, "Min-SE", i2astr);
   }

   add_header(&req, "Allow", ALLOWED_METHODS);
   add_supported(p, &req);

   if (p->owner && ((p->options && p->options->addsipheaders)
              || (p->refer && global_refer_addheaders))) {
      struct ast_channel *chan = p->owner; /* The owner channel */
      struct varshead *headp;
   
      ast_channel_lock(chan);

      headp = ast_channel_varshead(chan);

      if (!headp) {
         ast_log(LOG_WARNING, "No Headp for the channel...ooops!\n");
      } else {
         const struct ast_var_t *current;
         AST_LIST_TRAVERSE(headp, current, entries) {
            /* SIPADDHEADER: Add SIP header to outgoing call */
            if (!strncasecmp(ast_var_name(current), "SIPADDHEADER", strlen("SIPADDHEADER"))) {
               char *content, *end;
               const char *header = ast_var_value(current);
               char *headdup = ast_strdupa(header);

               /* Strip of the starting " (if it's there) */
               if (*headdup == '"') {
                  headdup++;
               }
               if ((content = strchr(headdup, ':'))) {
                  *content++ = '\0';
                  content = ast_skip_blanks(content); /* Skip white space */
                  /* Strip the ending " (if it's there) */
                  end = content + strlen(content) -1;
                  if (*end == '"') {
                     *end = '\0';
                  }

                  add_header(&req, headdup, content);
                  if (sipdebug) {
                     ast_debug(1, "Adding SIP Header \"%s\" with content :%s: \n", headdup, content);
                  }
               }
            }
         }
      }

      ast_channel_unlock(chan);
   }
   if ((sipmethod == SIP_INVITE || sipmethod == SIP_UPDATE) && ast_test_flag(&p->flags[0], SIP_SENDRPID))
      add_rpid(&req, p);
   if (sipmethod == SIP_INVITE) {
      add_diversion(&req, p);
   }
   if (sdp) {
      offered_media_list_destroy(p);
      if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
         ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? ast_channel_name(p->owner) : "<none>");
         add_sdp(&req, p, FALSE, FALSE, TRUE);
      } else if (p->rtp) {
         try_suggested_sip_codec(p);
         add_sdp(&req, p, FALSE, TRUE, FALSE);
      }
   } else if (sipmethod == SIP_NOTIFY && p->notify) {
      for (var = p->notify->headers; var; var = var->next) {
         add_header(&req, var->name, var->value);
      }
      if (ast_str_strlen(p->notify->content)) {
         add_content(&req, ast_str_buffer(p->notify->content));
      }
   } else if (sipmethod == SIP_PUBLISH) {
      switch (p->epa_entry->static_data->event) {
      case CALL_COMPLETION:
         add_header(&req, "Event", "call-completion");
         add_expires(&req, p->expiry);
         if (p->epa_entry->publish_type != SIP_PUBLISH_INITIAL) {
            add_header(&req, "SIP-If-Match", p->epa_entry->entity_tag);
         }

         if (!ast_strlen_zero(p->epa_entry->body)) {
            add_header(&req, "Content-Type", "application/pidf+xml");
            add_content(&req, p->epa_entry->body);
         }
      default:
         break;
      }
   }

   if (!p->initreq.headers || init > 2) {
      initialize_initreq(p, &req);
   }
   if (sipmethod == SIP_INVITE || sipmethod == SIP_SUBSCRIBE) {
      p->lastinvite = p->ocseq;
   }
   return send_request(p, &req, init ? XMIT_CRITICAL : XMIT_RELIABLE, p->ocseq);
}

static int transmit_message	(	struct sip_pvt *	p,
		int	init,
		int	auth
	)		[static]

Transmit with SIP MESSAGE method.

Note:

The p->msg_headers and p->msg_body are already setup.

Definition at line 15467 of file chan_sip.c.

References add_text(), initialize_initreq(), initreqprep(), reqprep(), send_request(), and transmit_request_with_auth().

Referenced by do_message_auth(), sip_msg_send(), sip_park_thread(), and sip_sendtext().

{
   struct sip_request req;

   if (init) {
      initreqprep(&req, p, SIP_MESSAGE, NULL);
      initialize_initreq(p, &req);
   } else {
      reqprep(&req, p, SIP_MESSAGE, 0, 1);
   }
   if (auth) {
      return transmit_request_with_auth(p, SIP_MESSAGE, p->ocseq, XMIT_RELIABLE, 0);
   } else {
      add_text(&req, p);
      return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
   }
}

static int transmit_notify_with_mwi	(	struct sip_pvt *	p,
		int	newmsgs,
		int	oldmsgs,
		const char *	vmexten
	)		[static]

Notify user of messages waiting in voicemail (RFC3842)

Note:

- Notification only works for registered peers with mailbox= definitions in sip.conf

• We use the SIP Event package message-summary MIME type defaults to "application/simple-message-summary";

Definition at line 14851 of file chan_sip.c.

References add_content(), add_header(), ast_sockaddr_port, ast_sockaddr_stringify_host_remote(), ast_str_alloca, ast_str_append(), ast_str_buffer(), ast_test_flag, exten, initialize_initreq(), initreqprep(), ourport, S_OR, send_request(), sip_get_transport(), and sip_standard_port().

Referenced by sip_send_mwi_to_peer().

{
   struct sip_request req;
   struct ast_str *out = ast_str_alloca(500);
   int ourport = (p->fromdomainport) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
   const char *domain;
   const char *exten = S_OR(vmexten, default_vmexten);

   initreqprep(&req, p, SIP_NOTIFY, NULL);
   add_header(&req, "Event", "message-summary");
   add_header(&req, "Content-Type", default_notifymime);
   ast_str_append(&out, 0, "Messages-Waiting: %s\r\n", newmsgs ? "yes" : "no");

   /* domain initialization occurs here because initreqprep changes ast_sockaddr_stringify string. */
   domain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));

   if (!sip_standard_port(p->socket.type, ourport)) {
      if (p->socket.type == SIP_TRANSPORT_UDP) {
         ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d\r\n", exten, domain, ourport);
      } else {
         ast_str_append(&out, 0, "Message-Account: sip:%s@%s:%d;transport=%s\r\n", exten, domain, ourport, sip_get_transport(p->socket.type));
      }
   } else {
      if (p->socket.type == SIP_TRANSPORT_UDP) {
         ast_str_append(&out, 0, "Message-Account: sip:%s@%s\r\n", exten, domain);
      } else {
         ast_str_append(&out, 0, "Message-Account: sip:%s@%s;transport=%s\r\n", exten, domain, sip_get_transport(p->socket.type));
      }
   }
   /* Cisco has a bug in the SIP stack where it can't accept the
      (0/0) notification. This can temporarily be disabled in
      sip.conf with the "buggymwi" option */
   ast_str_append(&out, 0, "Voice-Message: %d/%d%s\r\n",
      newmsgs, oldmsgs, (ast_test_flag(&p->flags[1], SIP_PAGE2_BUGGY_MWI) ? "" : " (0/0)"));

   if (p->subscribed) {
      if (p->expiry) {
         add_header(&req, "Subscription-State", "active");
      } else { /* Expired */
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
      }
   }

   add_content(&req, ast_str_buffer(out));

   if (!p->initreq.headers) {
      initialize_initreq(p, &req);
   }
   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_notify_with_sipfrag	(	struct sip_pvt *	p,
		int	cseq,
		char *	message,
		int	terminate
	)		[static]

Notify a transferring party of the status of transfer (RFC3515)

Definition at line 14903 of file chan_sip.c.

References add_content(), add_header(), add_supported(), initialize_initreq(), reqprep(), and send_request().

Referenced by handle_request_refer(), local_attended_transfer(), and sip_park_thread().

{
   struct sip_request req;
   char tmp[SIPBUFSIZE/2];
   
   reqprep(&req, p, SIP_NOTIFY, 0, 1);
   snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
   add_header(&req, "Event", tmp);
   add_header(&req, "Subscription-state", terminate ? "terminated;reason=noresource" : "active");
   add_header(&req, "Content-Type", "message/sipfrag;version=2.0");
   add_header(&req, "Allow", ALLOWED_METHODS);
   add_supported(p, &req);

   snprintf(tmp, sizeof(tmp), "SIP/2.0 %s\r\n", message);
   add_content(&req, tmp);

   if (!p->initreq.headers) {
      initialize_initreq(p, &req);
   }

   return send_request(p, &req, XMIT_RELIABLE, p->ocseq);
}

static int transmit_provisional_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		int	with_sdp
	)		[static]

Definition at line 12381 of file chan_sip.c.

References FALSE, transmit_response(), transmit_response_with_sdp(), and update_provisional_keepalive().

Referenced by handle_request_invite(), sip_indicate(), and sip_write().

{
   int res;

   if (!(res = with_sdp ? transmit_response_with_sdp(p, msg, req, XMIT_UNRELIABLE, FALSE, FALSE) : transmit_response(p, msg, req))) {
      p->last_provisional = msg;
      update_provisional_keepalive(p, with_sdp);
   }

   return res;
}

static int transmit_publish	(	struct sip_epa_entry *	epa_entry,
		enum sip_publish_type	publish_type,
		const char *const	explicit_uri
	)		[static]

Definition at line 14054 of file chan_sip.c.

References ao2_ref, ast_set_flag, ast_sip_ouraddrfor(), create_addr(), dialog_unlink_all(), FALSE, sip_alloc(), sip_pvt_lock, sip_pvt_unlock, sip_scheddestroy(), transmit_invite(), and TRUE.

Referenced by handle_cc_notify(), sip_cc_monitor_suspend(), sip_cc_monitor_unsuspend(), and sip_monitor_instance_destructor().

{
   struct sip_pvt *pvt;
   int expires;

   epa_entry->publish_type = publish_type;

   if (!(pvt = sip_alloc(NULL, NULL, 0, SIP_PUBLISH, NULL, NULL))) {
      return -1;
   }

   sip_pvt_lock(pvt);

   if (create_addr(pvt, epa_entry->destination, NULL, TRUE)) {
      sip_pvt_unlock(pvt);
      dialog_unlink_all(pvt);
      dialog_unref(pvt, "create_addr failed in transmit_publish. Unref dialog");
      return -1;
   }
   ast_sip_ouraddrfor(&pvt->sa, &pvt->ourip, pvt);
   ast_set_flag(&pvt->flags[0], SIP_OUTGOING);
   expires = (publish_type == SIP_PUBLISH_REMOVE) ? 0 : DEFAULT_PUBLISH_EXPIRES;
   pvt->expiry = expires;

   /* Bump refcount for sip_pvt's reference */
   ao2_ref(epa_entry, +1);
   pvt->epa_entry = epa_entry;

   transmit_invite(pvt, SIP_PUBLISH, FALSE, 2, explicit_uri);
   sip_pvt_unlock(pvt);
   sip_scheddestroy(pvt, DEFAULT_TRANS_TIMEOUT);
   dialog_unref(pvt, "Done with the sip_pvt allocated for transmitting PUBLISH");
   return 0;
}

static int transmit_refer	(	struct sip_pvt *	p,
		const char *	dest
	)		[static]

Transmit SIP REFER message (initiated by the transfer() dialplan application.

Note:

this is currently broken as we have no way of telling the dialplan engine whether a transfer succeeds or fails.

Todo:

Fix the transfer() dialplan function so that a transfer may fail

Todo:

In theory, we should hang around and wait for a reply, before returning to the dial plan here. Don't know really how that would affect the transfer() app or the pbx, but, well, to make this useful we should have a STATUS code on transfer().

Definition at line 15507 of file chan_sip.c.

References ast_copy_string(), ast_debug, ast_log(), ast_string_field_set, ast_test_flag, FALSE, get_in_brackets(), LOG_NOTICE, sip_get_header(), sip_refer_alloc(), transmit_invite(), and TRUE.

Referenced by sip_transfer().

{
   char from[256];
   const char *of;
   char *c;
   char referto[256];
   int   use_tls=FALSE;

   if (sipdebug) {
      ast_debug(1, "SIP transfer of %s to %s\n", p->callid, dest);
   }

   /* Are we transfering an inbound or outbound call ? */
   if (ast_test_flag(&p->flags[0], SIP_OUTGOING))  {
      of = sip_get_header(&p->initreq, "To");
   } else {
      of = sip_get_header(&p->initreq, "From");
   }

   ast_copy_string(from, of, sizeof(from));
   of = get_in_brackets(from);
   ast_string_field_set(p, from, of);
   if (!strncasecmp(of, "sip:", 4)) {
      of += 4;
   } else if (!strncasecmp(of, "sips:", 5)) {
      of += 5;
      use_tls = TRUE;
   } else {
      ast_log(LOG_NOTICE, "From address missing 'sip(s):', assuming sip:\n");
   }
   /* Get just the username part */
   if (strchr(dest, '@')) {
      c = NULL;
   } else if ((c = strchr(of, '@'))) {
      *c++ = '\0';
   }
   if (c) {
      snprintf(referto, sizeof(referto), "<sip%s:%s@%s>", use_tls ? "s" : "", dest, c);
   } else {
      snprintf(referto, sizeof(referto), "<sip%s:%s>", use_tls ? "s" : "", dest);
   }

   /* save in case we get 407 challenge */
   sip_refer_alloc(p);
   ast_string_field_set(p->refer, refer_to, referto);
   ast_string_field_set(p->refer, referred_by, p->our_contact);
   p->refer->status = REFER_SENT;   /* Set refer status */

   return transmit_invite(p, SIP_REFER, FALSE, 0, NULL);
   /* We should propably wait for a NOTIFY here until we ack the transfer */
   /* Maybe fork a new thread and wait for a STATUS of REFER_200OK on the refer status before returning to app_transfer */

   /*! \todo In theory, we should hang around and wait for a reply, before
   returning to the dial plan here. Don't know really how that would
   affect the transfer() app or the pbx, but, well, to make this
   useful we should have a STATUS code on transfer().
   */
}

static int transmit_register	(	struct sip_registry *	r,
		int	sipmethod,
		const char *	auth,
		const char *	authheader
	)		[static]

Transmit register to SIP proxy or UA auth = NULL on the initial registration (from sip_reregister())

Definition at line 15205 of file chan_sip.c.

References add_expires(), add_header(), add_max_forwards(), append_history, ast_debug, ast_dnsmgr_lookup_cb(), ast_log(), ast_random(), ast_sched_add(), AST_SCHED_REPLACE_UNREF, ast_set_flag, ast_sip_ouraddrfor(), ast_sockaddr_cmp(), ast_sockaddr_isnull(), ast_sockaddr_port, ast_sockaddr_set_port, ast_string_field_set, ast_strlen_zero(), ast_verbose(), build_callid_registry(), build_contact(), build_localtag_registry(), build_reply_digest(), build_via(), create_addr(), dialog_unlink_all(), exten, FALSE, get_address_family_filter(), get_srv_protocol(), get_srv_service(), init_req(), initialize_initreq(), internip, ast_tcptls_session_args::local_address, LOG_NOTICE, LOG_WARNING, obproxy_get(), on_dns_update_peer(), on_dns_update_registry(), REG_STATE_AUTHSENT, REG_STATE_REGSENT, registry_addref(), registry_unref(), S_OR, send_request(), set_socket_transport(), sip_alloc(), sip_cfg, sip_debug_test_pvt(), sip_find_peer(), sip_methods, sip_reg_timeout(), sip_sanitized_host(), sip_unref_peer(), cfsip_methods::text, and TRUE.

Referenced by __sip_do_register(), do_register_auth(), handle_response_register(), and sip_reg_timeout().

{
   struct sip_request req;
   char from[256];
   char to[256];
   char tmp[80];
   char addr[80];
   struct sip_pvt *p;
   struct sip_peer *peer = NULL;
   int res;
   int portno = 0;

   /* exit if we are already in process with this registrar ?*/
   if (r == NULL || ((auth == NULL) && (r->regstate == REG_STATE_REGSENT || r->regstate == REG_STATE_AUTHSENT))) {
      if (r) {
         ast_log(LOG_NOTICE, "Strange, trying to register %s@%s when registration already pending\n", r->username, r->hostname);
      }
      return 0;
   }

   if (r->dnsmgr == NULL) {
      char transport[MAXHOSTNAMELEN];
      peer = sip_find_peer(r->hostname, NULL, TRUE, FINDPEERS, FALSE, 0);
      snprintf(transport, sizeof(transport), "_%s._%s",get_srv_service(r->transport), get_srv_protocol(r->transport)); /* have to use static sip_get_transport function */
      r->us.ss.ss_family = get_address_family_filter(r->transport); /* Filter address family */

      /* No point in doing a DNS lookup of the register hostname if we're just going to
       * end up using an outbound proxy. obproxy_get is safe to call with either of r->call
       * or peer NULL. Since we're only concerned with its existence, we're not going to
       * bother getting a ref to the proxy*/
      if (!obproxy_get(r->call, peer)) {
         registry_addref(r, "add reg ref for dnsmgr");
         ast_dnsmgr_lookup_cb(peer ? peer->tohost : r->hostname, &r->us, &r->dnsmgr, sip_cfg.srvlookup ? transport : NULL, on_dns_update_registry, r);
         if (!r->dnsmgr) {
            /*dnsmgr refresh disabled, no reference added! */
            registry_unref(r, "remove reg ref, dnsmgr disabled");
         }
      }
      if (peer) {
         peer = sip_unref_peer(peer, "removing peer ref for dnsmgr_lookup");
      }
   }

   if (r->call) { /* We have a registration */
      if (!auth) {
         ast_log(LOG_WARNING, "Already have a REGISTER going on to %s@%s?? \n", r->username, r->hostname);
         return 0;
      } else {
         p = dialog_ref(r->call, "getting a copy of the r->call dialog in transmit_register");
         ast_string_field_set(p, theirtag, NULL);  /* forget their old tag, so we don't match tags when getting response */
      }
   } else {
      /* Build callid for registration if we haven't registered before */
      if (!r->callid_valid) {
         build_callid_registry(r, &internip, default_fromdomain);
         build_localtag_registry(r);
         r->callid_valid = TRUE;
      }
      /* Allocate SIP dialog for registration */
      if (!(p = sip_alloc( r->callid, NULL, 0, SIP_REGISTER, NULL, NULL))) {
         ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
         return 0;
      }

      /* reset tag to consistent value from registry */
      ast_string_field_set(p, tag, r->localtag);

      if (p->do_history) {
         append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
      }

      p->socket.type = r->transport;

      /* Use port number specified if no SRV record was found */
      if (!ast_sockaddr_isnull(&r->us)) {
         if (!ast_sockaddr_port(&r->us) && r->portno) {
            ast_sockaddr_set_port(&r->us, r->portno);
         }

         /* It is possible that DNS was unavailable at the time the peer was created.
          * Here, if we've updated the address in the registry via manually calling
          * ast_dnsmgr_lookup_cb() above, then we call the same function that dnsmgr would
          * call if it was updating a peer's address */
         if ((peer = sip_find_peer(S_OR(r->peername, r->hostname), NULL, TRUE, FINDPEERS, FALSE, 0))) {
            if (ast_sockaddr_cmp(&peer->addr, &r->us)) {
               on_dns_update_peer(&peer->addr, &r->us, peer);
            }
            peer = sip_unref_peer(peer, "unref after sip_find_peer");
         }
      }

      /* Find address to hostname */
      if (create_addr(p, S_OR(r->peername, r->hostname), &r->us, 0)) {
         /* we have what we hope is a temporary network error,
          * probably DNS.  We need to reschedule a registration try */
         dialog_unlink_all(p);
         p = dialog_unref(p, "unref dialog after unlink_all");
         if (r->timeout > -1) {
            AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
                              registry_unref(_data, "del for REPLACE of registry ptr"),
                              registry_unref(r, "object ptr dec when SCHED_REPLACE add failed"),
                              registry_addref(r,"add for REPLACE registry ptr"));
            ast_log(LOG_WARNING, "Still have a registration timeout for %s@%s (create_addr() error), %d\n", r->username, r->hostname, r->timeout);
         } else {
            r->timeout = ast_sched_add(sched, global_reg_timeout * 1000, sip_reg_timeout, registry_addref(r, "add for REPLACE registry ptr"));
            ast_log(LOG_WARNING, "Probably a DNS error for registration to %s@%s, trying REGISTER again (after %d seconds)\n", r->username, r->hostname, global_reg_timeout);
         }
         r->regattempts++;
         return 0;
      }

      /* Copy back Call-ID in case create_addr changed it */
      ast_string_field_set(r, callid, p->callid);

      if (!r->dnsmgr && r->portno) {
         ast_sockaddr_set_port(&p->sa, r->portno);
         ast_sockaddr_set_port(&p->recv, r->portno);
      }
      if (!ast_strlen_zero(p->fromdomain)) {
         portno = (p->fromdomainport) ? p->fromdomainport : STANDARD_SIP_PORT;
      } else if (!ast_strlen_zero(r->regdomain)) {
         portno = (r->regdomainport) ? r->regdomainport : STANDARD_SIP_PORT;
      } else {
         portno = ast_sockaddr_port(&p->sa);
      }

      ast_set_flag(&p->flags[0], SIP_OUTGOING); /* Registration is outgoing call */
      r->call = dialog_ref(p, "copying dialog into registry r->call");     /* Save pointer to SIP dialog */
      p->registry = registry_addref(r, "transmit_register: addref to p->registry in transmit_register"); /* Add pointer to registry in packet */
      if (!ast_strlen_zero(r->secret)) {  /* Secret (password) */
         ast_string_field_set(p, peersecret, r->secret);
      }
      if (!ast_strlen_zero(r->md5secret))
         ast_string_field_set(p, peermd5secret, r->md5secret);
      /* User name in this realm
      - if authuser is set, use that, otherwise use username */
      if (!ast_strlen_zero(r->authuser)) {
         ast_string_field_set(p, peername, r->authuser);
         ast_string_field_set(p, authname, r->authuser);
      } else if (!ast_strlen_zero(r->username)) {
         ast_string_field_set(p, peername, r->username);
         ast_string_field_set(p, authname, r->username);
         ast_string_field_set(p, fromuser, r->username);
      }
      if (!ast_strlen_zero(r->username)) {
         ast_string_field_set(p, username, r->username);
      }
      /* Save extension in packet */
      if (!ast_strlen_zero(r->callback)) {
         ast_string_field_set(p, exten, r->callback);
      }

      /* Set transport and port so the correct contact is built */
      set_socket_transport(&p->socket, r->transport);
      if (r->transport == SIP_TRANSPORT_TLS || r->transport == SIP_TRANSPORT_TCP) {
         p->socket.port =
             htons(ast_sockaddr_port(&sip_tcp_desc.local_address));
      }

      /*
        check which address we should use in our contact header
        based on whether the remote host is on the external or
        internal network so we can register through nat
       */
      ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
      build_contact(p);
   }

   /* set up a timeout */
   if (auth == NULL)  {
      if (r->timeout > -1) {
         ast_log(LOG_WARNING, "Still have a registration timeout, #%d - deleting it\n", r->timeout);
      }
      AST_SCHED_REPLACE_UNREF(r->timeout, sched, global_reg_timeout * 1000, sip_reg_timeout, r,
                        registry_unref(_data,"reg ptr unrefed from del in SCHED_REPLACE"),
                        registry_unref(r,"reg ptr unrefed from add failure in SCHED_REPLACE"),
                        registry_addref(r,"reg ptr reffed from add in SCHED_REPLACE"));
      ast_debug(1, "Scheduled a registration timeout for %s id  #%d \n", r->hostname, r->timeout);
   }

   snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)), p->tag);
   if (!ast_strlen_zero(p->theirtag)) {
      snprintf(to, sizeof(to), "<sip:%s@%s>;tag=%s", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)), p->theirtag);
   } else {
      snprintf(to, sizeof(to), "<sip:%s@%s>", r->username, S_OR(r->regdomain, sip_sanitized_host(p->tohost)));
   }

   /* Fromdomain is what we are registering to, regardless of actual
      host name from SRV */
   if (portno && portno != STANDARD_SIP_PORT) {
      snprintf(addr, sizeof(addr), "sip:%s:%d", S_OR(p->fromdomain,S_OR(r->regdomain, sip_sanitized_host(r->hostname))), portno);
   } else {
      snprintf(addr, sizeof(addr), "sip:%s", S_OR(p->fromdomain,S_OR(r->regdomain, sip_sanitized_host(r->hostname))));
   }

   ast_string_field_set(p, uri, addr);

   p->branch ^= ast_random();

   init_req(&req, sipmethod, addr);

   /* Add to CSEQ */
   snprintf(tmp, sizeof(tmp), "%u %s", ++r->ocseq, sip_methods[sipmethod].text);
   p->ocseq = r->ocseq;

   build_via(p);
   add_header(&req, "Via", p->via);
   add_max_forwards(p, &req);
   add_header(&req, "From", from);
   add_header(&req, "To", to);
   add_header(&req, "Call-ID", p->callid);
   add_header(&req, "CSeq", tmp);
   if (!ast_strlen_zero(global_useragent))
      add_header(&req, "User-Agent", global_useragent);

   if (auth) { /* Add auth header */
      add_header(&req, authheader, auth);
   } else if (!ast_strlen_zero(r->nonce)) {
      char digest[1024];

      /* We have auth data to reuse, build a digest header.
       * Note, this is not always useful because some parties do not
       * like nonces to be reused (for good reasons!) so they will
       * challenge us anyways.
       */
      if (sipdebug) {
         ast_debug(1, "   >>> Re-using Auth data for %s@%s\n", r->username, r->hostname);
      }
      ast_string_field_set(p, realm, r->realm);
      ast_string_field_set(p, nonce, r->nonce);
      ast_string_field_set(p, domain, r->authdomain);
      ast_string_field_set(p, opaque, r->opaque);
      ast_string_field_set(p, qop, r->qop);
      p->noncecount = ++r->noncecount;

      memset(digest, 0, sizeof(digest));
      if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
         add_header(&req, "Authorization", digest);
      } else {
         ast_log(LOG_NOTICE, "No authorization available for authentication of registration to %s@%s\n", r->username, r->hostname);
      }
   }

   add_expires(&req, r->expiry);
   add_header(&req, "Contact", p->our_contact);

   initialize_initreq(p, &req);
   if (sip_debug_test_pvt(p)) {
      ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
   }
   r->regstate = auth ? REG_STATE_AUTHSENT : REG_STATE_REGSENT;
   r->regattempts++; /* Another attempt */
   ast_debug(4, "REGISTER attempt %d to %s@%s\n", r->regattempts, r->username, r->hostname);
   res = send_request(p, &req, XMIT_CRITICAL, p->ocseq);
   dialog_unref(p, "p is finished here at the end of transmit_register");
   return res;
}

static int transmit_reinvite_with_sdp	(	struct sip_pvt *	p,
		int	t38version,
		int	oldsdp
	)		[static]

Transmit reinvite with SDP.

Note:

A re-invite is basically a new INVITE with the same CALL-ID and TAG as the INVITE that opened the SIP dialogue We reinvite so that the audio stream (RTP) go directly between the SIP UAs. SIP Signalling stays with * in the path.

If t38version is TRUE, we send T38 SDP for re-invite from audio/video to T38 UDPTL transmission on the channel

If oldsdp is TRUE then the SDP version number is not incremented. This is needed for Session-Timers so we can send a re-invite to refresh the SIP session without modifying the media session.

Definition at line 13723 of file chan_sip.c.

References add_header(), add_rpid(), add_sdp(), add_supported(), append_history, ast_set_flag, ast_test_flag, FALSE, initialize_initreq(), offered_media_list_destroy(), reqprep(), send_request(), TRUE, and try_suggested_sip_codec().

Referenced by check_pendings(), handle_response_invite(), interpret_t38_parameters(), proc_session_timer(), sip_sendhtml(), sip_set_rtp_peer(), and sip_set_udptl_peer().

{
   struct sip_request req;
   
   reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ?  SIP_UPDATE : SIP_INVITE, 0, 1);

   add_header(&req, "Allow", ALLOWED_METHODS);
   add_supported(p, &req);
   if (sipdebug) {
      if (oldsdp == TRUE)
         add_header(&req, "X-asterisk-Info", "SIP re-invite (Session-Timers)");
      else
         add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
   }

   if (ast_test_flag(&p->flags[0], SIP_SENDRPID))
      add_rpid(&req, p);

   if (p->do_history) {
      append_history(p, "ReInv", "Re-invite sent");
   }

   offered_media_list_destroy(p);

   try_suggested_sip_codec(p);
   if (t38version) {
      add_sdp(&req, p, oldsdp, FALSE, TRUE);
   } else {
      add_sdp(&req, p, oldsdp, TRUE, FALSE);
   }

   /* Use this as the basis */
   initialize_initreq(p, &req);
   p->lastinvite = p->ocseq;
   ast_set_flag(&p->flags[0], SIP_OUTGOING);       /* Change direction of this dialog */
   p->ongoing_reinvite = 1;
   return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}

static int transmit_request	(	struct sip_pvt *	p,
		int	sipmethod,
		uint32_t	seqno,
		enum xmittype	reliable,
		int	newbranch
	)		[static]

Transmit generic SIP request returns XMIT_ERROR if transmit failed with a critical error (don't retry)

Definition at line 15637 of file chan_sip.c.

References add_header(), reqprep(), and send_request().

Referenced by check_pendings(), forked_invite_init(), handle_response(), handle_response_invite(), and sip_hangup().

{
   struct sip_request resp;
   
   reqprep(&resp, p, sipmethod, seqno, newbranch);
   if (sipmethod == SIP_CANCEL && p->answered_elsewhere) {
      add_header(&resp, "Reason", "SIP;cause=200;text=\"Call completed elsewhere\"");
   }

   if (sipmethod == SIP_ACK) {
      p->invitestate = INV_CONFIRMED;
   }

   return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);
}

static int transmit_request_with_auth	(	struct sip_pvt *	p,
		int	sipmethod,
		uint32_t	seqno,
		enum xmittype	reliable,
		int	newbranch
	)		[static]

Transmit SIP request, auth added.

Definition at line 15669 of file chan_sip.c.

References add_header(), add_text(), ast_cause2str(), ast_log(), ast_strlen_zero(), ast_test_flag, build_reply_digest(), dummy(), LOG_WARNING, reqprep(), send_request(), and sip_auth_headers().

Referenced by __sip_autodestruct(), check_pendings(), sip_hangup(), and transmit_message().

{
   struct sip_request resp;
   
   reqprep(&resp, p, sipmethod, seqno, newbranch);
   if (!ast_strlen_zero(p->realm)) {
      char digest[1024];

      memset(digest, 0, sizeof(digest));
      if(!build_reply_digest(p, sipmethod, digest, sizeof(digest))) {
         char *dummy, *response;
         enum sip_auth_type code = p->options ? p->options->auth_type : PROXY_AUTH; /* XXX force 407 if unknown */
         sip_auth_headers(code, &dummy, &response);
         add_header(&resp, response, digest);
      } else {
         ast_log(LOG_WARNING, "No authentication available for call %s\n", p->callid);
      }
   }

   switch (sipmethod) {
   case SIP_BYE:
   {
      char buf[20];

      /*
       * We are hanging up.  If we know a cause for that, send it in
       * clear text to make debugging easier.
       */
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_Q850_REASON) && p->hangupcause) {
         snprintf(buf, sizeof(buf), "Q.850;cause=%d", p->hangupcause & 0x7f);
         add_header(&resp, "Reason", buf);
      }

      add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->hangupcause));
      snprintf(buf, sizeof(buf), "%d", p->hangupcause);
      add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
      break;
   }
   case SIP_MESSAGE:
      add_text(&resp, p);
      break;
   default:
      break;
   }

   return send_request(p, &resp, reliable, seqno ? seqno : p->ocseq);   
}

static int transmit_response	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)		[static]

Transmit response, no retransmits.

Definition at line 12169 of file chan_sip.c.

References __transmit_response().

Referenced by cc_esc_publish_handler(), handle_cc_notify(), handle_cc_subscribe(), handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_info(), handle_request_invite(), handle_request_message(), handle_request_notify(), handle_request_options(), handle_request_publish(), handle_request_refer(), handle_request_subscribe(), handle_request_update(), handle_sip_publish_initial(), handle_sip_publish_modify(), handle_sip_publish_refresh(), handle_sip_publish_remove(), receive_message(), register_verify(), send_provisional_keepalive_full(), sip_cc_agent_destructor(), sip_cc_agent_respond(), sip_indicate(), sip_sendhtml(), and transmit_provisional_response().

{
   return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}

static int transmit_response_reliable	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)		[static]

Transmit response, Make sure you get an ACK This is only used for responses to INVITEs, where we need to make sure we get an ACK.

Definition at line 12202 of file chan_sip.c.

References __transmit_response().

Referenced by handle_incoming(), handle_invite_replaces(), handle_request_bye(), handle_request_cancel(), handle_request_invite(), handle_request_invite_st(), interpret_t38_parameters(), sip_hangup(), sip_indicate(), sip_sipredirect(), and sip_t38_abort().

{
   return __transmit_response(p, msg, req, req->ignore ? XMIT_UNRELIABLE : XMIT_CRITICAL);
}

static int transmit_response_using_temp	(	ast_string_field	callid,
		struct ast_sockaddr *	addr,
		int	useglobal_nat,
		const int	intended_method,
		const struct sip_request *	req,
		const char *	msg
	)		[static]

Transmit response, no retransmits, using a temporary pvt structure.

Definition at line 12115 of file chan_sip.c.

References __transmit_response(), ast_copy_flags, ast_log(), ast_random(), ast_sip_ouraddrfor(), ast_sockaddr_copy(), ast_string_field_init, ast_string_field_set, ast_threadstorage_get(), build_via(), check_via(), copy_socket_data(), default_fromdomainport, internip, LOG_ERROR, make_our_tag(), and ts_temp_pvt.

Referenced by find_call().

{
   struct sip_pvt *p = NULL;

   if (!(p = ast_threadstorage_get(&ts_temp_pvt, sizeof(*p)))) {
      ast_log(LOG_ERROR, "Failed to get temporary pvt\n");
      return -1;
   }

   /* XXX the structure may be dirty from previous usage.
    * Here we should state clearly how we should reinitialize it
    * before using it.
    * E.g. certainly the threadstorage should be left alone,
    * but other thihngs such as flags etc. maybe need cleanup ?
    */

   /* Initialize the bare minimum */
   p->method = intended_method;

   if (!addr) {
      ast_sockaddr_copy(&p->ourip, &internip);
   } else {
      ast_sockaddr_copy(&p->sa, addr);
      ast_sip_ouraddrfor(&p->sa, &p->ourip, p);
   }

   p->branch = ast_random();
   make_our_tag(p);
   p->ocseq = INITIAL_CSEQ;

   if (useglobal_nat && addr) {
      ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT_FORCE_RPORT);
      ast_copy_flags(&p->flags[2], &global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT);
      ast_sockaddr_copy(&p->recv, addr);
      check_via(p, req);
   }

   ast_string_field_set(p, fromdomain, default_fromdomain);
   p->fromdomainport = default_fromdomainport;
   build_via(p);
   ast_string_field_set(p, callid, callid);

   copy_socket_data(&p->socket, &req->socket);

   /* Use this temporary pvt structure to send the message */
   __transmit_response(p, msg, req, XMIT_UNRELIABLE);

   /* Free the string fields, but not the pool space */
   ast_string_field_init(p, 0);

   return 0;
}

static int transmit_response_with_allow	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable
	)		[static]

Append Accept header, content length before transmitting response.

Definition at line 12247 of file chan_sip.c.

References add_header(), respprep(), and send_response().

Referenced by handle_incoming(), and handle_request_options().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   add_header(&resp, "Accept", "application/sdp");
   return send_response(p, &resp, reliable, 0);
}

static int transmit_response_with_auth	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		const char *	rand,
		enum xmittype	reliable,
		const char *	header,
		int	stale
	)		[static]

Respond with authorization request.

Definition at line 12268 of file chan_sip.c.

References add_header(), append_history, ast_log(), get_realm(), LOG_WARNING, respprep(), send_response(), and sip_get_header().

Referenced by check_auth(), and transmit_fake_auth_response().

{
   struct sip_request resp;
   char tmp[512];
   uint32_t seqno = 0;

   if (reliable && (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1)) {
      ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", sip_get_header(req, "CSeq"));
      return -1;
   }
   /* Choose Realm */
   get_realm(p, req);

   /* Stale means that they sent us correct authentication, but
      based it on an old challenge (nonce) */
   snprintf(tmp, sizeof(tmp), "Digest algorithm=MD5, realm=\"%s\", nonce=\"%s\"%s", p->realm, nonce, stale ? ", stale=true" : "");
   respprep(&resp, p, msg, req);
   add_header(&resp, header, tmp);
   append_history(p, "AuthChal", "Auth challenge sent for %s - nc %d", p->username, p->noncecount);
   return send_response(p, &resp, reliable, seqno);
}

static int transmit_response_with_date	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req
	)		[static]

Add date before transmitting response.

Definition at line 12238 of file chan_sip.c.

References add_date(), respprep(), and send_response().

Referenced by handle_response_subscribe(), and register_verify().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   add_date(&resp);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_minexpires	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		int	minexpires
	)		[static]

Append Min-Expires header, content length before transmitting response.

Definition at line 12256 of file chan_sip.c.

References add_header(), respprep(), and send_response().

Referenced by handle_request_publish(), and handle_request_subscribe().

{
   struct sip_request resp;
   char tmp[32];

   snprintf(tmp, sizeof(tmp), "%d", minexpires);
   respprep(&resp, p, msg, req);
   add_header(&resp, "Min-Expires", tmp);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_minse	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		int	minse_int
	)		[static]

Transmit 422 response with Min-SE header (Session-Timers)

Definition at line 12185 of file chan_sip.c.

References add_date(), add_header(), respprep(), and send_response().

Referenced by handle_request_invite_st().

{
   struct sip_request resp;
   char minse_str[20];

   respprep(&resp, p, msg, req);
   add_date(&resp);

   snprintf(minse_str, sizeof(minse_str), "%d", minse_int);
   add_header(&resp, "Min-SE", minse_str);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_retry_after	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		const char *	seconds
	)		[static]

Append Retry-After header field when transmitting response.

Definition at line 12229 of file chan_sip.c.

References add_header(), respprep(), and send_response().

Referenced by handle_incoming().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   add_header(&resp, "Retry-After", seconds);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_response_with_sdp	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		enum xmittype	reliable,
		int	oldsdp,
		int	rpid
	)		[static]

Used for 200 OK and 183 early media.

Returns:

Will return XMIT_ERROR for network errors.

Definition at line 13632 of file chan_sip.c.

References add_cc_call_info_to_response(), add_required_respheader(), add_rpid(), add_sdp(), ast_debug, ast_log(), ast_rtp_codecs_packetization_set(), ast_rtp_instance_activate(), ast_rtp_instance_get_codecs(), ast_test_flag, FALSE, LOG_ERROR, LOG_WARNING, respprep(), send_response(), sip_get_header(), TRUE, and try_suggested_sip_codec().

Referenced by handle_invite_replaces(), handle_request_invite(), send_provisional_keepalive_full(), sip_answer(), and transmit_provisional_response().

{
   struct sip_request resp;
   uint32_t seqno;
   if (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1) {
      ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", sip_get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   if (rpid == TRUE) {
      add_rpid(&resp, p);
   }
   if (ast_test_flag(&p->flags[0], SIP_OFFER_CC)) {
      add_cc_call_info_to_response(p, &resp);
   }
   if (p->rtp) {
      if (!p->autoframing && !ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
         ast_debug(1, "Setting framing from config on incoming call\n");
         ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
      }
      ast_rtp_instance_activate(p->rtp);
      try_suggested_sip_codec(p);
      if (p->t38.state == T38_ENABLED) {
         add_sdp(&resp, p, oldsdp, TRUE, TRUE);
      } else {
         add_sdp(&resp, p, oldsdp, TRUE, FALSE);
      }
   } else
      ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
   if (reliable && !p->pendinginvite)
      p->pendinginvite = seqno;     /* Buggy clients sends ACK on RINGING too */
   add_required_respheader(&resp);
   return send_response(p, &resp, reliable, seqno);
}

static int transmit_response_with_sip_etag	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		struct sip_esc_entry *	esc_entry,
		int	need_new_etag
	)		[static]

Definition at line 12084 of file chan_sip.c.

References add_header(), create_new_sip_etag(), respprep(), and send_response().

Referenced by handle_sip_publish_initial(), handle_sip_publish_modify(), handle_sip_publish_refresh(), and handle_sip_publish_remove().

{
   struct sip_request resp;

   if (need_new_etag) {
      create_new_sip_etag(esc_entry, 1);
   }
   respprep(&resp, p, msg, req);
   add_header(&resp, "SIP-ETag", esc_entry->entity_tag);

   return send_response(p, &resp, 0, 0);
}

static int transmit_response_with_t38_sdp	(	struct sip_pvt *	p,
		char *	msg,
		struct sip_request *	req,
		int	retrans
	)		[static]

Used for 200 OK and 183 early media.

Definition at line 13548 of file chan_sip.c.

References add_sdp(), ast_log(), LOG_ERROR, LOG_WARNING, respprep(), send_response(), and sip_get_header().

Referenced by handle_request_invite(), and interpret_t38_parameters().

{
   struct sip_request resp;
   uint32_t seqno;

   if (sscanf(sip_get_header(req, "CSeq"), "%30u ", &seqno) != 1) {
      ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", sip_get_header(req, "CSeq"));
      return -1;
   }
   respprep(&resp, p, msg, req);
   if (p->udptl) {
      add_sdp(&resp, p, 0, 0, 1);
   } else
      ast_log(LOG_ERROR, "Can't add SDP to response, since we have no UDPTL session allocated. Call-ID %s\n", p->callid);
   if (retrans && !p->pendinginvite)
      p->pendinginvite = seqno;     /* Buggy clients sends ACK on RINGING too */
   return send_response(p, &resp, retrans, seqno);
}

static int transmit_response_with_unsupported	(	struct sip_pvt *	p,
		const char *	msg,
		const struct sip_request *	req,
		const char *	unsupported
	)		[static]

Transmit response, no retransmits.

Definition at line 12175 of file chan_sip.c.

References add_date(), add_header(), respprep(), and send_response().

Referenced by handle_request_bye(), handle_request_invite(), and handle_request_invite_st().

{
   struct sip_request resp;
   respprep(&resp, p, msg, req);
   add_date(&resp);
   add_header(&resp, "Unsupported", unsupported);
   return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

static int transmit_state_notify	(	struct sip_pvt *	p,
		struct state_notify_data *	data,
		int	full,
		int	timeout
	)		[static]

Used in the SUBSCRIBE notification subsystem (RFC3265)

Definition at line 14744 of file chan_sip.c.

References add_content(), add_header(), ast_copy_string(), AST_EXTENSION_DEACTIVATED, AST_EXTENSION_REMOVED, ast_log(), ast_str_alloca, ast_str_buffer(), ast_test_flag, cfsubscription_types::event, find_subscription_type(), get_in_brackets(), LOG_WARNING, cfsubscription_types::mediatype, NONE, remove_uri_parameters(), reqprep(), send_request(), sip_get_header(), state_notify_data::state, and state_notify_build_xml().

Referenced by __sip_autodestruct(), and extensionstate_update().

{
   struct ast_str *tmp = ast_str_alloca(4000);
   char from[256], to[256];
   char *c, *mfrom, *mto;
   struct sip_request req;
   const struct cfsubscription_types *subscriptiontype;

   /* If the subscription has not yet been accepted do not send a NOTIFY */
   if (!ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
      return 0;
   }

   memset(from, 0, sizeof(from));
   memset(to, 0, sizeof(to));

   subscriptiontype = find_subscription_type(p->subscribed);

   ast_copy_string(from, sip_get_header(&p->initreq, "From"), sizeof(from));
   c = get_in_brackets(from);
   if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", c);
      return -1;
   }

   mfrom = remove_uri_parameters(c);

   ast_copy_string(to, sip_get_header(&p->initreq, "To"), sizeof(to));
   c = get_in_brackets(to);
   if (strncasecmp(c, "sip:", 4) && strncasecmp(c, "sips:", 5)) {
      ast_log(LOG_WARNING, "Huh?  Not a SIP header (%s)?\n", c);
      return -1;
   }
   mto = remove_uri_parameters(c);

   reqprep(&req, p, SIP_NOTIFY, 0, 1);

   switch(data->state) {
   case AST_EXTENSION_DEACTIVATED:
      if (timeout)
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
      else {
         add_header(&req, "Subscription-State", "terminated;reason=probation");
         add_header(&req, "Retry-After", "60");
      }
      break;
   case AST_EXTENSION_REMOVED:
      add_header(&req, "Subscription-State", "terminated;reason=noresource");
      break;
   default:
      if (p->expiry)
         add_header(&req, "Subscription-State", "active");
      else  /* Expired */
         add_header(&req, "Subscription-State", "terminated;reason=timeout");
   }

   switch (p->subscribed) {
   case XPIDF_XML:
   case CPIM_PIDF_XML:
      add_header(&req, "Event", subscriptiontype->event);
      state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
      add_header(&req, "Content-Type", subscriptiontype->mediatype);
      p->dialogver++;
      break;
   case PIDF_XML: /* Eyebeam supports this format */
      add_header(&req, "Event", subscriptiontype->event);
      state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
      add_header(&req, "Content-Type", subscriptiontype->mediatype);
      p->dialogver++;
      break;
   case DIALOG_INFO_XML: /* SNOM subscribes in this format */
      add_header(&req, "Event", subscriptiontype->event);
      state_notify_build_xml(data, full, p->exten, p->context, &tmp, p, p->subscribed, mfrom, mto);
      add_header(&req, "Content-Type", subscriptiontype->mediatype);
      p->dialogver++;
      break;
   case NONE:
   default:
      break;
   }

   add_content(&req, ast_str_buffer(tmp));

   p->pendinginvite = p->ocseq;  /* Remember that we have a pending NOTIFY in order not to confuse the NOTIFY subsystem */

   /* Send as XMIT_CRITICAL as we may never receive a 200 OK Response which clears p->pendinginvite.
    *
    * extensionstate_update() uses p->pendinginvite for queuing control.
    * Updates stall if pendinginvite <> 0.
    *
    * The most appropriate solution is to remove the subscription when the NOTIFY transaction fails.
    * The client will re-subscribe after restarting or maxexpiry timeout.
    */

   /* RFC6665 4.2.2.  Sending State Information to Subscribers
    * If the NOTIFY request fails due to expiration of SIP Timer F (transaction timeout),
    * the notifier SHOULD remove the subscription.
    */
   return send_request(p, &req, XMIT_CRITICAL, p->ocseq);
}

static const char* trust_id_outbound2str

(

int

mode

)

[static]

Definition at line 19464 of file chan_sip.c.

References map_x_s().

Referenced by _sip_show_peer().

{
   return map_x_s(trust_id_outboundstr, mode, "<error>");
}

static void try_suggested_sip_codec

(

struct sip_pvt *

p

)

[static]

Try setting codec suggested by the SIP_CODEC channel variable.

Definition at line 7231 of file chan_sip.c.

References ast_format_cap_iscompatible(), ast_format_cap_set(), ast_format_clear(), ast_getformatbyname(), ast_log(), ast_format::id, LOG_NOTICE, and pbx_builtin_getvar_helper().

Referenced by sip_answer(), transmit_invite(), transmit_reinvite_with_sdp(), and transmit_response_with_sdp().

{
   struct ast_format fmt;
   const char *codec;

   ast_format_clear(&fmt);

   if (p->outgoing_call) {
      codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_OUTBOUND");
   } else if (!(codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC_INBOUND"))) {
      codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
   }

   if (!codec) 
      return;

   ast_getformatbyname(codec, &fmt);
   if (fmt.id) {
      ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC} variable\n", codec);
      if (ast_format_cap_iscompatible(p->jointcaps, &fmt)) {
         ast_format_cap_set(p->jointcaps, &fmt);
         ast_format_cap_set(p->caps, &fmt);
      } else
         ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
   } else
      ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n", codec);
   return;
}

static void unlink_all_peers_from_tables

(

void

)

[static]

Definition at line 3345 of file chan_sip.c.

References SIP_PEERS_ALL, and unlink_peers_from_tables().

Referenced by unload_module().

{
   unlink_peers_from_tables(SIP_PEERS_ALL);
}

static void unlink_marked_peers_from_tables

(

void

)

[static]

Definition at line 3340 of file chan_sip.c.

References SIP_PEERS_MARKED, and unlink_peers_from_tables().

Referenced by sip_do_reload(), and sip_prune_realtime().

{
   unlink_peers_from_tables(SIP_PEERS_MARKED);
}

static void unlink_peer_from_tables

(

struct sip_peer *

peer

)

[static]

Definition at line 3351 of file chan_sip.c.

References ao2_t_unlink, and ast_sockaddr_isnull().

Referenced by expire_register().

{
   ao2_t_unlink(peers, peer, "ao2_unlink of peer from peers table");
   if (!ast_sockaddr_isnull(&peer->addr)) {
      ao2_t_unlink(peers_by_ip, peer, "ao2_unlink of peer from peers_by_ip table");
   }
}

static void unlink_peers_from_tables

(

peer_unlink_flag_t

flag

)

[static]

Definition at line 3331 of file chan_sip.c.

References ao2_t_callback, match_and_cleanup_peer_sched(), OBJ_MULTIPLE, OBJ_NODATA, and OBJ_UNLINK.

Referenced by unlink_all_peers_from_tables(), and unlink_marked_peers_from_tables().

{
   ao2_t_callback(peers, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE,
      match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers");
   ao2_t_callback(peers_by_ip, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE,
      match_and_cleanup_peer_sched, &flag, "initiating callback to remove marked peers");
}

static int unload_module

(

void

)

[static]

PBX unload module API.

Definition at line 34767 of file chan_sip.c.

References acl_change_event_unsubscribe(), ao2_container_count(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_t_iterator_next, ao2_t_ref, ARRAY_LEN, ast_cc_agent_unregister(), ast_cc_monitor_unregister(), ast_channel_unregister(), ast_cli_unregister_multiple(), ast_config_destroy(), ast_context_destroy(), ast_context_find(), ast_custom_function_unregister(), ast_data_unregister, ast_debug, ast_dnsmgr_release(), ast_format_cap_destroy(), ast_free, ast_free_acl_list(), ast_free_ha(), ast_io_remove(), ast_manager_unregister(), ast_msg_tech_unregister(), ast_mutex_lock, ast_mutex_unlock, AST_PTHREADT_NULL, AST_PTHREADT_STOP, ast_rtp_glue_unregister(), ast_sched_context_destroy(), ast_sched_dump(), ast_sip_api_provider_unregister(), ast_softhangup(), AST_SOFTHANGUP_APPUNLOAD, ast_ssl_teardown(), ast_tcptls_server_stop(), AST_TEST_UNREGISTER, ast_udptl_proto_unregister(), ast_unload_realtime(), ast_unregister_application(), ast_websocket_remove_protocol(), ASTOBJ_CONTAINER_DESTROY, ASTOBJ_CONTAINER_DESTROYALL, ASTOBJ_CONTAINER_TRAVERSE, ASTOBJ_UNLOCK, ASTOBJ_UNREF, ASTOBJ_WRLOCK, authl_lock, ast_tls_config::cafile, ast_channel_tech::capabilities, ast_tls_config::capath, ast_tls_config::certfile, ast_tls_config::cipher, cleanup_all_regs(), clear_sip_domains(), destroy_escs(), dialog_unlink_all(), io_context_destroy(), ast_tcptls_session_args::master, monitor_thread, monlock, network_change_event_unsubscribe(), ast_tls_config::pvtfile, regl, sip_cfg, sip_epa_unregister_all(), sip_registry_destroy(), sip_reqresp_parser_exit(), sip_subscribe_mwi_destroy(), sip_unregister_tests(), sip_websocket_callback(), submwil, thread, ast_tcptls_session_args::tls_cfg, and unlink_all_peers_from_tables().

{
   struct sip_pvt *p;
   struct sip_threadinfo *th;
   struct ast_context *con;
   struct ao2_iterator i;
   int wait_count;

   ast_sip_api_provider_unregister();

   ast_websocket_remove_protocol("sip", sip_websocket_callback);

   network_change_event_unsubscribe();
   acl_change_event_unsubscribe();

   ast_sched_dump(sched);
   
   /* First, take us out of the channel type list */
   ast_channel_unregister(&sip_tech);

   ast_msg_tech_unregister(&sip_msg_tech);

   /* Unregister dial plan functions */
   ast_custom_function_unregister(&sipchaninfo_function);
   ast_custom_function_unregister(&sippeer_function);
   ast_custom_function_unregister(&sip_header_function);
   ast_custom_function_unregister(&checksipdomain_function);

   /* Unregister dial plan applications */
   ast_unregister_application(app_dtmfmode);
   ast_unregister_application(app_sipaddheader);
   ast_unregister_application(app_sipremoveheader);
#ifdef TEST_FRAMEWORK
   ast_unregister_application(app_sipsendcustominfo);

   AST_TEST_UNREGISTER(test_sip_peers_get);
   AST_TEST_UNREGISTER(test_sip_mwi_subscribe_parse);
   AST_TEST_UNREGISTER(test_tcp_message_fragmentation);
   AST_TEST_UNREGISTER(get_in_brackets_const_test);
#endif
   /* Unregister all the AstData providers */
   ast_data_unregister(NULL);

   /* Unregister CLI commands */
   ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));

   /* Disconnect from UDPTL */
   ast_udptl_proto_unregister(&sip_udptl);

   /* Disconnect from RTP engine */
   ast_rtp_glue_unregister(&sip_rtp_glue);

   /* Unregister AMI actions */
   ast_manager_unregister("SIPpeers");
   ast_manager_unregister("SIPshowpeer");
   ast_manager_unregister("SIPqualifypeer");
   ast_manager_unregister("SIPshowregistry");
   ast_manager_unregister("SIPnotify");
   ast_manager_unregister("SIPpeerstatus");
   
   /* Kill TCP/TLS server threads */
   if (sip_tcp_desc.master) {
      ast_tcptls_server_stop(&sip_tcp_desc);
   }
   if (sip_tls_desc.master) {
      ast_tcptls_server_stop(&sip_tls_desc);
   }
   ast_ssl_teardown(sip_tls_desc.tls_cfg);

   /* Kill all existing TCP/TLS threads */
   i = ao2_iterator_init(threadt, 0);
   while ((th = ao2_t_iterator_next(&i, "iterate through tcp threads for 'sip show tcp'"))) {
      pthread_t thread = th->threadid;
      th->stop = 1;
      pthread_kill(thread, SIGURG);
      ao2_t_ref(th, -1, "decrement ref from iterator");
   }
   ao2_iterator_destroy(&i);

   /* Hangup all dialogs if they have an owner */
   i = ao2_iterator_init(dialogs, 0);
   while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      if (p->owner)
         ast_softhangup(p->owner, AST_SOFTHANGUP_APPUNLOAD);
      ao2_t_ref(p, -1, "toss dialog ptr from iterator_next");
   }
   ao2_iterator_destroy(&i);

   unlink_all_peers_from_tables();

   ast_mutex_lock(&monlock);
   if (monitor_thread && (monitor_thread != AST_PTHREADT_STOP) && (monitor_thread != AST_PTHREADT_NULL)) {
      pthread_t th = monitor_thread;
      monitor_thread = AST_PTHREADT_STOP;
      pthread_cancel(th);
      pthread_kill(th, SIGURG);
      ast_mutex_unlock(&monlock);
      pthread_join(th, NULL);
   } else {
      monitor_thread = AST_PTHREADT_STOP;
      ast_mutex_unlock(&monlock);
   }

   /* Destroy all the dialogs and free their memory */
   i = ao2_iterator_init(dialogs, 0);
   while ((p = ao2_t_iterator_next(&i, "iterate thru dialogs"))) {
      dialog_unlink_all(p);
      ao2_t_ref(p, -1, "throw away iterator result");
   }
   ao2_iterator_destroy(&i);

   /* Free memory for local network address mask */
   ast_free_ha(localaddr);

   ast_mutex_lock(&authl_lock);
   if (authl) {
      ao2_t_ref(authl, -1, "Removing global authentication");
      authl = NULL;
   }
   ast_mutex_unlock(&authl_lock);

   sip_epa_unregister_all();
   destroy_escs();

   ast_free(default_tls_cfg.certfile);
   ast_free(default_tls_cfg.pvtfile);
   ast_free(default_tls_cfg.cipher);
   ast_free(default_tls_cfg.cafile);
   ast_free(default_tls_cfg.capath);

   cleanup_all_regs();
   ASTOBJ_CONTAINER_DESTROYALL(&regl, sip_registry_destroy);
   ASTOBJ_CONTAINER_DESTROY(&regl);

   ASTOBJ_CONTAINER_TRAVERSE(&submwil, 1, do {
      ASTOBJ_WRLOCK(iterator);
      if (iterator->dnsmgr) {
         ast_dnsmgr_release(iterator->dnsmgr);
         iterator->dnsmgr = NULL;
         ASTOBJ_UNREF(iterator, sip_subscribe_mwi_destroy);
      }
      ASTOBJ_UNLOCK(iterator);
   } while(0));
   ASTOBJ_CONTAINER_DESTROYALL(&submwil, sip_subscribe_mwi_destroy);
   ASTOBJ_CONTAINER_DESTROY(&submwil);

   /*
    * Wait awhile for the TCP/TLS thread container to become empty.
    *
    * XXX This is a hack, but the worker threads cannot be created
    * joinable.  They can die on their own and remove themselves
    * from the container thus resulting in a huge memory leak.
    */
   wait_count = 1000;
   while (ao2_container_count(threadt) && --wait_count) {
      sched_yield();
   }
   if (!wait_count) {
      ast_debug(2, "TCP/TLS thread container did not become empty :(\n");
   }

   ao2_t_ref(bogus_peer, -1, "unref the bogus_peer");

   ao2_t_ref(peers, -1, "unref the peers table");
   ao2_t_ref(peers_by_ip, -1, "unref the peers_by_ip table");
   ao2_t_ref(dialogs, -1, "unref the dialogs table");
   ao2_t_ref(dialogs_needdestroy, -1, "unref dialogs_needdestroy");
   ao2_t_ref(dialogs_rtpcheck, -1, "unref dialogs_rtpcheck");
   ao2_t_ref(threadt, -1, "unref the thread table");
   ao2_t_ref(sip_monitor_instances, -1, "unref the sip_monitor_instances table");

   clear_sip_domains();
   sip_cfg.contact_acl = ast_free_acl_list(sip_cfg.contact_acl);
   if (sipsock_read_id) {
      ast_io_remove(io, sipsock_read_id);
      sipsock_read_id = NULL;
   }
   close(sipsock);
   io_context_destroy(io);
   ast_sched_context_destroy(sched);
   con = ast_context_find(used_context);
   if (con) {
      ast_context_destroy(con, "SIP");
   }
   ast_unload_realtime("sipregs");
   ast_unload_realtime("sippeers");
   ast_cc_monitor_unregister(&sip_cc_monitor_callbacks);
   ast_cc_agent_unregister(&sip_cc_agent_callbacks);

   sip_reqresp_parser_exit();
   sip_unregister_tests();

   if (notify_types) {
      ast_config_destroy(notify_types);
      notify_types = NULL;
   }

   ast_format_cap_destroy(sip_tech.capabilities);
   sip_cfg.caps = ast_format_cap_destroy(sip_cfg.caps);

   return 0;
}

static int update_call_counter	(	struct sip_pvt *	fup,
		int	event
	)		[static]

update_call_counter: Handle call_limit for SIP devices Setting a call-limit will cause calls above the limit not to be accepted.

Remember that for a type=friend, there's one limit for the user and another for the peer, not a combined call limit. This will cause unexpected behaviour in subscriptions, since a "friend" is *two* devices in Asterisk, not one.

Thought: For realtime, we should probably update storage with inuse counter...

Returns:

0 if call is ok (no call limit, below threshold) -1 on rejection of call

Definition at line 6676 of file chan_sip.c.

References ao2_lock, ao2_unlock, ast_clear_flag, ast_copy_string(), ast_debug, AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, ast_devstate_changed(), ast_log(), ast_set_flag, ast_test_flag, FALSE, LOG_ERROR, LOG_NOTICE, name, sip_cfg, sip_peer_hold(), sip_pvt_lock, sip_pvt_unlock, sip_ref_peer(), and sip_unref_peer().

Referenced by __sip_destroy(), handle_request_cancel(), handle_request_invite(), handle_response_invite(), sip_call(), and sip_hangup().

{
   char name[256];
   int *inuse = NULL, *call_limit = NULL, *ringing = NULL;
   int outgoing = fup->outgoing_call;
   struct sip_peer *p = NULL;

   ast_debug(3, "Updating call counter for %s call\n", outgoing ? "outgoing" : "incoming");


   /* Test if we need to check call limits, in order to avoid
      realtime lookups if we do not need it */
   if (!ast_test_flag(&fup->flags[0], SIP_CALL_LIMIT) && !ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD))
      return 0;

   ast_copy_string(name, fup->username, sizeof(name));

   /* Check the list of devices */
   if (fup->relatedpeer) {
      p = sip_ref_peer(fup->relatedpeer, "ref related peer for update_call_counter");
      inuse = &p->inuse;
      call_limit = &p->call_limit;
      ringing = &p->ringing;
      ast_copy_string(name, fup->peername, sizeof(name));
   }
   if (!p) {
      ast_debug(2, "%s is not a local device, no call limit\n", name);
      return 0;
   }

   switch(event) {
   /* incoming and outgoing affects the inuse counter */
   case DEC_CALL_LIMIT:
      /* Decrement inuse count if applicable */
      if (inuse) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (*inuse > 0) {
            if (ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
               (*inuse)--;
               ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
            }
         } else {
            *inuse = 0;
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }

      /* Decrement ringing count if applicable */
      if (ringing) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (*ringing > 0) {
            if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
               (*ringing)--;
               ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
            }
         } else {
            *ringing = 0;
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }

      /* Decrement onhold count if applicable */
      sip_pvt_lock(fup);
      ao2_lock(p);
      if (ast_test_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD) && sip_cfg.notifyhold) {
         ast_clear_flag(&fup->flags[1], SIP_PAGE2_CALL_ONHOLD);
         ao2_unlock(p);
         sip_pvt_unlock(fup);
         sip_peer_hold(fup, FALSE);
      } else {
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (sipdebug)
         ast_debug(2, "Call %s %s '%s' removed from call limit %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
      break;

   case INC_CALL_RINGING:
   case INC_CALL_LIMIT:
      /* If call limit is active and we have reached the limit, reject the call */
      if (*call_limit > 0 ) {
         if (*inuse >= *call_limit) {
            ast_log(LOG_NOTICE, "Call %s %s '%s' rejected due to usage limit of %d\n", outgoing ? "to" : "from", "peer", name, *call_limit);
            sip_unref_peer(p, "update_call_counter: unref peer p, call limit exceeded");
            return -1;
         }
      }
      if (ringing && (event == INC_CALL_RINGING)) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (!ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
            (*ringing)++;
            ast_set_flag(&fup->flags[0], SIP_INC_RINGING);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (inuse) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (!ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
            (*inuse)++;
            ast_set_flag(&fup->flags[0], SIP_INC_COUNT);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      if (sipdebug) {
         ast_debug(2, "Call %s %s '%s' is %d out of %d\n", outgoing ? "to" : "from", "peer", name, *inuse, *call_limit);
      }
      break;

   case DEC_CALL_RINGING:
      if (ringing) {
         sip_pvt_lock(fup);
         ao2_lock(p);
         if (ast_test_flag(&fup->flags[0], SIP_INC_RINGING)) {
            if (*ringing > 0) {
               (*ringing)--;
            }
            ast_clear_flag(&fup->flags[0], SIP_INC_RINGING);
         }
         ao2_unlock(p);
         sip_pvt_unlock(fup);
      }
      break;

   default:
      ast_log(LOG_ERROR, "update_call_counter(%s, %d) called with no event!\n", name, event);
   }

   if (p) {
      ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "SIP/%s", p->name);
      sip_unref_peer(p, "update_call_counter: sip_unref_peer from call counter");
   }
   return 0;
}

static void update_connectedline	(	struct sip_pvt *	p,
		const void *	data,
		size_t	datalen
	)		[static]

Notify peer that the connected line has changed.

Definition at line 15000 of file chan_sip.c.

References add_header(), add_rpid(), add_sdp(), add_supported(), append_history, ast_channel_connected_effective_id(), ast_channel_name(), ast_clear_flag, ast_debug, ast_set_flag, ast_state2str(), AST_STATE_RING, AST_STATE_RINGING, AST_STATE_UP, ast_strlen_zero(), ast_test_flag, FALSE, initialize_initreq(), is_method_allowed(), ast_party_id::name, ast_party_id::number, reqprep(), respprep(), S_COR, send_request(), send_response(), ast_party_name::str, ast_party_number::str, TRUE, ast_party_name::valid, and ast_party_number::valid.

Referenced by sip_indicate().

{
   struct ast_party_id connected_id = ast_channel_connected_effective_id(p->owner);

   if (!ast_test_flag(&p->flags[0], SIP_SENDRPID)) {
      return;
   }
   if (!connected_id.number.valid
      || ast_strlen_zero(connected_id.number.str)) {
      return;
   }

   append_history(p, "ConnectedLine", "%s party is now %s <%s>",
      ast_test_flag(&p->flags[0], SIP_OUTGOING) ? "Calling" : "Called",
      S_COR(connected_id.name.valid, connected_id.name.str, ""),
      S_COR(connected_id.number.valid, connected_id.number.str, ""));

   if (ast_channel_state(p->owner) == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      struct sip_request req;

      if (!p->pendinginvite && (p->invitestate == INV_CONFIRMED || p->invitestate == INV_TERMINATED)) {
         reqprep(&req, p, ast_test_flag(&p->flags[0], SIP_REINVITE_UPDATE) ? SIP_UPDATE : SIP_INVITE, 0, 1);

         add_header(&req, "Allow", ALLOWED_METHODS);
         add_supported(p, &req);
         add_rpid(&req, p);
         add_sdp(&req, p, FALSE, TRUE, FALSE);

         initialize_initreq(p, &req);
         p->lastinvite = p->ocseq;
         ast_set_flag(&p->flags[0], SIP_OUTGOING);
         p->invitestate = INV_CALLING;
         send_request(p, &req, XMIT_CRITICAL, p->ocseq);
      } else if ((is_method_allowed(&p->allowed_methods, SIP_UPDATE)) && (!ast_strlen_zero(p->okcontacturi))) { 
         reqprep(&req, p, SIP_UPDATE, 0, 1);
         add_rpid(&req, p);
         add_header(&req, "X-Asterisk-rpid-update", "Yes");
         send_request(p, &req, XMIT_CRITICAL, p->ocseq);
      } else {
         /* We cannot send the update yet, so we have to wait until we can */
         ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
      }
   } else {
      ast_set_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
      if (ast_test_flag(&p->flags[1], SIP_PAGE2_RPID_IMMEDIATE)) {
         struct sip_request resp;

         if ((ast_channel_state(p->owner) == AST_STATE_RING) && !ast_test_flag(&p->flags[0], SIP_PROGRESS_SENT)) {
            ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
            respprep(&resp, p, "180 Ringing", &p->initreq);
            add_rpid(&resp, p);
            send_response(p, &resp, XMIT_UNRELIABLE, 0);
            ast_set_flag(&p->flags[0], SIP_RINGING);
         } else if (ast_channel_state(p->owner) == AST_STATE_RINGING) {
            ast_clear_flag(&p->flags[1], SIP_PAGE2_CONNECTLINEUPDATE_PEND);
            respprep(&resp, p, "183 Session Progress", &p->initreq);
            add_rpid(&resp, p);
            send_response(p, &resp, XMIT_UNRELIABLE, 0);
            ast_set_flag(&p->flags[0], SIP_PROGRESS_SENT);
         } else {
            ast_debug(1, "Unable able to send update to '%s' in state '%s'\n", ast_channel_name(p->owner), ast_state2str(ast_channel_state(p->owner)));
         }
      }
   }
}

static void update_peer	(	struct sip_peer *	p,
		int	expire
	)		[static]

Update peer data in database (if used)

Definition at line 5254 of file chan_sip.c.

References ast_test_flag, realtime_update_peer(), and sip_cfg.

Referenced by register_verify().

{
   int rtcachefriends = ast_test_flag(&p->flags[1], SIP_PAGE2_RTCACHEFRIENDS);
   if (sip_cfg.peer_rtupdate &&
       (p->is_realtime || rtcachefriends)) {
      realtime_update_peer(p->name, &p->addr, p->username, p->fullcontact, p->useragent, expire, p->deprecated_username, p->lastms);
   }
}

static void update_peer_lastmsgssent	(	struct sip_peer *	peer,
		int	value,
		int	locked
	)		[static]

Definition at line 16881 of file chan_sip.c.

References ao2_lock, ao2_unlock, and value.

Referenced by register_verify(), and sip_send_mwi_to_peer().

{
   if (!locked) {
      ao2_lock(peer);
   }
   peer->lastmsgssent = value;
   if (!locked) {
      ao2_unlock(peer);
   }
}

static void update_provisional_keepalive	(	struct sip_pvt *	pvt,
		int	with_sdp
	)		[static]

Definition at line 4694 of file chan_sip.c.

References ast_sched_add(), AST_SCHED_DEL_UNREF, send_provisional_keepalive(), and send_provisional_keepalive_with_sdp().

Referenced by transmit_provisional_response().

{
   AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_sched_id, dialog_unref(pvt, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));

   pvt->provisional_keepalive_sched_id = ast_sched_add(sched, PROVIS_KEEPALIVE_TIMEOUT,
      with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive, dialog_ref(pvt, "Increment refcount to pass dialog pointer to sched callback"));
}

static void update_redirecting	(	struct sip_pvt *	p,
		const void *	data,
		size_t	datalen
	)		[static]

Send a provisional response indicating that a call was redirected.

Definition at line 14986 of file chan_sip.c.

References add_diversion(), AST_STATE_UP, ast_test_flag, respprep(), and send_response().

Referenced by handle_request_invite(), handle_request_refer(), handle_response(), handle_response_invite(), and sip_indicate().

{
   struct sip_request resp;

   if (ast_channel_state(p->owner) == AST_STATE_UP || ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
      return;
   }

   respprep(&resp, p, "181 Call is being forwarded", &p->initreq);
   add_diversion(&resp, p);
   send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

---

Variable Documentation

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Session Initiation Protocol (SIP)" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .load = load_module, .unload = unload_module, .reload = reload, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .nonoptreq = "res_crypto,chan_local,res_http_websocket", } [static]

Definition at line 34976 of file chan_sip.c.

struct ast_event_sub* acl_change_event_subscription [static]

subscription id for named ACL system change events

Definition at line 832 of file chan_sip.c.

struct cfalias aliases[] [static]

Definition at line 8204 of file chan_sip.c.

struct _map_x_s allowoverlapstr[] [static]

Definition at line 19444 of file chan_sip.c.

int apeerobjs = 0 [static]

Autocreated peer objects

Definition at line 824 of file chan_sip.c.

char* app_dtmfmode = "SIPDtmfMode" [static]

Definition at line 32967 of file chan_sip.c.

char* app_sipaddheader = "SIPAddHeader" [static]

Definition at line 32968 of file chan_sip.c.

char* app_sipremoveheader = "SIPRemoveHeader" [static]

Definition at line 32969 of file chan_sip.c.

struct ast_module_info* ast_module_info = &__mod_info [static]

Definition at line 34976 of file chan_sip.c.

struct sip_auth_container* authl = NULL [static]

Authentication container for realm authentication.

Definition at line 1219 of file chan_sip.c.

Referenced by build_reply_digest(), and sip_show_settings().

ast_mutex_t authl_lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 } [static]

Global authentication container protection while adjusting the references.

Definition at line 1221 of file chan_sip.c.

Referenced by build_reply_digest(), reload_config(), sip_show_settings(), and unload_module().

int authlimit = DEFAULT_AUTHLIMIT [static]

Definition at line 617 of file chan_sip.c.

int authtimeout = DEFAULT_AUTHTIMEOUT [static]

Definition at line 618 of file chan_sip.c.

struct _map_x_s autopeermodes[] [static]

Definition at line 18861 of file chan_sip.c.

struct ast_sockaddr bindaddr

UDP: The address we bind to

Definition at line 1241 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), create_addr(), dialog_initialize_rtp(), get_address_family_filter(), initialize_udptl(), reload_config(), sip_show_settings(), and sipsock_read().

struct sip_peer* bogus_peer [static]

A bogus peer, to be used when authentication should fail.

Definition at line 1192 of file chan_sip.c.

Referenced by check_peer_ok(), register_verify(), and sip_reload().

int can_parse_xml [static]

We use libxml2 in order to parse XML that may appear in the body of a SIP message. Currently, the only usage is for parsing PIDF bodies of incoming PUBLISH requests in the call-completion event package. This variable is set at module load time and may be checked at runtime to determine if XML parsing support was found.

Definition at line 817 of file chan_sip.c.

struct epa_static_data cc_epa_static_data [static]

Definition at line 957 of file chan_sip.c.

struct sip_esc_publish_callbacks cc_esc_publish_callbacks [static]

Initial value:

 {
   .initial_handler = cc_esc_publish_handler,
   .modify_handler = cc_esc_publish_handler,
}

Definition at line 1005 of file chan_sip.c.

unsigned int chan_idx [static]

used in naming sip channel

Definition at line 763 of file chan_sip.c.

struct ast_sip_api_tech chan_sip_api_provider [static]

Initial value:

 {
   .version = AST_SIP_API_VERSION,
   .name = "chan_sip",
   .sipinfo_send = sipinfo_send,
}

Definition at line 34576 of file chan_sip.c.

struct ast_custom_function checksipdomain_function [static]

Initial value:

 {
   .name = "CHECKSIPDOMAIN",
   .read = func_check_sipdomain,
}

Definition at line 22101 of file chan_sip.c.

struct ast_cli_entry cli_sip[] [static]

SIP Cli commands definition.

Definition at line 33617 of file chan_sip.c.

const char config[] = "sip.conf" [static]

Main configuration file

Definition at line 632 of file chan_sip.c.

struct ast_sockaddr debugaddr [static]

Definition at line 1285 of file chan_sip.c.

Referenced by sip_debug_test_addr(), sip_do_debug(), sip_do_debug_ip(), and sip_do_debug_peer().

char default_callerid[AST_MAX_EXTENSION] [static]

Default caller ID for sip messages

Definition at line 734 of file chan_sip.c.

Referenced by initreqprep().

char default_engine[256] [static]

Default RTP engine

Definition at line 746 of file chan_sip.c.

int default_expiry = DEFAULT_DEFAULT_EXPIRY [static]

Definition at line 611 of file chan_sip.c.

Referenced by AST_TEST_DEFINE(), handle_response_register(), parse_register_contact(), sip_parse_register_line(), and sip_register().

char default_fromdomain[AST_MAX_EXTENSION] [static]

Default domain on outound messages

Definition at line 736 of file chan_sip.c.

int default_fromdomainport [static]

Default domain port on outbound messages

Definition at line 737 of file chan_sip.c.

Referenced by sip_alloc(), and transmit_response_using_temp().

struct ast_jb_conf default_jbconf [static]

Global jitterbuffer configuration - by default, jb is disabled.

Note:

Values shown here match the defaults shown in sip.conf.sample

Definition at line 622 of file chan_sip.c.

int default_keepalive [static]

Default keepalive= setting

Definition at line 741 of file chan_sip.c.

Referenced by set_peer_defaults().

char default_language[MAX_LANGUAGE] [static]

Default language setting for new channels

Definition at line 733 of file chan_sip.c.

int default_maxcallbitrate [static]

Maximum bitrate for call

Definition at line 747 of file chan_sip.c.

Referenced by build_peer(), set_peer_defaults(), and sip_alloc().

char default_mohinterpret[MAX_MUSICCLASS] [static]

Global setting for moh class to use when put on hold

Definition at line 742 of file chan_sip.c.

char default_mohsuggest[MAX_MUSICCLASS] [static]

Global setting for moh class to suggest when putting a bridged channel on hold

Definition at line 743 of file chan_sip.c.

char default_mwi_from[80] [static]

Default caller ID for MWI updates

Definition at line 735 of file chan_sip.c.

char default_notifymime[AST_MAX_EXTENSION] [static]

Default MIME media type for MWI notify messages

Definition at line 738 of file chan_sip.c.

char default_parkinglot[AST_MAX_CONTEXT] [static]

Parkinglot

Definition at line 745 of file chan_sip.c.

struct ast_codec_pref default_prefs [static]

Default codec prefs

Definition at line 748 of file chan_sip.c.

Referenced by reload_config(), set_peer_defaults(), sip_alloc(), sip_show_settings(), and temp_peer().

unsigned int default_primary_transport [static]

Default primary Transport (enum sip_transport) for outbound connections to devices

Definition at line 751 of file chan_sip.c.

Referenced by build_peer(), and set_peer_defaults().

const int DEFAULT_PUBLISH_EXPIRES = 3600 [static]

Definition at line 1000 of file chan_sip.c.

Referenced by determine_sip_publish_type().

int default_qualify [static]

Default Qualify= setting

Definition at line 740 of file chan_sip.c.

Referenced by set_peer_defaults().

struct ast_tls_config default_tls_cfg [static]

Default TLS connection configuration.

Definition at line 2299 of file chan_sip.c.

unsigned int default_transports [static]

Default Transports (enum sip_transport) that are acceptable

Definition at line 750 of file chan_sip.c.

Referenced by build_peer(), reload_config(), and set_peer_defaults().

char default_vmexten[AST_MAX_EXTENSION] [static]

Default From Username on MWI updates

Definition at line 739 of file chan_sip.c.

char default_zone[MAX_TONEZONE_COUNTRY] [static]

Default tone zone for channels created from the SIP driver

Definition at line 749 of file chan_sip.c.

struct ao2_container* dialogs [static]

Here we implement the container for dialogs (sip_pvt), defining generic wrapper functions to ease the transition from the current implementation (a single linked list) to a different container. In addition to a reference to the container, we need functions to lock/unlock the container and individual items, and functions to add/remove references to the individual items.

Definition at line 1179 of file chan_sip.c.

struct ao2_container* dialogs_needdestroy

Here we implement the container for dialogs which are in the dialog_needdestroy state to iterate only through the dialogs unlink them instead of iterate through all dialogs

Definition at line 1160 of file chan_sip.c.

struct ao2_container* dialogs_rtpcheck

Here we implement the container for dialogs which have rtp traffic and rtptimeout, rtpholdtimeout or rtpkeepalive set. We use this container instead the whole dialog list.

Definition at line 1168 of file chan_sip.c.

struct domain_list domain_list [static]

The SIP domain list

struct _map_x_s dtmfstr[] [static]

mapping between dtmf flags and strings

Definition at line 19409 of file chan_sip.c.

Referenced by conf_run(), and send_dtmf().

unsigned int dumphistory [static]

Dump history to verbose before destroying SIP dialog

Definition at line 787 of file chan_sip.c.

struct epa_static_data_list epa_static_data_list

int esc_etag_counter [static]

Used to create new entity IDs by ESCs.

Definition at line 999 of file chan_sip.c.

const int ESC_MAX_BUCKETS = 37 [static]

Definition at line 1034 of file chan_sip.c.

struct event_state_compositor event_state_compositors[] [static]

Referenced by destroy_escs(), get_esc(), and initialize_escs().

struct ast_sockaddr externaddr [static]

our external IP address/port for SIP sessions. externaddr.sin_addr is only set when we know we might be behind a NAT, and this is done using a variety of (mutually exclusive) ways from the config file:

+ with "externaddr = host[:port]" we specify the address/port explicitly. The address is looked up only once when (re)loading the config file;

+ with "externhost = host[:port]" we do a similar thing, but the hostname is stored in externhost, and the hostname->IP mapping is refreshed every 'externrefresh' seconds;

Other variables (externhost, externexpire, externrefresh) are used to support the above functions. External IP address if we are behind NAT

Definition at line 1266 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), reload_config(), and sip_show_settings().

time_t externexpire [static]

Expiration counter for re-resolving external host name in dynamic DNS

Definition at line 1270 of file chan_sip.c.

char externhost[MAXHOSTNAMELEN] [static]

External host name

Definition at line 1269 of file chan_sip.c.

int externrefresh = 10 [static]

Refresh timer for DNS-based external address (dyndns)

Definition at line 1271 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor().

uint16_t externtcpport [static]

external tcp port

Definition at line 1272 of file chan_sip.c.

uint16_t externtlsport [static]

external tls port

Definition at line 1273 of file chan_sip.c.

struct _map_x_s faxecmodes[] [static]

Definition at line 19976 of file chan_sip.c.

int global_authfailureevents [static]

Whether we send authentication failure manager events or not. Default no.

Definition at line 791 of file chan_sip.c.

unsigned int global_autoframing [static]

Turn autoframing on or off.

Definition at line 795 of file chan_sip.c.

Referenced by set_peer_defaults(), and sip_alloc().

int global_callcounter [static]

Enable call counters for all devices. This is currently enabled by setting the peer call-limit to INT_MAX. When we remove the call-limit from the code, we can make it with just a boolean flag in the device structure

Definition at line 775 of file chan_sip.c.

unsigned int global_cos_audio [static]

802.1p class of service for audio RTP packets

Definition at line 783 of file chan_sip.c.

unsigned int global_cos_sip [static]

802.1p class of service for SIP packets

Definition at line 782 of file chan_sip.c.

unsigned int global_cos_text [static]

802.1p class of service for text RTP packets

Definition at line 785 of file chan_sip.c.

unsigned int global_cos_video [static]

802.1p class of service for video RTP packets

Definition at line 784 of file chan_sip.c.

int global_dynamic_exclude_static = 0 [static]

Exclude static peers from contact registrations

Definition at line 807 of file chan_sip.c.

struct ast_flags global_flags[3] = {{0}} [static]

global SIP_ flags

Definition at line 828 of file chan_sip.c.

struct ast_jb_conf global_jbconf [static]

Global jitterbuffer configuration

Definition at line 630 of file chan_sip.c.

Referenced by reload_config(), sip_get_rtp_peer(), sip_new(), and sip_show_settings().

int global_match_auth_username [static]

Match auth username if available instead of From: Default off.

Definition at line 764 of file chan_sip.c.

int global_max_se [static]

Highest threshold for session refresh interval

Definition at line 803 of file chan_sip.c.

Referenced by build_peer(), handle_request_invite_st(), set_peer_defaults(), and st_get_se().

int global_min_se [static]

Lowest threshold for session refresh interval

Definition at line 802 of file chan_sip.c.

Referenced by build_peer(), set_peer_defaults(), and st_get_se().

int global_prematuremediafilter [static]

Enable/disable premature frames in a call (causing 183 early media)

Definition at line 767 of file chan_sip.c.

int global_qualify_gap [static]

Time between our group of peer pokes

Definition at line 797 of file chan_sip.c.

Referenced by sip_poke_all_peers().

int global_qualify_peers [static]

Number of peers to poke at a given time

Definition at line 798 of file chan_sip.c.

int global_qualifyfreq [static]

Qualify frequency

Definition at line 796 of file chan_sip.c.

Referenced by build_peer(), and set_peer_defaults().

unsigned char global_refer_addheaders [static]

Add extra headers to outgoing REFER

Definition at line 808 of file chan_sip.c.

int global_reg_retry_403 [static]

Treat 403 responses to registrations as 401 responses

Definition at line 773 of file chan_sip.c.

int global_reg_timeout [static]

Global time between attempts for outbound registrations

Definition at line 771 of file chan_sip.c.

int global_regattempts_max [static]

Registration attempts before giving up

Definition at line 772 of file chan_sip.c.

int global_relaxdtmf [static]

Relax DTMF

Definition at line 766 of file chan_sip.c.

int global_rtpholdtimeout [static]

Time out call if no RTP during hold

Definition at line 769 of file chan_sip.c.

Referenced by build_peer(), check_user_full(), create_addr(), and set_peer_defaults().

int global_rtpkeepalive [static]

Send RTP keepalives

Definition at line 770 of file chan_sip.c.

Referenced by build_peer(), check_user_full(), create_addr(), and set_peer_defaults().

int global_rtptimeout [static]

Time out call if no RTP

Definition at line 768 of file chan_sip.c.

Referenced by build_peer(), check_user_full(), create_addr(), and set_peer_defaults().

char global_sdpowner[AST_MAX_EXTENSION] [static]

SDP owner name for the SIP channel

Definition at line 790 of file chan_sip.c.

char global_sdpsession[AST_MAX_EXTENSION] [static]

SDP session name for the SIP channel

Definition at line 789 of file chan_sip.c.

int global_shrinkcallerid [static]

enable or disable shrinking of caller id

Definition at line 774 of file chan_sip.c.

enum st_mode global_st_mode [static]

Mode of operation for Session-Timers

Definition at line 800 of file chan_sip.c.

Referenced by build_peer(), set_peer_defaults(), and st_get_mode().

enum st_refresher_param global_st_refresher [static]

Session-Timer refresher

Definition at line 801 of file chan_sip.c.

Referenced by build_peer(), and set_peer_defaults().

int global_store_sip_cause [static]

Whether the MASTER_CHANNEL(HASH(SIP_CAUSE,[chan_name])) var should be set

Definition at line 805 of file chan_sip.c.

int global_t1 [static]

T1 time

Definition at line 792 of file chan_sip.c.

Referenced by create_addr(), set_peer_defaults(), sip_alloc(), and sip_scheddestroy().

int global_t1min [static]

T1 roundtrip time minimum

Definition at line 793 of file chan_sip.c.

Referenced by build_peer(), and reload_config().

int global_t38_maxdatagram [static]

global T.38 FaxMaxDatagram override

Definition at line 829 of file chan_sip.c.

Referenced by handle_t38_options(), initialize_udptl(), and set_peer_defaults().

int global_timer_b [static]

Timer B - RFC 3261 Section 17.1.1.2

Definition at line 794 of file chan_sip.c.

Referenced by build_peer(), create_addr(), set_peer_defaults(), sip_alloc(), and sip_scheddestroy().

unsigned int global_tos_audio [static]

IP type of service for audio RTP packets

Definition at line 779 of file chan_sip.c.

unsigned int global_tos_sip [static]

IP type of service for SIP packets

Definition at line 778 of file chan_sip.c.

unsigned int global_tos_text [static]

IP type of service for text RTP packets

Definition at line 781 of file chan_sip.c.

unsigned int global_tos_video [static]

IP type of service for video RTP packets

Definition at line 780 of file chan_sip.c.

char global_useragent[AST_MAX_EXTENSION] [static]

Useragent for the SIP channel

Definition at line 788 of file chan_sip.c.

const int HASH_DIALOG_SIZE = 563 [static]

Definition at line 887 of file chan_sip.c.

const int HASH_PEER_SIZE = 563 [static]

Size of peer hash table, prime number preferred!

Definition at line 886 of file chan_sip.c.

struct _map_x_s insecurestr[] [static]

Definition at line 19430 of file chan_sip.c.

struct ast_sockaddr internip [static]

our (internal) default address/port to put in SIP/SDP messages internip is initialized picking a suitable address from one of the interfaces, and the same port number we bind to. It is used as the default address/port in SIP messages, and as the default address (but not port) in SDP messages.

Definition at line 1249 of file chan_sip.c.

Referenced by ast_sip_ouraddrfor(), reload_config(), sip_alloc(), transmit_register(), and transmit_response_using_temp().

struct invstate2stringtable invitestate2string[] [static]

Referenced by show_chanstats_cb().

struct io_context* io [static]

The IO context

Definition at line 853 of file chan_sip.c.

struct ast_ha* localaddr [static]

List of local networks We store "localnet" addresses from the config file into an access list, marked as 'DENY', so the call to ast_apply_ha() will return AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local' (i.e. presumably public) addresses.

List of local networks, on the same side of NAT as this Asterisk

Definition at line 1281 of file chan_sip.c.

int max_expiry = DEFAULT_MAX_EXPIRY [static]

Maximum accepted registration time

Definition at line 610 of file chan_sip.c.

Referenced by handle_request_publish(), parse_register_contact(), and reload_config().

int max_subexpiry = DEFAULT_MAX_EXPIRY [static]

Maximum accepted subscription time

Definition at line 613 of file chan_sip.c.

Referenced by handle_request_subscribe().

struct ast_sockaddr media_address [static]

External RTP IP address if we are behind NAT

Definition at line 1267 of file chan_sip.c.

Referenced by get_our_media_address(), and reload_config().

int min_expiry = DEFAULT_MIN_EXPIRY [static]

Minimum accepted registration time

Definition at line 609 of file chan_sip.c.

Referenced by parse_register_contact(), and reload_config().

int min_subexpiry = DEFAULT_MIN_EXPIRY [static]

Minimum accepted subscription time

Definition at line 612 of file chan_sip.c.

pthread_t monitor_thread = AST_PTHREADT_NULL [static]

This is the thread for the monitor which checks for input on the channels which are not currently in use.

Definition at line 847 of file chan_sip.c.

Referenced by unload_module().

ast_mutex_t monlock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 } [static]

Protect the monitoring thread, so only one process can kill or start it, and not when it's doing something critical.

Definition at line 841 of file chan_sip.c.

Referenced by do_monitor(), restart_monitor(), and unload_module().

int mwi_expiry = DEFAULT_MWI_EXPIRY [static]

Definition at line 614 of file chan_sip.c.

Referenced by __sip_subscribe_mwi_do().

ast_mutex_t netlock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 } [static]

Definition at line 837 of file chan_sip.c.

Referenced by handle_request_do(), and reload_config().

int network_change_event_sched_id = -1 [static]

Definition at line 833 of file chan_sip.c.

struct ast_event_sub* network_change_event_subscription [static]

subscription id for network change events

Definition at line 831 of file chan_sip.c.

const char notify_config[] = "sip_notify.conf" [static]

Configuration file for sending Notify with CLI commands to reconfigure or reboot phones

Definition at line 633 of file chan_sip.c.

struct ast_config* notify_types = NULL [static]

The list of manual NOTIFY types we know how to send

Definition at line 1287 of file chan_sip.c.

int ourport_tcp [static]

The port used for TCP connections

Definition at line 1283 of file chan_sip.c.

int ourport_tls [static]

The port used for TCP/TLS connections

Definition at line 1284 of file chan_sip.c.

struct ao2_container* peers [static]

The peer list: Users, Peers and Friends.

Definition at line 1188 of file chan_sip.c.

Referenced by get_insecure_variable_from_sipregs().

struct ao2_container* peers_by_ip [static]

Definition at line 1189 of file chan_sip.c.

struct ast_data_handler peers_data_provider [static]

Initial value:

 {
   .version = AST_DATA_HANDLER_VERSION,
   .get = peers_data_provider_get
}

Definition at line 34567 of file chan_sip.c.

unsigned int recordhistory [static]

Record SIP history. Off by default

Definition at line 786 of file chan_sip.c.

Referenced by sip_alloc().

struct _map_x_s referstatusstrings[] [static]

Definition at line 868 of file chan_sip.c.

struct ast_register_list regl [static]

Referenced by cleanup_all_regs(), load_module(), manager_show_registry(), reload_config(), sip_register(), sip_send_all_registers(), sip_show_objects(), sip_show_registry(), and unload_module().

int regobjs = 0 [static]

Registry objects

Definition at line 825 of file chan_sip.c.

Referenced by sip_send_all_registers().

struct _map_x_s regstatestrings[] [static]

Definition at line 15066 of file chan_sip.c.

int rpeerobjs = 0 [static]

Realtime peers

Definition at line 823 of file chan_sip.c.

struct ast_sched_context* sched

The scheduling context

Definition at line 852 of file chan_sip.c.

Referenced by ast_rtp_new(), and ast_udptl_new_with_bindaddr().

enum ast_cc_service_type service

Definition at line 891 of file chan_sip.c.

Referenced by cc_build_payload(), cc_generic_monitor_request_cc(), cc_service_to_string(), create_addr(), launch_ha_netscript(), service_string_to_service_type(), sig_pri_handle_subcmds(), and sip_cc_monitor_request_cc().

const char* service_string

Definition at line 892 of file chan_sip.c.

struct ast_cc_agent_callbacks sip_cc_agent_callbacks [static]

Definition at line 1748 of file chan_sip.c.

struct ast_cc_monitor_callbacks sip_cc_monitor_callbacks [static]

Definition at line 2015 of file chan_sip.c.

struct { ... } sip_cc_notify_state_map[] [static]

Referenced by transmit_cc_notify().

struct { ... } sip_cc_service_map[] [static]

Referenced by service_string_to_service_type().

struct sip_settings sip_cfg [static]

SIP configuration data.

Note:

in the future we could have multiple of these (per domain, per device group etc) use this macro when ast_uri_decode is dependent on pedantic checking to be on.

Definition at line 754 of file chan_sip.c.

Referenced by __get_header(), __sip_subscribe_mwi_do(), _sip_show_peer(), add_diversion(), add_header(), ast_sip_ouraddrfor(), build_peer(), change_hold_state(), check_peer_ok(), check_user_full(), create_addr(), destroy_association(), do_magic_pickup(), find_call(), get_also_info(), get_destination(), get_realm(), get_refer_info(), get_sip_pvt_byid_locked(), handle_incoming(), handle_request_bye(), handle_request_do(), handle_request_invite(), handle_request_options(), handle_request_refer(), handle_request_subscribe(), handle_response_invite(), handle_response_peerpoke(), initreqprep(), load_module(), obproxy_get(), parse_register_contact(), parse_uri_legacy_check(), proxy_update(), read_raw_content_length(), realtime_peer(), realtime_update_peer(), receive_message(), ref_proxy(), reg_source_db(), register_peer_exten(), register_verify(), reload_config(), set_peer_defaults(), sip_alloc(), sip_new(), sip_poke_noanswer(), sip_request_call(), sip_set_rtp_peer(), sip_show_settings(), state_notify_build_xml(), transmit_register(), unload_module(), update_call_counter(), and update_peer().

struct ast_threadstorage sip_content_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_sip_content_buf , .custom_init = NULL , } [static]

Definition at line 8303 of file chan_sip.c.

Referenced by get_content().

struct ast_data_entry sip_data_providers[] [static]

Initial value:

 {
   AST_DATA_ENTRY("asterisk/channel/sip/peers", &peers_data_provider),
}

Definition at line 34572 of file chan_sip.c.

struct ast_custom_function sip_header_function [static]

Initial value:

 {
   .name = "SIP_HEADER",
   .read = func_header_read,
}

Definition at line 22082 of file chan_sip.c.

struct cfsip_methods sip_methods[] [static]

Referenced by __sip_autodestruct(), __sip_destroy(), __sip_pretend_ack(), __sip_semi_ack(), build_reply_digest(), check_auth(), dialog_initialize_rtp(), dialog_needdestroy(), do_proxy_auth(), find_call(), find_sip_method(), get_destination(), handle_incoming(), handle_request_bye(), handle_request_invite(), handle_request_register(), handle_request_subscribe(), handle_response(), handle_response_info(), handle_response_message(), init_req(), initialize_initreq(), initreqprep(), method_match(), reqprep(), retrans_pkt(), send_request(), send_response(), sip_alloc(), sip_scheddestroy(), and transmit_register().

struct ao2_container* sip_monitor_instances

Definition at line 1942 of file chan_sip.c.

struct ast_msg_tech sip_msg_tech [static]

Initial value:

 {
   .name = "sip",
   .msg_send = sip_msg_send,
}

Definition at line 26948 of file chan_sip.c.

struct sip_reasons sip_reason_table[] [static]

Referenced by sip_reason_code_to_str(), and sip_reason_str_to_code().

ast_mutex_t sip_reload_lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, 1 } [static]

Definition at line 843 of file chan_sip.c.

Referenced by acl_change_event_cb(), do_monitor(), and sip_reload().

int sip_reloading = FALSE [static]

Flag for avoiding multiple reloads at the same time

Definition at line 849 of file chan_sip.c.

Referenced by do_monitor().

enum channelreloadreason sip_reloadreason [static]

Reason for last reload/load of configuration

Definition at line 850 of file chan_sip.c.

struct ast_rtp_glue sip_rtp_glue [static]

Definition at line 32956 of file chan_sip.c.

struct ast_tcptls_session_args sip_tcp_desc [static]

The TCP server definition.

Definition at line 2302 of file chan_sip.c.

struct ast_channel_tech sip_tech

Definition of this channel for PBX channel registration.

Definition at line 1702 of file chan_sip.c.

struct ast_channel_tech sip_tech_info

This version of the sip channel tech has no send_digit_begin callback so that the core knows that the channel does not want DTMF BEGIN frames. The struct is initialized just before registering the channel driver, and is for use with channels using SIP INFO DTMF.

Definition at line 1736 of file chan_sip.c.

Referenced by load_module(), and sip_new().

struct ast_tls_config sip_tls_cfg [static]

Working TLS connection configuration.

Definition at line 2296 of file chan_sip.c.

struct ast_tcptls_session_args sip_tls_desc [static]

The TCP/TLS server definition.

Definition at line 2313 of file chan_sip.c.

struct ast_threadstorage sip_transport_str_buf = { .once = PTHREAD_ONCE_INIT , .key_init = __init_sip_transport_str_buf , .custom_init = NULL , } [static]

Definition at line 1213 of file chan_sip.c.

Referenced by get_transport_list().

struct ast_udptl_protocol sip_udptl [static]

Initial value:

 {
   .type = "SIP",
   .get_udptl_info = sip_get_udptl_peer,
   .set_udptl_peer = sip_set_udptl_peer,
}

Interface structure with callbacks used to connect to UDPTL module.

Definition at line 3483 of file chan_sip.c.

struct ast_custom_function sipchaninfo_function [static]

Initial value:

 {
   .name = "SIPCHANINFO",
   .read = function_sipchaninfo_read,
}

Structure to declare a dialplan function: SIPCHANINFO.

Definition at line 22283 of file chan_sip.c.

enum sip_debug_e sipdebug [static]

Definition at line 860 of file chan_sip.c.

int sipdebug_text [static]

extra debugging for 'text' related events. At the moment this is set together with sip_debug_console.

Note:

It should either go away or be implemented properly.

Definition at line 866 of file chan_sip.c.

struct ast_custom_function sippeer_function [static]

Initial value:

 {
   .name = "SIPPEER",
   .read = function_sippeer,
}

Structure to declare a dialplan function: SIPPEER.

Definition at line 22212 of file chan_sip.c.

int sipsock = -1 [static]

Main socket for UDP SIP communication.

sipsock is shared between the SIP manager thread (which handles reload requests), the udp io handler (sipsock_read()) and the user routines that issue udp writes (using __sip_xmit()). The socket is -1 only when opening fails (this is a permanent condition), or when we are handling a reload() that changes its address (this is a transient situation during which we might have a harmless race, see below). Because the conditions for the race to be possible are extremely rare, we don't want to pay the cost of locking on every I/O. Rather, we remember that when the race may occur, communication is bound to fail anyways, so we just live with this event and let the protocol handle this above us.

Definition at line 1239 of file chan_sip.c.

Referenced by sip_prepare_socket(), and sipsock_read().

int* sipsock_read_id [static]

ID of IO entry for sipsock FD

Definition at line 854 of file chan_sip.c.

int speerobjs = 0 [static]

Static peers

Definition at line 822 of file chan_sip.c.

enum sip_cc_notify_state state

Definition at line 912 of file chan_sip.c.

Referenced by action_presencestate(), aji_publish_device_state(), alsa_read(), alsa_write(), ast_async_goto(), ast_bridge_join(), ast_channel_get_t38_state(), ast_channel_start_silence_generator(), ast_devstate_aggregate_add(), ast_devstate_str(), ast_extension_state3(), ast_monitor_set_state(), ast_speech_change_state(), bc_next_state_change(), bc_state_change(), bridge_channel_join(), bridge_channel_thread(), calendarstate(), cc_request_state_change(), cc_state_to_devstate(), cc_state_to_string(), change_t38_state(), complete_meetmecmd_list(), complete_meetmecmd_mute_kick(), custom_presence_callback(), destroy_all_channels(), device_state_cb(), devstate_change_collector_cb(), do_presence_state_change(), do_state_change(), extensionstate2devicestate(), feature_request_and_dial(), gen_alloc(), handle_cli_devstate_change(), handle_cli_presencestate_change(), handle_cli_presencestate_list(), handle_logger_set_level(), handle_statechange(), initialize_cc_devstate_map_helper(), jingle_endpoint_state_create(), load_module(), misdn_get_ch_state(), mkintf(), moh_alloc(), moh_files_alloc(), moh_files_release(), moh_release(), poll_mailbox(), presence_read(), presence_write(), process_collection(), set_state(), setsubstate(), sip_queryoption(), skinny_extensionstate_cb(), sla_change_trunk_state(), update_state(), xmpp_client_change_state(), xmpp_pubsub_publish_device_state(), and xmpp_show_clients().

const char* state_string

Definition at line 913 of file chan_sip.c.

Referenced by transmit_cc_notify().

struct _map_x_s stmodes[] [static]

Report Peer status in character string.

Returns:

0 if peer is unreachable, 1 if peer is online, -1 if unmonitored

Definition at line 18818 of file chan_sip.c.

struct _map_x_s strefresher_params[] [static]

Definition at line 18836 of file chan_sip.c.

struct _map_x_s strefreshers[] [static]

Definition at line 18843 of file chan_sip.c.

struct ast_subscription_mwi_list submwil [static]

Referenced by load_module(), sip_send_all_mwi_subscriptions(), sip_show_mwi(), sip_subscribe_mwi(), and unload_module().

struct cfsubscription_types subscription_types[] [static]

Referenced by find_subscription_type(), and subscription_type2str().

struct ao2_container* threadt [static]

The table of TCP threads.

Definition at line 1185 of file chan_sip.c.

struct _map_x_s trust_id_outboundstr[] [static]

Definition at line 19457 of file chan_sip.c.

struct ast_threadstorage ts_temp_pvt = { .once = PTHREAD_ONCE_INIT , .key_init = __init_ts_temp_pvt , .custom_init = temp_pvt_init , } [static]

Definition at line 1210 of file chan_sip.c.

Referenced by transmit_response_using_temp().

int unauth_sessions = 0 [static]

Definition at line 616 of file chan_sip.c.

char used_context[AST_MAX_CONTEXT] [static]

name of automatically created context for unloading

Definition at line 835 of file chan_sip.c.