dial() & retrydial() - Trivial application to dial a channel and send an URL on answer More...
#include "asterisk.h"#include <sys/time.h>#include <sys/signal.h>#include <sys/stat.h>#include <netinet/in.h>#include "asterisk/paths.h"#include "asterisk/lock.h"#include "asterisk/file.h"#include "asterisk/channel.h"#include "asterisk/pbx.h"#include "asterisk/module.h"#include "asterisk/translate.h"#include "asterisk/say.h"#include "asterisk/config.h"#include "asterisk/features.h"#include "asterisk/musiconhold.h"#include "asterisk/callerid.h"#include "asterisk/utils.h"#include "asterisk/app.h"#include "asterisk/causes.h"#include "asterisk/rtp_engine.h"#include "asterisk/cdr.h"#include "asterisk/manager.h"#include "asterisk/privacy.h"#include "asterisk/stringfields.h"#include "asterisk/global_datastores.h"#include "asterisk/dsp.h"#include "asterisk/cel.h"#include "asterisk/aoc.h"#include "asterisk/ccss.h"#include "asterisk/indications.h"#include "asterisk/framehook.h"
Go to the source code of this file.
Data Structures | |
| struct | cause_args |
| struct | chanlist |
| List of channel drivers. More... | |
| struct | dial_head |
| struct | privacy_args |
Defines | |
| #define | AST_MAX_WATCHERS 256 |
| #define | CAN_EARLY_BRIDGE(flags, chan, peer) |
| #define | DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */ |
| #define | DIAL_NOFORWARDHTML (1LLU << 32) |
| #define | DIAL_STILLGOING (1LLU << 31) |
| #define | OPT_CALLEE_GO_ON (1LLU << 36) |
| #define | OPT_CALLER_ANSWER (1LLU << 40) |
| #define | OPT_CANCEL_ELSEWHERE (1LLU << 34) |
| #define | OPT_CANCEL_TIMEOUT (1LLU << 37) |
| #define | OPT_FORCE_CID_PRES (1LLU << 39) |
| #define | OPT_FORCE_CID_TAG (1LLU << 38) |
| #define | OPT_PEER_H (1LLU << 35) |
| #define | OPT_PREDIAL_CALLEE (1LLU << 41) |
| #define | OPT_PREDIAL_CALLER (1LLU << 42) |
Enumerations | |
| enum | { OPT_ANNOUNCE = (1 << 0), OPT_RESETCDR = (1 << 1), OPT_DTMF_EXIT = (1 << 2), OPT_SENDDTMF = (1 << 3), OPT_FORCECLID = (1 << 4), OPT_GO_ON = (1 << 5), OPT_CALLEE_HANGUP = (1 << 6), OPT_CALLER_HANGUP = (1 << 7), OPT_ORIGINAL_CLID = (1 << 8), OPT_DURATION_LIMIT = (1 << 9), OPT_MUSICBACK = (1 << 10), OPT_CALLEE_MACRO = (1 << 11), OPT_SCREEN_NOINTRO = (1 << 12), OPT_SCREEN_NOCALLERID = (1 << 13), OPT_IGNORE_CONNECTEDLINE = (1 << 14), OPT_SCREENING = (1 << 15), OPT_PRIVACY = (1 << 16), OPT_RINGBACK = (1 << 17), OPT_DURATION_STOP = (1 << 18), OPT_CALLEE_TRANSFER = (1 << 19), OPT_CALLER_TRANSFER = (1 << 20), OPT_CALLEE_MONITOR = (1 << 21), OPT_CALLER_MONITOR = (1 << 22), OPT_GOTO = (1 << 23), OPT_OPERMODE = (1 << 24), OPT_CALLEE_PARK = (1 << 25), OPT_CALLER_PARK = (1 << 26), OPT_IGNORE_FORWARDING = (1 << 27), OPT_CALLEE_GOSUB = (1 << 28), OPT_CALLEE_MIXMONITOR = (1 << 29), OPT_CALLER_MIXMONITOR = (1 << 30) } |
| enum | { OPT_ARG_ANNOUNCE = 0, OPT_ARG_SENDDTMF, OPT_ARG_GOTO, OPT_ARG_DURATION_LIMIT, OPT_ARG_MUSICBACK, OPT_ARG_CALLEE_MACRO, OPT_ARG_RINGBACK, OPT_ARG_CALLEE_GOSUB, OPT_ARG_CALLEE_GO_ON, OPT_ARG_PRIVACY, OPT_ARG_DURATION_STOP, OPT_ARG_OPERMODE, OPT_ARG_SCREEN_NOINTRO, OPT_ARG_ORIGINAL_CLID, OPT_ARG_FORCECLID, OPT_ARG_FORCE_CID_TAG, OPT_ARG_FORCE_CID_PRES, OPT_ARG_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLER, OPT_ARG_ARRAY_SIZE } |
Functions | |
| static void | __reg_module (void) |
| static void | __unreg_module (void) |
| static void | chanlist_free (struct chanlist *outgoing) |
| static int | detect_disconnect (struct ast_channel *chan, char code, struct ast_str **featurecode) |
| static int | dial_exec (struct ast_channel *chan, const char *data) |
| static int | dial_exec_full (struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec) |
| static int | dial_handle_playtones (struct ast_channel *chan, const char *data) |
| static void | do_forward (struct chanlist *o, struct cause_args *num, struct ast_flags64 *peerflags, int single, int caller_entertained, int *to, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid) |
| static void | end_bridge_callback (void *data) |
| static void | end_bridge_callback_data_fixup (struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) |
| static const char * | get_cid_name (char *name, int namelen, struct ast_channel *chan) |
| static void | handle_cause (int cause, struct cause_args *num) |
| static void | hanguptree (struct dial_head *out_chans, struct ast_channel *exception, int answered_elsewhere) |
| static int | load_module (void) |
| static int | onedigit_goto (struct ast_channel *chan, const char *context, char exten, int pri) |
| static int | retrydial_exec (struct ast_channel *chan, const char *data) |
| static void | senddialendevent (struct ast_channel *src, const char *dialstatus) |
| static void | senddialevent (struct ast_channel *src, struct ast_channel *dst, const char *dialstring) |
| static int | setup_privacy_args (struct privacy_args *pa, struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan) |
| returns 1 if successful, 0 or <0 if the caller should 'goto out' | |
| static int | unload_module (void) |
| static int | valid_priv_reply (struct ast_flags64 *opts, int res) |
| static struct ast_channel * | wait_for_answer (struct ast_channel *in, struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags, char *opt_args[], struct privacy_args *pa, const struct cause_args *num_in, int *result, char *dtmf_progress, const int ignore_cc, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid) |
Variables | |
| static struct ast_module_info | __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Dialing Application" , .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, .load_pri = AST_MODPRI_DEFAULT, } |
| static const char | app [] = "Dial" |
| static struct ast_module_info * | ast_module_info = &__mod_info |
| static struct ast_app_option | dial_exec_options [128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'a' ] = { .flag = (1LLU << 40) }, [ 'b' ] = { .flag = (1LLU << 41) , .arg_index = OPT_ARG_PREDIAL_CALLEE + 1 }, [ 'B' ] = { .flag = (1LLU << 42) , .arg_index = OPT_ARG_PREDIAL_CALLER + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = (1LLU << 34) }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'e' ] = { .flag = (1LLU << 35) }, [ 'f' ] = { .flag = OPT_FORCECLID , .arg_index = OPT_ARG_FORCECLID + 1 }, [ 'F' ] = { .flag = (1LLU << 36) , .arg_index = OPT_ARG_CALLEE_GO_ON + 1 }, [ 'g' ] = { .flag = OPT_GO_ON }, [ 'G' ] = { .flag = OPT_GOTO , .arg_index = OPT_ARG_GOTO + 1 }, [ 'h' ] = { .flag = OPT_CALLEE_HANGUP }, [ 'H' ] = { .flag = OPT_CALLER_HANGUP }, [ 'i' ] = { .flag = OPT_IGNORE_FORWARDING }, [ 'I' ] = { .flag = OPT_IGNORE_CONNECTEDLINE }, [ 'k' ] = { .flag = OPT_CALLEE_PARK }, [ 'K' ] = { .flag = OPT_CALLER_PARK }, [ 'L' ] = { .flag = OPT_DURATION_LIMIT , .arg_index = OPT_ARG_DURATION_LIMIT + 1 }, [ 'm' ] = { .flag = OPT_MUSICBACK , .arg_index = OPT_ARG_MUSICBACK + 1 }, [ 'M' ] = { .flag = OPT_CALLEE_MACRO , .arg_index = OPT_ARG_CALLEE_MACRO + 1 }, [ 'n' ] = { .flag = OPT_SCREEN_NOINTRO , .arg_index = OPT_ARG_SCREEN_NOINTRO + 1 }, [ 'N' ] = { .flag = OPT_SCREEN_NOCALLERID }, [ 'o' ] = { .flag = OPT_ORIGINAL_CLID , .arg_index = OPT_ARG_ORIGINAL_CLID + 1 }, [ 'O' ] = { .flag = OPT_OPERMODE , .arg_index = OPT_ARG_OPERMODE + 1 }, [ 'p' ] = { .flag = OPT_SCREENING }, [ 'P' ] = { .flag = OPT_PRIVACY , .arg_index = OPT_ARG_PRIVACY + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK , .arg_index = OPT_ARG_RINGBACK + 1 }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 's' ] = { .flag = (1LLU << 38) , .arg_index = OPT_ARG_FORCE_CID_TAG + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'u' ] = { .flag = (1LLU << 39) , .arg_index = OPT_ARG_FORCE_CID_PRES + 1 }, [ 'U' ] = { .flag = OPT_CALLEE_GOSUB , .arg_index = OPT_ARG_CALLEE_GOSUB + 1 }, [ 'w' ] = { .flag = OPT_CALLEE_MONITOR }, [ 'W' ] = { .flag = OPT_CALLER_MONITOR }, [ 'x' ] = { .flag = OPT_CALLEE_MIXMONITOR }, [ 'X' ] = { .flag = OPT_CALLER_MIXMONITOR }, [ 'z' ] = { .flag = (1LLU << 37) }, } |
| static const char | rapp [] = "RetryDial" |
dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
Definition in file app_dial.c.
| #define AST_MAX_WATCHERS 256 |
Definition at line 738 of file app_dial.c.
Referenced by wait_for_answer().
| #define CAN_EARLY_BRIDGE | ( | flags, | |
| chan, | |||
| peer | |||
| ) |
Definition at line 680 of file app_dial.c.
Referenced by dial_exec_full(), do_forward(), and wait_for_answer().
| #define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */ |
Definition at line 602 of file app_dial.c.
Referenced by dial_exec_full(), do_forward(), and wait_for_answer().
| #define DIAL_NOFORWARDHTML (1LLU << 32) |
Definition at line 601 of file app_dial.c.
Referenced by dial_exec_full(), and wait_for_answer().
| #define DIAL_STILLGOING (1LLU << 31) |
Definition at line 600 of file app_dial.c.
Referenced by dial_exec_full(), do_forward(), and wait_for_answer().
| #define OPT_CALLEE_GO_ON (1LLU << 36) |
Definition at line 605 of file app_dial.c.
Referenced by bridge_exec(), dial_exec_full(), and try_calling().
| #define OPT_CALLER_ANSWER (1LLU << 40) |
Definition at line 609 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_CANCEL_ELSEWHERE (1LLU << 34) |
Definition at line 603 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_CANCEL_TIMEOUT (1LLU << 37) |
Definition at line 606 of file app_dial.c.
Referenced by dial_exec_full(), and do_forward().
| #define OPT_FORCE_CID_PRES (1LLU << 39) |
Definition at line 608 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_FORCE_CID_TAG (1LLU << 38) |
Definition at line 607 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_PEER_H (1LLU << 35) |
Definition at line 604 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_PREDIAL_CALLEE (1LLU << 41) |
Definition at line 610 of file app_dial.c.
Referenced by dial_exec_full().
| #define OPT_PREDIAL_CALLER (1LLU << 42) |
Definition at line 611 of file app_dial.c.
Referenced by dial_exec_full().
| anonymous enum |
Definition at line 565 of file app_dial.c.
{
OPT_ANNOUNCE = (1 << 0),
OPT_RESETCDR = (1 << 1),
OPT_DTMF_EXIT = (1 << 2),
OPT_SENDDTMF = (1 << 3),
OPT_FORCECLID = (1 << 4),
OPT_GO_ON = (1 << 5),
OPT_CALLEE_HANGUP = (1 << 6),
OPT_CALLER_HANGUP = (1 << 7),
OPT_ORIGINAL_CLID = (1 << 8),
OPT_DURATION_LIMIT = (1 << 9),
OPT_MUSICBACK = (1 << 10),
OPT_CALLEE_MACRO = (1 << 11),
OPT_SCREEN_NOINTRO = (1 << 12),
OPT_SCREEN_NOCALLERID = (1 << 13),
OPT_IGNORE_CONNECTEDLINE = (1 << 14),
OPT_SCREENING = (1 << 15),
OPT_PRIVACY = (1 << 16),
OPT_RINGBACK = (1 << 17),
OPT_DURATION_STOP = (1 << 18),
OPT_CALLEE_TRANSFER = (1 << 19),
OPT_CALLER_TRANSFER = (1 << 20),
OPT_CALLEE_MONITOR = (1 << 21),
OPT_CALLER_MONITOR = (1 << 22),
OPT_GOTO = (1 << 23),
OPT_OPERMODE = (1 << 24),
OPT_CALLEE_PARK = (1 << 25),
OPT_CALLER_PARK = (1 << 26),
OPT_IGNORE_FORWARDING = (1 << 27),
OPT_CALLEE_GOSUB = (1 << 28),
OPT_CALLEE_MIXMONITOR = (1 << 29),
OPT_CALLER_MIXMONITOR = (1 << 30),
};
| anonymous enum |
Definition at line 613 of file app_dial.c.
{
OPT_ARG_ANNOUNCE = 0,
OPT_ARG_SENDDTMF,
OPT_ARG_GOTO,
OPT_ARG_DURATION_LIMIT,
OPT_ARG_MUSICBACK,
OPT_ARG_CALLEE_MACRO,
OPT_ARG_RINGBACK,
OPT_ARG_CALLEE_GOSUB,
OPT_ARG_CALLEE_GO_ON,
OPT_ARG_PRIVACY,
OPT_ARG_DURATION_STOP,
OPT_ARG_OPERMODE,
OPT_ARG_SCREEN_NOINTRO,
OPT_ARG_ORIGINAL_CLID,
OPT_ARG_FORCECLID,
OPT_ARG_FORCE_CID_TAG,
OPT_ARG_FORCE_CID_PRES,
OPT_ARG_PREDIAL_CALLEE,
OPT_ARG_PREDIAL_CALLER,
/* note: this entry _MUST_ be the last one in the enum */
OPT_ARG_ARRAY_SIZE,
};
| static void __reg_module | ( | void | ) | [static] |
Definition at line 3261 of file app_dial.c.
| static void __unreg_module | ( | void | ) | [static] |
Definition at line 3261 of file app_dial.c.
| static void chanlist_free | ( | struct chanlist * | outgoing | ) | [static] |
Definition at line 713 of file app_dial.c.
References ast_aoc_destroy_decoded(), ast_free, and ast_party_connected_line_free().
Referenced by dial_exec_full(), and hanguptree().
{
ast_party_connected_line_free(&outgoing->connected);
ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
ast_free(outgoing);
}
| static int detect_disconnect | ( | struct ast_channel * | chan, |
| char | code, | ||
| struct ast_str ** | featurecode | ||
| ) | [static] |
Definition at line 1727 of file app_dial.c.
References ast_feature_detect(), AST_FEATURE_DISCONNECT, AST_FEATURE_RETURN_STOREDIGITS, ast_str_append(), ast_str_buffer(), ast_str_reset(), and ast_call_feature::feature_mask.
Referenced by wait_for_answer().
{
struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
struct ast_call_feature feature = { 0, };
int res;
ast_str_append(featurecode, 1, "%c", code);
res = ast_feature_detect(chan, &features, ast_str_buffer(*featurecode), &feature);
if (res != AST_FEATURE_RETURN_STOREDIGITS) {
ast_str_reset(*featurecode);
}
if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
return 1;
}
return 0;
}
| static int dial_exec | ( | struct ast_channel * | chan, |
| const char * | data | ||
| ) | [static] |
Definition at line 3124 of file app_dial.c.
References dial_exec_full().
Referenced by load_module().
{
struct ast_flags64 peerflags;
memset(&peerflags, 0, sizeof(peerflags));
return dial_exec_full(chan, data, &peerflags, NULL);
}
| static int dial_exec_full | ( | struct ast_channel * | chan, |
| const char * | data, | ||
| struct ast_flags64 * | peerflags, | ||
| int * | continue_exec | ||
| ) | [static] |
< TRUE if force CallerID on call forward only. Legacy behaviour.
Forced CallerID party information to send.
Stored CallerID information if needed.
CallerID party information to store.
Definition at line 2055 of file app_dial.c.
References args, ast_answer(), AST_APP_ARG, ast_app_exec_macro(), ast_app_exec_sub(), ast_app_expand_sub_args(), ast_app_group_set_channel(), ast_app_parse_options64(), ast_asprintf, ast_autoservice_chan_hangup_peer(), ast_autoservice_start(), ast_autoservice_stop(), ast_bridge_call(), ast_bridge_timelimit(), ast_call(), ast_callerid_parse(), ast_calloc, ast_cause2str(), AST_CAUSE_ANSWERED_ELSEWHERE, AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, ast_cc_busy_interface(), ast_cc_call_failed(), ast_cc_call_init(), ast_cc_callback(), ast_cc_extension_monitor_add_dialstring(), AST_CDR_FLAG_DIALED, ast_cdr_reset(), ast_cdr_setanswer(), ast_cdr_setdestchan(), ast_channel_accountcode(), ast_channel_adsicpe_set(), ast_channel_appl_set(), ast_channel_caller(), ast_channel_cdr(), ast_channel_connected(), ast_channel_context(), ast_channel_context_set(), ast_channel_data_set(), ast_channel_datastore_add(), ast_channel_datastore_find(), ast_channel_datastore_inherit(), ast_channel_datastore_remove(), ast_channel_dialed(), ast_channel_early_bridge(), ast_channel_exten(), ast_channel_exten_set(), ast_channel_flags(), ast_channel_get_device_name(), ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_inherit_variables(), ast_channel_language(), ast_channel_lock, ast_channel_lock_both, ast_channel_macrocontext(), ast_channel_macroexten(), ast_channel_make_compatible(), ast_channel_musicclass(), AST_CHANNEL_NAME, ast_channel_name(), ast_channel_nativeformats(), ast_channel_priority(), ast_channel_priority_set(), ast_channel_redirecting(), ast_channel_sched(), ast_channel_sendurl(), ast_channel_set_caller_event(), ast_channel_set_connected_line(), ast_channel_setoption(), ast_channel_stream(), ast_channel_supports_html(), ast_channel_timingfunc(), ast_channel_transfercapability(), ast_channel_transfercapability_set(), ast_channel_unlock, ast_channel_visible_indication_set(), ast_channel_whentohangup(), ast_channel_whentohangup_set(), ast_check_hangup(), ast_clear_flag, ast_connected_line_copy_from_caller(), AST_CONTROL_HANGUP, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_copy_flags64, ast_copy_string(), ast_datastore_alloc(), ast_datastore_free(), ast_deactivate_generator(), ast_debug, AST_DECLARE_APP_ARGS, AST_DIGIT_ANY, ast_dtmf_stream(), ast_exists_extension(), AST_FEATURE_AUTOMIXMON, AST_FEATURE_AUTOMON, AST_FEATURE_DISCONNECT, AST_FEATURE_NO_H_EXTEN, AST_FEATURE_PARKCALL, AST_FEATURE_REDIRECT, ast_filedelete(), ast_fileexists(), AST_FLAG_END_DTMF_ONLY, AST_FRAME_CONTROL, AST_FRAME_DTMF_END, ast_free, ast_frfree, ast_goto_if_exists(), ast_hangup(), ast_ignore_cc(), ast_indicate(), AST_LIST_EMPTY, AST_LIST_FIRST, AST_LIST_HEAD, AST_LIST_HEAD_INIT, AST_LIST_HEAD_NOLOCK_INIT_VALUE, AST_LIST_INSERT_TAIL, AST_LIST_LOCK, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, AST_LIST_UNLOCK, ast_log(), AST_MAX_EXTENSION, ast_moh_start(), ast_moh_stop(), AST_OPTION_OPRMODE, ast_parse_caller_presentation(), ast_parseable_goto(), ast_party_caller_set_init(), ast_party_connected_line_copy(), ast_party_connected_line_set_init(), ast_party_id_init(), ast_party_id_set_init(), ast_party_redirecting_copy(), ast_pbx_h_exten_run(), AST_PBX_INCOMPLETE, ast_pbx_start(), ast_pre_call(), AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN, AST_PRIVACY_UNKNOWN, ast_read(), ast_replace_subargument_delimiter(), ast_request(), ast_rtp_instance_early_bridge_make_compatible(), ast_sched_runq(), ast_sched_wait(), ast_senddigit(), ast_set2_flag64, ast_set_flag, ast_set_flag64, AST_STANDARD_APP_ARGS, AST_STATE_UP, ast_stopstream(), ast_streamfile(), ast_strlen_zero(), ast_test_flag64, ast_tvadd(), ast_tvnow(), ast_tvzero(), ast_verb, ast_waitfor_n(), CAN_EARLY_BRIDGE, chanlist::chan, chanlist_free(), chanlist::connected, ast_datastore::data, DATASTORE_INHERIT_FOREVER, di, DIAL_CALLERID_ABSENT, dial_exec_options, dial_handle_playtones(), DIAL_NOFORWARDHTML, DIAL_STILLGOING, dialed_interface_info, ast_bridge_config::end_bridge_callback, end_bridge_callback(), ast_bridge_config::end_bridge_callback_data, ast_bridge_config::end_bridge_callback_data_fixup, end_bridge_callback_data_fixup(), ast_bridge_config::end_sound, ast_bridge_config::features_callee, ast_bridge_config::features_caller, ast_flags64::flags, ast_frame::frametype, get_cid_name(), handle_cause(), hanguptree(), ast_party_caller::id, ast_party_connected_line::id, ast_datastore::inheritance, ast_frame_subclass::integer, ast_dialed_interface::interface, chanlist::interface, chanlist::list, LOG_ERROR, LOG_NOTICE, LOG_WARNING, oprmode::mode, moh, name, ast_party_id::name, chanlist::node, ast_party_id::number, chanlist::number, OPT_ANNOUNCE, OPT_ARG_ANNOUNCE, OPT_ARG_ARRAY_SIZE, OPT_ARG_CALLEE_GO_ON, OPT_ARG_CALLEE_GOSUB, OPT_ARG_CALLEE_MACRO, OPT_ARG_DURATION_LIMIT, OPT_ARG_DURATION_STOP, OPT_ARG_FORCE_CID_PRES, OPT_ARG_FORCE_CID_TAG, OPT_ARG_FORCECLID, OPT_ARG_GOTO, OPT_ARG_OPERMODE, OPT_ARG_ORIGINAL_CLID, OPT_ARG_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLER, OPT_ARG_PRIVACY, OPT_ARG_RINGBACK, OPT_ARG_SCREEN_NOINTRO, OPT_ARG_SENDDTMF, OPT_CALLEE_GO_ON, OPT_CALLEE_GOSUB, OPT_CALLEE_HANGUP, OPT_CALLEE_MACRO, OPT_CALLEE_MIXMONITOR, OPT_CALLEE_MONITOR, OPT_CALLEE_PARK, OPT_CALLEE_TRANSFER, OPT_CALLER_ANSWER, OPT_CALLER_HANGUP, OPT_CALLER_MIXMONITOR, OPT_CALLER_MONITOR, OPT_CALLER_PARK, OPT_CALLER_TRANSFER, OPT_CANCEL_ELSEWHERE, OPT_CANCEL_TIMEOUT, OPT_DTMF_EXIT, OPT_DURATION_LIMIT, OPT_DURATION_STOP, OPT_FORCE_CID_PRES, OPT_FORCE_CID_TAG, OPT_FORCECLID, OPT_GO_ON, OPT_GOTO, OPT_IGNORE_CONNECTEDLINE, OPT_IGNORE_FORWARDING, OPT_MUSICBACK, OPT_OPERMODE, OPT_ORIGINAL_CLID, OPT_PEER_H, OPT_PREDIAL_CALLEE, OPT_PREDIAL_CALLER, OPT_PRIVACY, OPT_RESETCDR, OPT_RINGBACK, OPT_SCREEN_NOINTRO, OPT_SCREENING, OPT_SENDDTMF, parse(), pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), oprmode::peer, ast_party_number::presentation, privacy_args::privdb_val, privacy_args::privintro, S_COR, S_OR, senddialendevent(), senddialevent(), privacy_args::sentringing, setup_privacy_args(), ast_bridge_config::start_sound, privacy_args::status, ast_party_name::str, ast_party_number::str, ast_party_subaddress::str, chanlist::stuff, ast_party_id::subaddress, ast_frame::subclass, ast_party_id::tag, chanlist::tech, ast_party_dialed::transit_network_select, url, ast_party_name::valid, ast_party_number::valid, wait_for_answer(), and ast_bridge_config::warning_sound.
Referenced by dial_exec(), and retrydial_exec().
{
int res = -1; /* default: error */
char *rest, *cur; /* scan the list of destinations */
struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
struct chanlist *outgoing;
struct chanlist *tmp;
struct ast_channel *peer;
int to; /* timeout */
struct cause_args num = { chan, 0, 0, 0 };
int cause;
struct ast_bridge_config config = { { 0, } };
struct timeval calldurationlimit = { 0, };
char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
struct privacy_args pa = {
.sentringing = 0,
.privdb_val = 0,
.status = "INVALIDARGS",
};
int sentringing = 0, moh = 0;
const char *outbound_group = NULL;
int result = 0;
char *parse;
int opermode = 0;
int delprivintro = 0;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(peers);
AST_APP_ARG(timeout);
AST_APP_ARG(options);
AST_APP_ARG(url);
);
struct ast_flags64 opts = { 0, };
char *opt_args[OPT_ARG_ARRAY_SIZE];
struct ast_datastore *datastore = NULL;
int fulldial = 0, num_dialed = 0;
int ignore_cc = 0;
char device_name[AST_CHANNEL_NAME];
char forced_clid_name[AST_MAX_EXTENSION];
char stored_clid_name[AST_MAX_EXTENSION];
int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
/*!
* \brief Forced CallerID party information to send.
* \note This will not have any malloced strings so do not free it.
*/
struct ast_party_id forced_clid;
/*!
* \brief Stored CallerID information if needed.
*
* \note If OPT_ORIGINAL_CLID set then this is the o option
* CallerID. Otherwise it is the dialplan extension and hint
* name.
*
* \note This will not have any malloced strings so do not free it.
*/
struct ast_party_id stored_clid;
/*!
* \brief CallerID party information to store.
* \note This will not have any malloced strings so do not free it.
*/
struct ast_party_caller caller;
/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.options) &&
ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
}
if (ast_strlen_zero(args.peers)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
}
if (ast_cc_call_init(chan, &ignore_cc)) {
goto done;
}
if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
if (delprivintro < 0 || delprivintro > 1) {
ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
delprivintro = 0;
}
}
if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
opt_args[OPT_ARG_RINGBACK] = NULL;
}
if (ast_test_flag64(&opts, OPT_OPERMODE)) {
opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
ast_verb(3, "Setting operator services mode to %d.\n", opermode);
}
if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
if (!calldurationlimit.tv_sec) {
ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
}
ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
}
if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
dtmfcalled = strsep(&dtmf_progress, ":");
dtmfcalling = strsep(&dtmf_progress, ":");
}
if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
goto done;
}
/* Setup the forced CallerID information to send if used. */
ast_party_id_init(&forced_clid);
force_forwards_only = 0;
if (ast_test_flag64(&opts, OPT_FORCECLID)) {
if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
ast_channel_lock(chan);
forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
ast_channel_unlock(chan);
forced_clid_name[0] = '\0';
forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
sizeof(forced_clid_name), chan);
force_forwards_only = 1;
} else {
/* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
&forced_clid.number.str);
}
if (!ast_strlen_zero(forced_clid.name.str)) {
forced_clid.name.valid = 1;
}
if (!ast_strlen_zero(forced_clid.number.str)) {
forced_clid.number.valid = 1;
}
}
if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
&& !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
}
forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
&& !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
int pres;
pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
if (0 <= pres) {
forced_clid.number.presentation = pres;
}
}
/* Setup the stored CallerID information if needed. */
ast_party_id_init(&stored_clid);
if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
ast_channel_lock(chan);
ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
}
if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
}
if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
}
if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
}
ast_channel_unlock(chan);
} else {
/* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
&stored_clid.number.str);
if (!ast_strlen_zero(stored_clid.name.str)) {
stored_clid.name.valid = 1;
}
if (!ast_strlen_zero(stored_clid.number.str)) {
stored_clid.number.valid = 1;
}
}
} else {
/*
* In case the new channel has no preset CallerID number by the
* channel driver, setup the dialplan extension and hint name.
*/
stored_clid_name[0] = '\0';
stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
sizeof(stored_clid_name), chan);
if (ast_strlen_zero(stored_clid.name.str)) {
stored_clid.name.str = NULL;
} else {
stored_clid.name.valid = 1;
}
ast_channel_lock(chan);
stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
stored_clid.number.valid = 1;
ast_channel_unlock(chan);
}
if (ast_test_flag64(&opts, OPT_RESETCDR) && ast_channel_cdr(chan))
ast_cdr_reset(ast_channel_cdr(chan), NULL);
if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
res = setup_privacy_args(&pa, &opts, opt_args, chan);
if (res <= 0)
goto out;
res = -1; /* reset default */
}
if (continue_exec)
*continue_exec = 0;
/* If a channel group has been specified, get it for use when we create peer channels */
ast_channel_lock(chan);
if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
outbound_group = ast_strdupa(outbound_group);
pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
} else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
outbound_group = ast_strdupa(outbound_group);
}
ast_channel_unlock(chan);
/* Set per dial instance flags. These flags are also passed back to RetryDial. */
ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
| OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
| OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
/* PREDIAL: Run gosub on the caller's channel */
if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
&& !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
}
/* loop through the list of dial destinations */
rest = args.peers;
while ((cur = strsep(&rest, "&")) ) {
struct ast_channel *tc; /* channel for this destination */
/* Get a technology/resource pair */
char *number = cur;
char *tech = strsep(&number, "/");
size_t tech_len;
size_t number_len;
/* find if we already dialed this interface */
struct ast_dialed_interface *di;
AST_LIST_HEAD(,ast_dialed_interface) *dialed_interfaces;
num_dialed++;
if (ast_strlen_zero(number)) {
ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
goto out;
}
tech_len = strlen(tech) + 1;
number_len = strlen(number) + 1;
tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
if (!tmp) {
goto out;
}
/* Save tech, number, and interface. */
cur = tmp->stuff;
strcpy(cur, tech);
tmp->tech = cur;
cur += tech_len;
strcpy(cur, tech);
cur[tech_len - 1] = '/';
tmp->interface = cur;
cur += tech_len;
strcpy(cur, number);
tmp->number = cur;
if (opts.flags) {
/* Set per outgoing call leg options. */
ast_copy_flags64(tmp, &opts,
OPT_CANCEL_ELSEWHERE |
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE);
ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
}
/* Request the peer */
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
/*
* Seed the chanlist's connected line information with previously
* acquired connected line info from the incoming channel. The
* previously acquired connected line info could have been set
* through the CONNECTED_LINE dialplan function.
*/
ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
ast_channel_unlock(chan);
if (datastore)
dialed_interfaces = datastore->data;
else {
if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
chanlist_free(tmp);
goto out;
}
datastore->inheritance = DATASTORE_INHERIT_FOREVER;
if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
ast_datastore_free(datastore);
chanlist_free(tmp);
goto out;
}
datastore->data = dialed_interfaces;
AST_LIST_HEAD_INIT(dialed_interfaces);
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
}
AST_LIST_LOCK(dialed_interfaces);
AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
if (!strcasecmp(di->interface, tmp->interface)) {
ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
di->interface);
break;
}
}
AST_LIST_UNLOCK(dialed_interfaces);
if (di) {
fulldial++;
chanlist_free(tmp);
continue;
}
/* It is always ok to dial a Local interface. We only keep track of
* which "real" interfaces have been dialed. The Local channel will
* inherit this list so that if it ends up dialing a real interface,
* it won't call one that has already been called. */
if (strcasecmp(tmp->tech, "Local")) {
if (!(di = ast_calloc(1, sizeof(*di) + strlen(tmp->interface)))) {
chanlist_free(tmp);
goto out;
}
strcpy(di->interface, tmp->interface);
AST_LIST_LOCK(dialed_interfaces);
AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
AST_LIST_UNLOCK(dialed_interfaces);
}
tc = ast_request(tmp->tech, ast_channel_nativeformats(chan), chan, tmp->number, &cause);
if (!tc) {
/* If we can't, just go on to the next call */
ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
tmp->tech, cause, ast_cause2str(cause));
handle_cause(cause, &num);
if (!rest) {
/* we are on the last destination */
ast_channel_hangupcause_set(chan, cause);
}
if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
}
}
chanlist_free(tmp);
continue;
}
ast_channel_get_device_name(tc, device_name, sizeof(device_name));
if (!ignore_cc) {
ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
}
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
ast_channel_lock_both(tc, chan);
/* Setup outgoing SDP to match incoming one */
if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
/* We are on the only destination. */
ast_rtp_instance_early_bridge_make_compatible(tc, chan);
}
/* Inherit specially named variables from parent channel */
ast_channel_inherit_variables(chan, tc);
ast_channel_datastore_inherit(chan, tc);
ast_channel_appl_set(tc, "AppDial");
ast_channel_data_set(tc, "(Outgoing Line)");
memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
/* Determine CallerID to store in outgoing channel. */
ast_party_caller_set_init(&caller, ast_channel_caller(tc));
if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
caller.id = stored_clid;
ast_channel_set_caller_event(tc, &caller, NULL);
ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
} else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
ast_channel_caller(tc)->id.number.str, NULL))) {
/*
* The new channel has no preset CallerID number by the channel
* driver. Use the dialplan extension and hint name.
*/
caller.id = stored_clid;
if (!caller.id.name.valid
&& !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
ast_channel_connected(chan)->id.name.str, NULL))) {
/*
* No hint name available. We have a connected name supplied by
* the dialplan we can use instead.
*/
caller.id.name.valid = 1;
caller.id.name = ast_channel_connected(chan)->id.name;
}
ast_channel_set_caller_event(tc, &caller, NULL);
ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
} else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
NULL))) {
/* The new channel has no preset CallerID name by the channel driver. */
if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
ast_channel_connected(chan)->id.name.str, NULL))) {
/*
* We have a connected name supplied by the dialplan we can
* use instead.
*/
caller.id.name.valid = 1;
caller.id.name = ast_channel_connected(chan)->id.name;
ast_channel_set_caller_event(tc, &caller, NULL);
}
}
/* Determine CallerID for outgoing channel to send. */
if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
struct ast_party_connected_line connected;
ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
connected.id = forced_clid;
ast_channel_set_connected_line(tc, &connected, NULL);
} else {
ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
}
ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
if (!ast_strlen_zero(ast_channel_accountcode(chan))) {
ast_channel_accountcode_set(tc, ast_channel_accountcode(chan));
}
if (ast_strlen_zero(ast_channel_musicclass(tc))) {
ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
}
/* Pass ADSI CPE and transfer capability */
ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
/* If we have an outbound group, set this peer channel to it */
if (outbound_group)
ast_app_group_set_channel(tc, outbound_group);
/* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
/* Check if we're forced by configuration */
if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
/* Inherit context and extension */
ast_channel_dialcontext_set(tc, ast_strlen_zero(ast_channel_macrocontext(chan)) ? ast_channel_context(chan) : ast_channel_macrocontext(chan));
if (!ast_strlen_zero(ast_channel_macroexten(chan)))
ast_channel_exten_set(tc, ast_channel_macroexten(chan));
else
ast_channel_exten_set(tc, ast_channel_exten(chan));
ast_channel_unlock(tc);
ast_channel_unlock(chan);
/* Put channel in the list of outgoing thingies. */
tmp->chan = tc;
AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
}
/*
* PREDIAL: Run gosub on all of the callee channels
*
* We run the callee predial before ast_call() in case the user
* wishes to do something on the newly created channels before
* the channel does anything important.
*
* Inside the target gosub we will be able to do something with
* the newly created channel name ie: now the calling channel
* can know what channel will be used to call the destination
* ex: now we will know that SIP/abc-123 is calling SIP/def-124
*/
if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
&& !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
&& !AST_LIST_EMPTY(&out_chans)) {
const char *predial_callee;
ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
if (predial_callee) {
ast_autoservice_start(chan);
AST_LIST_TRAVERSE(&out_chans, tmp, node) {
ast_pre_call(tmp->chan, predial_callee);
}
ast_autoservice_stop(chan);
ast_free((char *) predial_callee);
}
}
/* Start all outgoing calls */
AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
ast_channel_lock(chan);
/* Save the info in cdr's that we called them */
if (ast_channel_cdr(chan))
ast_cdr_setdestchan(ast_channel_cdr(chan), ast_channel_name(tmp->chan));
/* check the results of ast_call */
if (res) {
/* Again, keep going even if there's an error */
ast_debug(1, "ast call on peer returned %d\n", res);
ast_verb(3, "Couldn't call %s\n", tmp->interface);
if (ast_channel_hangupcause(tmp->chan)) {
ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
}
ast_channel_unlock(chan);
ast_cc_call_failed(chan, tmp->chan, tmp->interface);
ast_hangup(tmp->chan);
tmp->chan = NULL;
AST_LIST_REMOVE_CURRENT(node);
chanlist_free(tmp);
continue;
}
senddialevent(chan, tmp->chan, tmp->number);
ast_channel_unlock(chan);
ast_verb(3, "Called %s\n", tmp->interface);
ast_set_flag64(tmp, DIAL_STILLGOING);
/* If this line is up, don't try anybody else */
if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
break;
}
}
AST_LIST_TRAVERSE_SAFE_END;
if (ast_strlen_zero(args.timeout)) {
to = -1;
} else {
to = atoi(args.timeout);
if (to > 0)
to *= 1000;
else {
ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
to = -1;
}
}
outgoing = AST_LIST_FIRST(&out_chans);
if (!outgoing) {
strcpy(pa.status, "CHANUNAVAIL");
if (fulldial == num_dialed) {
res = -1;
goto out;
}
} else {
/* Our status will at least be NOANSWER */
strcpy(pa.status, "NOANSWER");
if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
moh = 1;
if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
ast_channel_musicclass_set(chan, original_moh);
} else {
ast_moh_start(chan, NULL, NULL);
}
ast_indicate(chan, AST_CONTROL_PROGRESS);
} else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
} else {
ast_indicate(chan, AST_CONTROL_PROGRESS);
}
} else {
ast_indicate(chan, AST_CONTROL_RINGING);
sentringing++;
}
}
}
peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
dtmf_progress, ignore_cc, &forced_clid, &stored_clid);
/* The ast_channel_datastore_remove() function could fail here if the
* datastore was moved to another channel during a masquerade. If this is
* the case, don't free the datastore here because later, when the channel
* to which the datastore was moved hangs up, it will attempt to free this
* datastore again, causing a crash
*/
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL); /* make sure we weren't cleaned up already */
if (datastore && !ast_channel_datastore_remove(chan, datastore)) {
ast_datastore_free(datastore);
}
ast_channel_unlock(chan);
if (!peer) {
if (result) {
res = result;
} else if (to) { /* Musta gotten hung up */
res = -1;
} else { /* Nobody answered, next please? */
res = 0;
}
} else {
const char *number;
if (ast_test_flag64(&opts, OPT_CALLER_ANSWER))
ast_answer(chan);
strcpy(pa.status, "ANSWER");
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
/* Ah ha! Someone answered within the desired timeframe. Of course after this
we will always return with -1 so that it is hung up properly after the
conversation. */
hanguptree(&out_chans, peer, 1);
/* If appropriate, log that we have a destination channel and set the answer time */
if (ast_channel_cdr(chan)) {
ast_cdr_setdestchan(ast_channel_cdr(chan), ast_channel_name(peer));
ast_cdr_setanswer(ast_channel_cdr(chan), ast_channel_cdr(peer)->answer);
}
if (ast_channel_name(peer))
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", ast_channel_name(peer));
ast_channel_lock(peer);
number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
if (ast_strlen_zero(number)) {
number = NULL;
} else {
number = ast_strdupa(number);
}
ast_channel_unlock(peer);
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
ast_channel_sendurl( peer, args.url );
}
if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
res = 0;
goto out;
}
}
if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
res = 0;
} else {
int digit = 0;
struct ast_channel *chans[2];
struct ast_channel *active_chan;
chans[0] = chan;
chans[1] = peer;
/* we need to stream the announcment while monitoring the caller for a hangup */
/* stream the file */
res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
if (res) {
res = 0;
ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
}
ast_set_flag(ast_channel_flags(peer), AST_FLAG_END_DTMF_ONLY);
while (ast_channel_stream(peer)) {
int ms;
ms = ast_sched_wait(ast_channel_sched(peer));
if (ms < 0 && !ast_channel_timingfunc(peer)) {
ast_stopstream(peer);
break;
}
if (ms < 0)
ms = 1000;
active_chan = ast_waitfor_n(chans, 2, &ms);
if (active_chan) {
struct ast_frame *fr = ast_read(active_chan);
if (!fr) {
ast_autoservice_chan_hangup_peer(chan, peer);
res = -1;
goto done;
}
switch(fr->frametype) {
case AST_FRAME_DTMF_END:
digit = fr->subclass.integer;
if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
ast_stopstream(peer);
res = ast_senddigit(chan, digit, 0);
}
break;
case AST_FRAME_CONTROL:
switch (fr->subclass.integer) {
case AST_CONTROL_HANGUP:
ast_frfree(fr);
ast_autoservice_chan_hangup_peer(chan, peer);
res = -1;
goto done;
default:
break;
}
break;
default:
/* Ignore all others */
break;
}
ast_frfree(fr);
}
ast_sched_runq(ast_channel_sched(peer));
}
ast_clear_flag(ast_channel_flags(peer), AST_FLAG_END_DTMF_ONLY);
}
if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
/* chan and peer are going into the PBX, they both
* should probably get CDR records. */
ast_clear_flag(ast_channel_cdr(chan), AST_CDR_FLAG_DIALED);
ast_clear_flag(ast_channel_cdr(peer), AST_CDR_FLAG_DIALED);
ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
/* peer goes to the same context and extension as chan, so just copy info from chan*/
ast_channel_context_set(peer, ast_channel_context(chan));
ast_channel_exten_set(peer, ast_channel_exten(chan));
ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
if (ast_pbx_start(peer)) {
ast_autoservice_chan_hangup_peer(chan, peer);
}
hanguptree(&out_chans, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
if (continue_exec)
*continue_exec = 1;
res = 0;
goto done;
}
if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
const char *macro_result_peer;
/* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
ast_channel_lock_both(chan, peer);
ast_channel_context_set(peer, ast_channel_context(chan));
ast_channel_exten_set(peer, ast_channel_exten(chan));
ast_channel_unlock(peer);
ast_channel_unlock(chan);
ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
res = ast_app_exec_macro(chan, peer, opt_args[OPT_ARG_CALLEE_MACRO]);
ast_channel_lock(peer);
if (!res && (macro_result_peer = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
char *macro_result = ast_strdupa(macro_result_peer);
char *macro_transfer_dest;
ast_channel_unlock(peer);
if (!strcasecmp(macro_result, "BUSY")) {
ast_copy_string(pa.status, macro_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
ast_copy_string(pa.status, macro_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(macro_result, "CONTINUE")) {
/* hangup peer and keep chan alive assuming the macro has changed
the context / exten / priority or perhaps
the next priority in the current exten is desired.
*/
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(macro_result, "ABORT")) {
/* Hangup both ends unless the caller has the g flag */
res = -1;
} else if (!strncasecmp(macro_result, "GOTO:", 5)) {
macro_transfer_dest = macro_result + 5;
res = -1;
/* perform a transfer to a new extension */
if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
ast_replace_subargument_delimiter(macro_transfer_dest);
}
if (!ast_parseable_goto(chan, macro_transfer_dest)) {
ast_set_flag64(peerflags, OPT_GO_ON);
}
}
} else {
ast_channel_unlock(peer);
}
}
if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
const char *gosub_result_peer;
char *gosub_argstart;
char *gosub_args = NULL;
int res9 = -1;
ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
if (gosub_argstart) {
const char *what_is_s = "s";
*gosub_argstart = 0;
if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
what_is_s = "~~s~~";
}
if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
gosub_args = NULL;
}
*gosub_argstart = ',';
} else {
const char *what_is_s = "s";
if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
what_is_s = "~~s~~";
}
if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
gosub_args = NULL;
}
}
if (gosub_args) {
res9 = ast_app_exec_sub(chan, peer, gosub_args, 0);
ast_free(gosub_args);
} else {
ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
}
ast_channel_lock_both(chan, peer);
if (!res9 && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
char *gosub_transfer_dest;
char *gosub_result = ast_strdupa(gosub_result_peer);
const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
/* Inherit return value from the peer, so it can be used in the master */
if (gosub_retval) {
pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
}
ast_channel_unlock(peer);
ast_channel_unlock(chan);
if (!strcasecmp(gosub_result, "BUSY")) {
ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(gosub_result, "CONTINUE")) {
/* Hangup peer and continue with the next extension priority. */
ast_set_flag64(peerflags, OPT_GO_ON);
res = -1;
} else if (!strcasecmp(gosub_result, "ABORT")) {
/* Hangup both ends unless the caller has the g flag */
res = -1;
} else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
gosub_transfer_dest = gosub_result + 5;
res = -1;
/* perform a transfer to a new extension */
if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
ast_replace_subargument_delimiter(gosub_transfer_dest);
}
if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
ast_set_flag64(peerflags, OPT_GO_ON);
}
}
} else {
ast_channel_unlock(peer);
ast_channel_unlock(chan);
}
}
if (!res) {
if (!ast_tvzero(calldurationlimit)) {
struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
ast_channel_whentohangup_set(peer, &whentohangup);
}
if (!ast_strlen_zero(dtmfcalled)) {
ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
}
if (!ast_strlen_zero(dtmfcalling)) {
ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
}
}
if (res) { /* some error */
res = -1;
} else {
if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
if (ast_test_flag64(peerflags, OPT_GO_ON))
ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
config.end_bridge_callback = end_bridge_callback;
config.end_bridge_callback_data = chan;
config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
/* Be sure no generators are left on it and reset the visible indication */
ast_deactivate_generator(chan);
ast_channel_visible_indication_set(chan, 0);
/* Make sure channels are compatible */
res = ast_channel_make_compatible(chan, peer);
if (res < 0) {
ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", ast_channel_name(chan), ast_channel_name(peer));
ast_autoservice_chan_hangup_peer(chan, peer);
res = -1;
goto done;
}
if (opermode) {
struct oprmode oprmode;
oprmode.peer = peer;
oprmode.mode = opermode;
ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
}
res = ast_bridge_call(chan, peer, &config);
}
ast_channel_context_set(peer, ast_channel_context(chan));
if (ast_test_flag64(&opts, OPT_PEER_H)
&& ast_exists_extension(peer, ast_channel_context(peer), "h", 1,
S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
ast_autoservice_start(chan);
ast_pbx_h_exten_run(peer, ast_channel_context(peer));
ast_autoservice_stop(chan);
}
if (!ast_check_hangup(peer)) {
if (ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
int goto_res;
if (!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
goto_res = ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
} else { /* F() */
goto_res = ast_goto_if_exists(peer, ast_channel_context(chan),
ast_channel_exten(chan), ast_channel_priority(chan) + 1);
}
if (!goto_res && !ast_pbx_start(peer)) {
/* The peer is now running its own PBX. */
goto out;
}
}
} else if (!ast_check_hangup(chan)) {
ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
}
ast_autoservice_chan_hangup_peer(chan, peer);
}
out:
if (moh) {
moh = 0;
ast_moh_stop(chan);
} else if (sentringing) {
sentringing = 0;
ast_indicate(chan, -1);
}
if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
ast_filedelete(pa.privintro, NULL);
if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
} else {
ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
}
}
ast_channel_early_bridge(chan, NULL);
hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
senddialendevent(chan, pa.status);
ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
if (!ast_tvzero(calldurationlimit))
memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
res = 0;
}
done:
if (config.warning_sound) {
ast_free((char *)config.warning_sound);
}
if (config.end_sound) {
ast_free((char *)config.end_sound);
}
if (config.start_sound) {
ast_free((char *)config.start_sound);
}
ast_ignore_cc(chan);
return res;
}
| static int dial_handle_playtones | ( | struct ast_channel * | chan, |
| const char * | data | ||
| ) | [static] |
Definition at line 2025 of file app_dial.c.
References ast_channel_zone(), ast_debug, ast_get_indication_tone(), ast_log(), ast_playtones_start(), ast_strlen_zero(), ast_tone_zone_sound_unref(), ast_tone_zone_sound::data, LOG_WARNING, and str.
Referenced by dial_exec_full().
{
struct ast_tone_zone_sound *ts = NULL;
int res;
const char *str = data;
if (ast_strlen_zero(str)) {
ast_debug(1,"Nothing to play\n");
return -1;
}
ts = ast_get_indication_tone(ast_channel_zone(chan), str);
if (ts && ts->data[0]) {
res = ast_playtones_start(chan, 0, ts->data, 0);
} else {
res = -1;
}
if (ts) {
ts = ast_tone_zone_sound_unref(ts);
}
if (res) {
ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
}
return res;
}
| static void do_forward | ( | struct chanlist * | o, |
| struct cause_args * | num, | ||
| struct ast_flags64 * | peerflags, | ||
| int | single, | ||
| int | caller_entertained, | ||
| int * | to, | ||
| struct ast_party_id * | forced_clid, | ||
| struct ast_party_id * | stored_clid | ||
| ) | [static] |
helper function for wait_for_answer()
| o | Outgoing call channel list. |
| num | Incoming call channel cause accumulation |
| peerflags | Dial option flags |
| single | TRUE if there is only one outgoing call. |
| caller_entertained | TRUE if the caller is being entertained by MOH or ringback. |
| to | Remaining call timeout time. |
| forced_clid | OPT_FORCECLID caller id to send |
| stored_clid | Caller id representing the called party if needed |
XXX this code is highly suspicious, as it essentially overwrites the outgoing channel without properly deleting it.
Definition at line 894 of file app_dial.c.
References ast_call(), AST_CAUSE_BUSY, AST_CEL_FORWARD, ast_cel_report_event(), ast_channel_accountcode(), ast_channel_appl_set(), ast_channel_call_forward(), ast_channel_caller(), ast_channel_connected(), ast_channel_context(), ast_channel_data_set(), ast_channel_datastore_inherit(), ast_channel_dialed(), ast_channel_exten(), ast_channel_inherit_variables(), ast_channel_lock, ast_channel_lock_both, ast_channel_macroexten(), ast_channel_make_compatible(), ast_channel_name(), ast_channel_nativeformats(), ast_channel_redirecting(), ast_channel_redirecting_macro(), ast_channel_redirecting_sub(), ast_channel_set_caller_event(), ast_channel_unlock, ast_channel_update_redirecting(), ast_clear_flag64, ast_connected_line_copy_from_caller(), ast_copy_string(), ast_hangup(), ast_ignore_cc(), ast_indicate(), ast_log(), ast_party_caller_set_init(), ast_party_connected_line_copy(), ast_party_connected_line_init(), ast_party_number_free(), ast_party_number_init(), ast_party_redirecting_copy(), ast_party_redirecting_free(), ast_party_redirecting_init(), ast_request(), ast_rtp_instance_early_bridge_make_compatible(), ast_set_flag64, ast_strdup, ast_strlen_zero(), ast_test_flag64, ast_verb, CAN_EARLY_BRIDGE, chanlist::chan, cause_args::chan, DIAL_CALLERID_ABSENT, DIAL_STILLGOING, ast_party_redirecting::from, handle_cause(), ast_party_caller::id, ast_party_connected_line::id, LOG_NOTICE, cause_args::nochan, ast_party_id::number, OPT_CANCEL_TIMEOUT, OPT_FORCECLID, OPT_IGNORE_CONNECTEDLINE, OPT_IGNORE_FORWARDING, OPT_ORIGINAL_CLID, pbx_builtin_getvar_helper(), S_COR, S_OR, senddialevent(), ast_party_number::str, chanlist::stuff, chanlist::tech, ast_party_dialed::transit_network_select, and ast_party_number::valid.
Referenced by wait_for_answer().
{
char tmpchan[256];
struct ast_channel *original = o->chan;
struct ast_channel *c = o->chan; /* the winner */
struct ast_channel *in = num->chan; /* the input channel */
char *stuff;
char *tech;
int cause;
struct ast_party_caller caller;
ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
if ((stuff = strchr(tmpchan, '/'))) {
*stuff++ = '\0';
tech = tmpchan;
} else {
const char *forward_context;
ast_channel_lock(c);
forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
if (ast_strlen_zero(forward_context)) {
forward_context = NULL;
}
snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
ast_channel_unlock(c);
stuff = tmpchan;
tech = "Local";
}
if (!strcasecmp(tech, "Local")) {
/*
* Drop the connected line update block for local channels since
* this is going to run dialplan and the user can change his
* mind about what connected line information he wants to send.
*/
ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
}
ast_cel_report_event(in, AST_CEL_FORWARD, NULL, ast_channel_call_forward(c), NULL);
/* Before processing channel, go ahead and check for forwarding */
ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
/* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
c = o->chan = NULL;
cause = AST_CAUSE_BUSY;
} else {
/* Setup parameters */
c = o->chan = ast_request(tech, ast_channel_nativeformats(in), in, stuff, &cause);
if (c) {
if (single && !caller_entertained) {
ast_channel_make_compatible(o->chan, in);
}
ast_channel_lock_both(in, o->chan);
ast_channel_inherit_variables(in, o->chan);
ast_channel_datastore_inherit(in, o->chan);
ast_channel_unlock(in);
ast_channel_unlock(o->chan);
/* When a call is forwarded, we don't want to track new interfaces
* dialed for CC purposes. Setting the done flag will ensure that
* any Dial operations that happen later won't record CC interfaces.
*/
ast_ignore_cc(o->chan);
ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
} else
ast_log(LOG_NOTICE,
"Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
tech, stuff, cause);
}
if (!c) {
ast_clear_flag64(o, DIAL_STILLGOING);
handle_cause(cause, num);
ast_hangup(original);
} else {
ast_channel_lock_both(c, original);
ast_party_redirecting_copy(ast_channel_redirecting(c),
ast_channel_redirecting(original));
ast_channel_unlock(c);
ast_channel_unlock(original);
ast_channel_lock_both(c, in);
if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
ast_rtp_instance_early_bridge_make_compatible(c, in);
}
if (!ast_channel_redirecting(c)->from.number.valid
|| ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
/*
* The call was not previously redirected so it is
* now redirected from this number.
*/
ast_party_number_free(&ast_channel_redirecting(c)->from.number);
ast_party_number_init(&ast_channel_redirecting(c)->from.number);
ast_channel_redirecting(c)->from.number.valid = 1;
ast_channel_redirecting(c)->from.number.str =
ast_strdup(S_OR(ast_channel_macroexten(in), ast_channel_exten(in)));
}
ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
/* Determine CallerID to store in outgoing channel. */
ast_party_caller_set_init(&caller, ast_channel_caller(c));
if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
caller.id = *stored_clid;
ast_channel_set_caller_event(c, &caller, NULL);
ast_set_flag64(o, DIAL_CALLERID_ABSENT);
} else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
ast_channel_caller(c)->id.number.str, NULL))) {
/*
* The new channel has no preset CallerID number by the channel
* driver. Use the dialplan extension and hint name.
*/
caller.id = *stored_clid;
ast_channel_set_caller_event(c, &caller, NULL);
ast_set_flag64(o, DIAL_CALLERID_ABSENT);
} else {
ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
}
/* Determine CallerID for outgoing channel to send. */
if (ast_test_flag64(o, OPT_FORCECLID)) {
struct ast_party_connected_line connected;
ast_party_connected_line_init(&connected);
connected.id = *forced_clid;
ast_party_connected_line_copy(ast_channel_connected(c), &connected);
} else {
ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
}
ast_channel_accountcode_set(c, ast_channel_accountcode(in));
ast_channel_appl_set(c, "AppDial");
ast_channel_data_set(c, "(Outgoing Line)");
ast_channel_unlock(in);
if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
struct ast_party_redirecting redirecting;
/*
* Redirecting updates to the caller make sense only on single
* calls.
*
* We must unlock c before calling
* ast_channel_redirecting_macro, because we put c into
* autoservice there. That is pretty much a guaranteed
* deadlock. This is why the handling of c's lock may seem a
* bit unusual here.
*/
ast_party_redirecting_init(&redirecting);
ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
ast_channel_unlock(c);
if (ast_channel_redirecting_sub(c, in, &redirecting, 0) &&
ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
ast_channel_update_redirecting(in, &redirecting, NULL);
}
ast_party_redirecting_free(&redirecting);
} else {
ast_channel_unlock(c);
}
if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
*to = -1;
}
if (ast_call(c, stuff, 0)) {
ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
tech, stuff);
ast_clear_flag64(o, DIAL_STILLGOING);
ast_hangup(original);
ast_hangup(c);
c = o->chan = NULL;
num->nochan++;
} else {
ast_channel_lock_both(c, in);
senddialevent(in, c, stuff);
ast_channel_unlock(in);
ast_channel_unlock(c);
/* Hangup the original channel now, in case we needed it */
ast_hangup(original);
}
if (single && !caller_entertained) {
ast_indicate(in, -1);
}
}
}
| static void end_bridge_callback | ( | void * | data | ) | [static] |
Definition at line 1996 of file app_dial.c.
References ast_channel_cdr(), ast_channel_lock, ast_channel_unlock, ast_channel::data, and pbx_builtin_setvar_helper().
Referenced by dial_exec_full().
{
char buf[80];
time_t end;
struct ast_channel *chan = data;
if (!ast_channel_cdr(chan)) {
return;
}
time(&end);
ast_channel_lock(chan);
if (ast_channel_cdr(chan)->answer.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", (long) end - ast_channel_cdr(chan)->answer.tv_sec);
pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
}
if (ast_channel_cdr(chan)->start.tv_sec) {
snprintf(buf, sizeof(buf), "%ld", (long) end - ast_channel_cdr(chan)->start.tv_sec);
pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
}
ast_channel_unlock(chan);
}
| static void end_bridge_callback_data_fixup | ( | struct ast_bridge_config * | bconfig, |
| struct ast_channel * | originator, | ||
| struct ast_channel * | terminator | ||
| ) | [static] |
Definition at line 2021 of file app_dial.c.
References ast_bridge_config::end_bridge_callback_data.
Referenced by dial_exec_full().
{
bconfig->end_bridge_callback_data = originator;
}
| static const char* get_cid_name | ( | char * | name, |
| int | namelen, | ||
| struct ast_channel * | chan | ||
| ) | [static] |
Definition at line 807 of file app_dial.c.
References ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_macrocontext(), ast_channel_macroexten(), ast_channel_unlock, ast_get_hint(), context, exten, and S_OR.
Referenced by dial_exec_full().
{
const char *context;
const char *exten;
ast_channel_lock(chan);
context = ast_strdupa(S_OR(ast_channel_macrocontext(chan), ast_channel_context(chan)));
exten = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
ast_channel_unlock(chan);
return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
}
| static void handle_cause | ( | int | cause, |
| struct cause_args * | num | ||
| ) | [static] |
Definition at line 750 of file app_dial.c.
References AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NORMAL_CLEARING, AST_CAUSE_UNREGISTERED, ast_cdr_busy(), ast_cdr_failed(), ast_cdr_noanswer(), ast_channel_cdr(), cause_args::busy, cause_args::chan, cause_args::congestion, and cause_args::nochan.
Referenced by dial_exec_full(), do_forward(), and wait_for_answer().
{
struct ast_cdr *cdr = ast_channel_cdr(num->chan);
switch(cause) {
case AST_CAUSE_BUSY:
if (cdr)
ast_cdr_busy(cdr);
num->busy++;
break;
case AST_CAUSE_CONGESTION:
if (cdr)
ast_cdr_failed(cdr);
num->congestion++;
break;
case AST_CAUSE_NO_ROUTE_DESTINATION:
case AST_CAUSE_UNREGISTERED:
if (cdr)
ast_cdr_failed(cdr);
num->nochan++;
break;
case AST_CAUSE_NO_ANSWER:
if (cdr) {
ast_cdr_noanswer(cdr);
}
break;
case AST_CAUSE_NORMAL_CLEARING:
break;
default:
num->nochan++;
break;
}
}
| static void hanguptree | ( | struct dial_head * | out_chans, |
| struct ast_channel * | exception, | ||
| int | answered_elsewhere | ||
| ) | [static] |
Definition at line 720 of file app_dial.c.
References AST_CAUSE_ANSWERED_ELSEWHERE, ast_channel_hangupcause_set(), ast_hangup(), AST_LIST_REMOVE_HEAD, chanlist::chan, chanlist_free(), and chanlist::node.
Referenced by dial_exec_full().
{
/* Hang up a tree of stuff */
struct chanlist *outgoing;
while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
/* Hangup any existing lines we have open */
if (outgoing->chan && (outgoing->chan != exception)) {
if (answered_elsewhere) {
/* This is for the channel drivers */
ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
}
ast_hangup(outgoing->chan);
}
chanlist_free(outgoing);
}
}
| static int load_module | ( | void | ) | [static] |
Definition at line 3251 of file app_dial.c.
References ast_register_application_xml, dial_exec(), and retrydial_exec().
{
int res;
res = ast_register_application_xml(app, dial_exec);
res |= ast_register_application_xml(rapp, retrydial_exec);
return res;
}
| static int onedigit_goto | ( | struct ast_channel * | chan, |
| const char * | context, | ||
| char | exten, | ||
| int | pri | ||
| ) | [static] |
Definition at line 788 of file app_dial.c.
References ast_channel_context(), ast_channel_macrocontext(), ast_goto_if_exists(), ast_strlen_zero(), and exten.
Referenced by retrydial_exec(), and wait_for_answer().
{
char rexten[2] = { exten, '\0' };
if (context) {
if (!ast_goto_if_exists(chan, context, rexten, pri))
return 1;
} else {
if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
return 1;
else if (!ast_strlen_zero(ast_channel_macrocontext(chan))) {
if (!ast_goto_if_exists(chan, ast_channel_macrocontext(chan), rexten, pri))
return 1;
}
}
return 0;
}
| static int retrydial_exec | ( | struct ast_channel * | chan, |
| const char * | data | ||
| ) | [static] |
Definition at line 3133 of file app_dial.c.
References args, AST_APP_ARG, ast_channel_data_set(), ast_channel_flags(), ast_channel_language(), ast_channel_lock, ast_channel_unlock, AST_DECLARE_APP_ARGS, AST_DIGIT_ANY, ast_fileexists(), AST_FLAG_MOH, ast_log(), ast_moh_start(), ast_moh_stop(), AST_PBX_INCOMPLETE, AST_STANDARD_APP_ARGS, ast_streamfile(), ast_strlen_zero(), ast_test_flag, ast_test_flag64, ast_waitfordigit(), ast_waitstream(), context, dial_exec_full(), LOG_ERROR, LOG_WARNING, onedigit_goto(), OPT_DTMF_EXIT, parse(), and pbx_builtin_getvar_helper().
Referenced by load_module().
{
char *parse;
const char *context = NULL;
int sleepms = 0, loops = 0, res = -1;
struct ast_flags64 peerflags = { 0, };
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(announce);
AST_APP_ARG(sleep);
AST_APP_ARG(retries);
AST_APP_ARG(dialdata);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
sleepms *= 1000;
if (!ast_strlen_zero(args.retries)) {
loops = atoi(args.retries);
}
if (!args.dialdata) {
ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
goto done;
}
if (sleepms < 1000)
sleepms = 10000;
if (!loops)
loops = -1; /* run forever */
ast_channel_lock(chan);
context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
ast_channel_unlock(chan);
res = 0;
while (loops) {
int continue_exec;
ast_channel_data_set(chan, "Retrying");
if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
ast_moh_stop(chan);
res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
if (continue_exec)
break;
if (res == 0) {
if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
if (!ast_strlen_zero(args.announce)) {
if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
ast_waitstream(chan, AST_DIGIT_ANY);
} else
ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
}
if (!res && sleepms) {
if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
ast_moh_start(chan, NULL, NULL);
res = ast_waitfordigit(chan, sleepms);
}
} else {
if (!ast_strlen_zero(args.announce)) {
if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
res = ast_waitstream(chan, "");
} else
ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
}
if (sleepms) {
if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
ast_moh_start(chan, NULL, NULL);
if (!res)
res = ast_waitfordigit(chan, sleepms);
}
}
}
if (res < 0 || res == AST_PBX_INCOMPLETE) {
break;
} else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
if (onedigit_goto(chan, context, (char) res, 1)) {
res = 0;
break;
}
}
loops--;
}
if (loops == 0)
res = 0;
else if (res == 1)
res = 0;
if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
ast_moh_stop(chan);
done:
return res;
}
| static void senddialendevent | ( | struct ast_channel * | src, |
| const char * | dialstatus | ||
| ) | [static] |
Definition at line 857 of file app_dial.c.
References ast_channel_name(), ast_channel_uniqueid(), ast_manager_event, and EVENT_FLAG_CALL.
Referenced by dial_exec_full().
{
/*** DOCUMENTATION
<managerEventInstance>
<synopsis>Raised when a dial action has ended.</synopsis>
<syntax>
<parameter name="DialStatus">
<para>The value of the <variable>DIALSTATUS</variable> channel variable.</para>
</parameter>
</syntax>
</managerEventInstance>
***/
ast_manager_event(src, EVENT_FLAG_CALL, "Dial",
"SubEvent: End\r\n"
"Channel: %s\r\n"
"UniqueID: %s\r\n"
"DialStatus: %s\r\n",
ast_channel_name(src), ast_channel_uniqueid(src), dialstatus);
}
| static void senddialevent | ( | struct ast_channel * | src, |
| struct ast_channel * | dst, | ||
| const char * | dialstring | ||
| ) | [static] |
Definition at line 820 of file app_dial.c.
References ast_channel_caller(), ast_channel_connected(), ast_channel_name(), ast_channel_uniqueid(), ast_manager_event_multichan, EVENT_FLAG_CALL, name, and S_COR.
Referenced by dial_exec_full(), and do_forward().
{
struct ast_channel *chans[] = { src, dst };
/*** DOCUMENTATION
<managerEventInstance>
<synopsis>Raised when a dial action has started.</synopsis>
<syntax>
<parameter name="SubEvent">
<para>A sub event type, specifying whether the dial action has begun or ended.</para>
<enumlist>
<enum name="Begin"/>
<enum name="End"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
***/
ast_manager_event_multichan(EVENT_FLAG_CALL, "Dial", 2, chans,
"SubEvent: Begin\r\n"
"Channel: %s\r\n"
"Destination: %s\r\n"
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n"
"ConnectedLineNum: %s\r\n"
"ConnectedLineName: %s\r\n"
"UniqueID: %s\r\n"
"DestUniqueID: %s\r\n"
"Dialstring: %s\r\n",
ast_channel_name(src), ast_channel_name(dst),
S_COR(ast_channel_caller(src)->id.number.valid, ast_channel_caller(src)->id.number.str, "<unknown>"),
S_COR(ast_channel_caller(src)->id.name.valid, ast_channel_caller(src)->id.name.str, "<unknown>"),
S_COR(ast_channel_connected(src)->id.number.valid, ast_channel_connected(src)->id.number.str, "<unknown>"),
S_COR(ast_channel_connected(src)->id.name.valid, ast_channel_connected(src)->id.name.str, "<unknown>"),
ast_channel_uniqueid(src), ast_channel_uniqueid(dst),
dialstring ? dialstring : "");
}
| static int setup_privacy_args | ( | struct privacy_args * | pa, |
| struct ast_flags64 * | opts, | ||
| char * | opt_args[], | ||
| struct ast_channel * | chan | ||
| ) | [static] |
returns 1 if successful, 0 or <0 if the caller should 'goto out'
Definition at line 1895 of file app_dial.c.
References ast_answer(), ast_channel_caller(), ast_channel_exten(), ast_channel_language(), ast_channel_name(), ast_config_AST_DATA_DIR, ast_copy_string(), ast_dsp_get_threshold_from_settings(), ast_filedelete(), ast_fileexists(), ast_log(), ast_mkdir(), ast_play_and_record(), AST_PRIVACY_ALLOW, ast_privacy_check(), AST_PRIVACY_DENY, AST_PRIVACY_KILL, AST_PRIVACY_TORTURE, AST_PRIVACY_UNKNOWN, ast_shrink_phone_number(), ast_streamfile(), ast_strlen_zero(), ast_test_flag64, ast_verb, ast_waitstream(), LOG_NOTICE, LOG_WARNING, OPT_ARG_PRIVACY, OPT_PRIVACY, OPT_SCREEN_NOCALLERID, privacy_args::privcid, privacy_args::privdb_val, privacy_args::privintro, silencethreshold, privacy_args::status, and THRESHOLD_SILENCE.
Referenced by dial_exec_full().
{
char callerid[60];
int res;
char *l;
if (ast_channel_caller(chan)->id.number.valid
&& !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
ast_shrink_phone_number(l);
if (ast_test_flag64(opts, OPT_PRIVACY) ) {
ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
} else {
ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
pa->privdb_val = AST_PRIVACY_UNKNOWN;
}
} else {
char *tnam, *tn2;
tnam = ast_strdupa(ast_channel_name(chan));
/* clean the channel name so slashes don't try to end up in disk file name */
for (tn2 = tnam; *tn2; tn2++) {
if (*tn2 == '/') /* any other chars to be afraid of? */
*tn2 = '=';
}
ast_verb(3, "Privacy-- callerid is empty\n");
snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
l = callerid;
pa->privdb_val = AST_PRIVACY_UNKNOWN;
}
ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
/* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
pa->privdb_val = AST_PRIVACY_ALLOW;
} else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
}
if (pa->privdb_val == AST_PRIVACY_DENY) {
ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
return 0;
} else if (pa->privdb_val == AST_PRIVACY_KILL) {
ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
return 0; /* Is this right? */
} else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
return 0; /* is this right??? */
} else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
/* Get the user's intro, store it in priv-callerintros/$CID,
unless it is already there-- this should be done before the
call is actually dialed */
/* make sure the priv-callerintros dir actually exists */
snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
if ((res = ast_mkdir(pa->privintro, 0755))) {
ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
return -1;
}
snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
/* the DELUX version of this code would allow this caller the
option to hear and retape their previously recorded intro.
*/
} else {
int duration; /* for feedback from play_and_wait */
/* the file doesn't exist yet. Let the caller submit his
vocal intro for posterity */
/* priv-recordintro script:
"At the tone, please say your name:"
*/
int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
ast_answer(chan);
res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, NULL, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
/* don't think we'll need a lock removed, we took care of
conflicts by naming the pa.privintro file */
if (res == -1) {
/* Delete the file regardless since they hung up during recording */
ast_filedelete(pa->privintro, NULL);
if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
else
ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
return -1;
}
if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
ast_waitstream(chan, "");
}
}
return 1; /* success */
}
| static int unload_module | ( | void | ) | [static] |
Definition at line 3241 of file app_dial.c.
References ast_unregister_application().
{
int res;
res = ast_unregister_application(app);
res |= ast_unregister_application(rapp);
return res;
}
| static int valid_priv_reply | ( | struct ast_flags64 * | opts, |
| int | res | ||
| ) | [static] |
Definition at line 1748 of file app_dial.c.
References ast_test_flag64, OPT_PRIVACY, and OPT_SCREENING.
{
if (res < '1')
return 0;
if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
return 1;
if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
return 1;
return 0;
}
| static struct ast_channel* wait_for_answer | ( | struct ast_channel * | in, |
| struct dial_head * | out_chans, | ||
| int * | to, | ||
| struct ast_flags64 * | peerflags, | ||
| char * | opt_args[], | ||
| struct privacy_args * | pa, | ||
| const struct cause_args * | num_in, | ||
| int * | result, | ||
| char * | dtmf_progress, | ||
| const int | ignore_cc, | ||
| struct ast_party_id * | forced_clid, | ||
| struct ast_party_id * | stored_clid | ||
| ) | [static, read] |
Definition at line 1092 of file app_dial.c.
References ast_cdr::answer, chanlist::aoc_s_rate_list, ast_aoc_decode(), ast_aoc_destroy_decoded(), ast_aoc_destroy_encoded(), ast_aoc_encode(), ast_aoc_get_msg_type(), AST_AOC_S, AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NORMAL_CLEARING, ast_cc_completed(), ast_cc_failed(), ast_cc_is_recall(), AST_CDR_ANSWERED, ast_cdr_failed(), ast_cdr_noanswer(), ast_channel_call_forward(), ast_channel_caller(), ast_channel_cdr(), ast_channel_connected(), ast_channel_connected_line_macro(), ast_channel_connected_line_sub(), ast_channel_early_bridge(), ast_channel_exten_set(), ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_lock, ast_channel_make_compatible(), ast_channel_name(), ast_channel_redirecting_macro(), ast_channel_redirecting_sub(), ast_channel_sendhtml(), ast_channel_unlock, ast_channel_update_connected_line(), ast_check_hangup(), ast_clear_flag64, ast_connected_line_copy_from_caller(), ast_connected_line_parse_data(), AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, AST_CONTROL_ANSWER, AST_CONTROL_AOC, AST_CONTROL_BUSY, AST_CONTROL_CC, AST_CONTROL_CONGESTION, AST_CONTROL_CONNECTED_LINE, AST_CONTROL_FLASH, AST_CONTROL_HANGUP, AST_CONTROL_HOLD, AST_CONTROL_OFFHOOK, 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_UNHOLD, AST_CONTROL_VIDUPDATE, ast_copy_flags64, ast_deactivate_generator(), ast_debug, ast_dtmf_stream(), AST_FRAME_CONTROL, AST_FRAME_DTMF, AST_FRAME_DTMF_BEGIN, AST_FRAME_DTMF_END, AST_FRAME_HTML, AST_FRAME_IMAGE, AST_FRAME_TEXT, AST_FRAME_VOICE, ast_frfree, ast_handle_cc_control_frame(), ast_hangup(), ast_indicate(), ast_indicate_data(), AST_LIST_FIRST, AST_LIST_NEXT, AST_LIST_TRAVERSE, ast_log(), AST_MAX_WATCHERS, ast_party_connected_line_copy(), ast_party_connected_line_free(), ast_party_connected_line_init(), ast_party_connected_line_set(), ast_party_connected_line_set_init(), ast_poll_channel_add(), ast_poll_channel_del(), ast_read(), ast_remaining_ms(), AST_STATE_UP, ast_str_alloca, ast_strlen_zero(), ast_test_flag64, ast_tvnow(), ast_verb, ast_waitfor_n(), ast_write(), cause_args::busy, CAN_EARLY_BRIDGE, chanlist::chan, cause_args::congestion, chanlist::connected, context, ast_frame::data, ast_frame::datalen, detect_disconnect(), DIAL_CALLERID_ABSENT, DIAL_NOFORWARDHTML, DIAL_STILLGOING, ast_cdr::disposition, do_forward(), f, FEATURE_MAX_LEN, ast_frame::frametype, handle_cause(), ast_frame_subclass::integer, LOG_WARNING, ast_channel::name, cause_args::nochan, chanlist::node, onedigit_goto(), OPT_ARG_RINGBACK, OPT_CALLEE_HANGUP, OPT_CALLEE_MIXMONITOR, OPT_CALLEE_MONITOR, OPT_CALLEE_PARK, OPT_CALLEE_TRANSFER, OPT_CALLER_HANGUP, OPT_CALLER_MIXMONITOR, OPT_CALLER_MONITOR, OPT_CALLER_PARK, OPT_CALLER_TRANSFER, OPT_DTMF_EXIT, OPT_IGNORE_CONNECTEDLINE, OPT_MUSICBACK, OPT_RINGBACK, pbx_builtin_getvar_helper(), chanlist::pending_connected_update, ast_frame::ptr, privacy_args::sentringing, ast_party_connected_line::source, privacy_args::status, ast_frame::subclass, and ast_frame::uint32.
Referenced by dial_exec_full().
{
struct cause_args num = *num_in;
int prestart = num.busy + num.congestion + num.nochan;
int orig = *to;
struct ast_channel *peer = NULL;
#ifdef HAVE_EPOLL
struct chanlist *epollo;
#endif
struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
/* single is set if only one destination is enabled */
int single = outgoing && !AST_LIST_NEXT(outgoing, node);
int caller_entertained = outgoing
&& ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
struct ast_party_connected_line connected_caller;
struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
int cc_recall_core_id;
int is_cc_recall;
int cc_frame_received = 0;
int num_ringing = 0;
struct timeval start = ast_tvnow();
ast_party_connected_line_init(&connected_caller);
if (single) {
/* Turn off hold music, etc */
if (!caller_entertained) {
ast_deactivate_generator(in);
/* If we are calling a single channel, and not providing ringback or music, */
/* then, make them compatible for in-band tone purpose */
if (ast_channel_make_compatible(outgoing->chan, in) < 0) {
/* If these channels can not be made compatible,
* there is no point in continuing. The bridge
* will just fail if it gets that far.
*/
*to = -1;
strcpy(pa->status, "CONGESTION");
ast_cdr_failed(ast_channel_cdr(in));
return NULL;
}
}
if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
&& !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
ast_channel_lock(outgoing->chan);
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(outgoing->chan));
ast_channel_unlock(outgoing->chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(outgoing->chan, in, &connected_caller, 0) &&
ast_channel_connected_line_macro(outgoing->chan, in, &connected_caller, 1, 0)) {
ast_channel_update_connected_line(in, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
}
}
is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
#ifdef HAVE_EPOLL
AST_LIST_TRAVERSE(out_chans, epollo, node) {
ast_poll_channel_add(in, epollo->chan);
}
#endif
while ((*to = ast_remaining_ms(start, orig)) && !peer) {
struct chanlist *o;
int pos = 0; /* how many channels do we handle */
int numlines = prestart;
struct ast_channel *winner;
struct ast_channel *watchers[AST_MAX_WATCHERS];
watchers[pos++] = in;
AST_LIST_TRAVERSE(out_chans, o, node) {
/* Keep track of important channels */
if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
watchers[pos++] = o->chan;
numlines++;
}
if (pos == 1) { /* only the input channel is available */
if (numlines == (num.busy + num.congestion + num.nochan)) {
ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
if (num.busy)
strcpy(pa->status, "BUSY");
else if (num.congestion)
strcpy(pa->status, "CONGESTION");
else if (num.nochan)
strcpy(pa->status, "CHANUNAVAIL");
} else {
ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
}
*to = 0;
if (is_cc_recall) {
ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
}
return NULL;
}
winner = ast_waitfor_n(watchers, pos, to);
AST_LIST_TRAVERSE(out_chans, o, node) {
struct ast_frame *f;
struct ast_channel *c = o->chan;
if (c == NULL)
continue;
if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
if (!peer) {
ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
if (o->pending_connected_update) {
if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
ast_channel_update_connected_line(in, &o->connected, NULL);
}
} else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
ast_channel_lock(c);
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
ast_channel_unlock(c);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
ast_channel_update_connected_line(in, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
}
}
if (o->aoc_s_rate_list) {
size_t encoded_size;
struct ast_aoc_encoded *encoded;
if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
ast_aoc_destroy_encoded(encoded);
}
}
peer = c;
ast_copy_flags64(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
DIAL_NOFORWARDHTML);
ast_channel_dialcontext_set(c, "");
ast_channel_exten_set(c, "");
}
continue;
}
if (c != winner)
continue;
/* here, o->chan == c == winner */
if (!ast_strlen_zero(ast_channel_call_forward(c))) {
pa->sentringing = 0;
if (!ignore_cc && (f = ast_read(c))) {
if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
/* This channel is forwarding the call, and is capable of CC, so
* be sure to add the new device interface to the list
*/
ast_handle_cc_control_frame(in, c, f->data.ptr);
}
ast_frfree(f);
}
if (o->pending_connected_update) {
/*
* Re-seed the chanlist's connected line information with
* previously acquired connected line info from the incoming
* channel. The previously acquired connected line info could
* have been set through the CONNECTED_LINE dialplan function.
*/
o->pending_connected_update = 0;
ast_channel_lock(in);
ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
ast_channel_unlock(in);
}
do_forward(o, &num, peerflags, single, caller_entertained, to,
forced_clid, stored_clid);
if (single && o->chan
&& !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
&& !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
ast_channel_lock(o->chan);
ast_connected_line_copy_from_caller(&connected_caller,
ast_channel_caller(o->chan));
ast_channel_unlock(o->chan);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(o->chan, in, &connected_caller, 0) &&
ast_channel_connected_line_macro(o->chan, in, &connected_caller, 1, 0)) {
ast_channel_update_connected_line(in, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
}
continue;
}
f = ast_read(winner);
if (!f) {
ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
#ifdef HAVE_EPOLL
ast_poll_channel_del(in, c);
#endif
ast_hangup(c);
c = o->chan = NULL;
ast_clear_flag64(o, DIAL_STILLGOING);
handle_cause(ast_channel_hangupcause(in), &num);
continue;
}
switch (f->frametype) {
case AST_FRAME_CONTROL:
switch (f->subclass.integer) {
case AST_CONTROL_ANSWER:
/* This is our guy if someone answered. */
if (!peer) {
ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
if (o->pending_connected_update) {
if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
ast_channel_update_connected_line(in, &o->connected, NULL);
}
} else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
ast_channel_lock(c);
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
ast_channel_unlock(c);
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
ast_channel_update_connected_line(in, &connected_caller, NULL);
}
ast_party_connected_line_free(&connected_caller);
}
}
if (o->aoc_s_rate_list) {
size_t encoded_size;
struct ast_aoc_encoded *encoded;
if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
ast_aoc_destroy_encoded(encoded);
}
}
peer = c;
if (ast_channel_cdr(peer)) {
ast_channel_cdr(peer)->answer = ast_tvnow();
ast_channel_cdr(peer)->disposition = AST_CDR_ANSWERED;
}
ast_copy_flags64(peerflags, o,
OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
OPT_CALLEE_PARK | OPT_CALLER_PARK |
OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
DIAL_NOFORWARDHTML);
ast_channel_dialcontext_set(c, "");
ast_channel_exten_set(c, "");
if (CAN_EARLY_BRIDGE(peerflags, in, peer))
/* Setup early bridge if appropriate */
ast_channel_early_bridge(in, peer);
}
/* If call has been answered, then the eventual hangup is likely to be normal hangup */
ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
break;
case AST_CONTROL_BUSY:
ast_verb(3, "%s is busy\n", ast_channel_name(c));
ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
ast_hangup(c);
c = o->chan = NULL;
ast_clear_flag64(o, DIAL_STILLGOING);
handle_cause(AST_CAUSE_BUSY, &num);
break;
case AST_CONTROL_CONGESTION:
ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
ast_hangup(c);
c = o->chan = NULL;
ast_clear_flag64(o, DIAL_STILLGOING);
handle_cause(AST_CAUSE_CONGESTION, &num);
break;
case AST_CONTROL_RINGING:
/* This is a tricky area to get right when using a native
* CC agent. The reason is that we do the best we can to send only a
* single ringing notification to the caller.
*
* Call completion complicates the logic used here. CCNR is typically
* offered during a ringing message. Let's say that party A calls
* parties B, C, and D. B and C do not support CC requests, but D
* does. If we were to receive a ringing notification from B before
* the others, then we would end up sending a ringing message to
* A with no CCNR offer present.
*
* The approach that we have taken is that if we receive a ringing
* response from a party and no CCNR offer is present, we need to
* wait. Specifically, we need to wait until either a) a called party
* offers CCNR in its ringing response or b) all called parties have
* responded in some way to our call and none offers CCNR.
*
* The drawback to this is that if one of the parties has a delayed
* response or, god forbid, one just plain doesn't respond to our
* outgoing call, then this will result in a significant delay between
* when the caller places the call and hears ringback.
*
* Note also that if CC is disabled for this call, then it is perfectly
* fine for ringing frames to get sent through.
*/
++num_ringing;
if (ignore_cc || cc_frame_received || num_ringing == numlines) {
ast_verb(3, "%s is ringing\n", ast_channel_name(c));
/* Setup early media if appropriate */
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
ast_indicate(in, AST_CONTROL_RINGING);
pa->sentringing++;
}
}
break;
case AST_CONTROL_PROGRESS:
ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
/* Setup early media if appropriate */
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
if (single || (!single && !pa->sentringing)) {
ast_indicate(in, AST_CONTROL_PROGRESS);
}
}
if (!ast_strlen_zero(dtmf_progress)) {
ast_verb(3,
"Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
dtmf_progress);
ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
}
break;
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
case AST_CONTROL_SRCCHANGE:
if (!single || caller_entertained) {
break;
}
ast_verb(3, "%s requested media update control %d, passing it to %s\n",
ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
ast_indicate(in, f->subclass.integer);
break;
case AST_CONTROL_CONNECTED_LINE:
if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
break;
}
if (!single) {
struct ast_party_connected_line connected;
ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
ast_channel_name(c), ast_channel_name(in));
ast_party_connected_line_set_init(&connected, &o->connected);
ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
ast_party_connected_line_set(&o->connected, &connected, NULL);
ast_party_connected_line_free(&connected);
o->pending_connected_update = 1;
break;
}
if (ast_channel_connected_line_sub(c, in, f, 1) &&
ast_channel_connected_line_macro(c, in, f, 1, 1)) {
ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
}
break;
case AST_CONTROL_AOC:
{
struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
ast_aoc_destroy_decoded(o->aoc_s_rate_list);
o->aoc_s_rate_list = decoded;
} else {
ast_aoc_destroy_decoded(decoded);
}
}
break;
case AST_CONTROL_REDIRECTING:
if (!single) {
/*
* Redirecting updates to the caller make sense only on single
* calls.
*/
break;
}
if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
break;
}
ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
ast_channel_name(c), ast_channel_name(in));
if (ast_channel_redirecting_sub(c, in, f, 1) &&
ast_channel_redirecting_macro(c, in, f, 1, 1)) {
ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
}
pa->sentringing = 0;
break;
case AST_CONTROL_PROCEEDING:
ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
if (single && !caller_entertained
&& CAN_EARLY_BRIDGE(peerflags, in, c)) {
ast_channel_early_bridge(in, c);
}
if (!ast_test_flag64(outgoing, OPT_RINGBACK))
ast_indicate(in, AST_CONTROL_PROCEEDING);
break;
case AST_CONTROL_HOLD:
/* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
break;
case AST_CONTROL_UNHOLD:
/* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
ast_indicate(in, AST_CONTROL_UNHOLD);
break;
case AST_CONTROL_OFFHOOK:
case AST_CONTROL_FLASH:
/* Ignore going off hook and flash */
break;
case AST_CONTROL_CC:
if (!ignore_cc) {
ast_handle_cc_control_frame(in, c, f->data.ptr);
cc_frame_received = 1;
}
break;
case AST_CONTROL_PVT_CAUSE_CODE:
ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
break;
case -1:
if (single && !caller_entertained) {
ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
ast_indicate(in, -1);
pa->sentringing = 0;
}
break;
default:
ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
break;
}
break;
case AST_FRAME_VOICE:
case AST_FRAME_IMAGE:
if (caller_entertained) {
break;
}
/* Fall through */
case AST_FRAME_TEXT:
if (single && ast_write(in, f)) {
ast_log(LOG_WARNING, "Unable to write frametype: %d\n",
f->frametype);
}
break;
case AST_FRAME_HTML:
if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
&& ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
ast_log(LOG_WARNING, "Unable to send URL\n");
}
break;
default:
break;
}
ast_frfree(f);
} /* end for */
if (winner == in) {
struct ast_frame *f = ast_read(in);
#if 0
if (f && (f->frametype != AST_FRAME_VOICE))
printf("Frame type: %d, %d\n", f->frametype, f->subclass);
else if (!f || (f->frametype != AST_FRAME_VOICE))
printf("Hangup received on %s\n", in->name);
#endif
if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
/* Got hung up */
*to = -1;
strcpy(pa->status, "CANCEL");
ast_cdr_noanswer(ast_channel_cdr(in));
if (f) {
if (f->data.uint32) {
ast_channel_hangupcause_set(in, f->data.uint32);
}
ast_frfree(f);
}
if (is_cc_recall) {
ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
}
return NULL;
}
/* now f is guaranteed non-NULL */
if (f->frametype == AST_FRAME_DTMF) {
if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
const char *context;
ast_channel_lock(in);
context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
*to = 0;
ast_cdr_noanswer(ast_channel_cdr(in));
*result = f->subclass.integer;
strcpy(pa->status, "CANCEL");
ast_frfree(f);
ast_channel_unlock(in);
if (is_cc_recall) {
ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
}
return NULL;
}
ast_channel_unlock(in);
}
if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
detect_disconnect(in, f->subclass.integer, &featurecode)) {
ast_verb(3, "User requested call disconnect.\n");
*to = 0;
strcpy(pa->status, "CANCEL");
ast_cdr_noanswer(ast_channel_cdr(in));
ast_frfree(f);
if (is_cc_recall) {
ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
}
return NULL;
}
}
/* Send the frame from the in channel to all outgoing channels. */
AST_LIST_TRAVERSE(out_chans, o, node) {
if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
/* This outgoing channel has died so don't send the frame to it. */
continue;
}
switch (f->frametype) {
case AST_FRAME_HTML:
/* Forward HTML stuff */
if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
&& ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
ast_log(LOG_WARNING, "Unable to send URL\n");
}
break;
case AST_FRAME_VOICE:
case AST_FRAME_IMAGE:
if (!single || caller_entertained) {
/*
* We are calling multiple parties or caller is being
* entertained and has thus not been made compatible.
* No need to check any other called parties.
*/
goto skip_frame;
}
/* Fall through */
case AST_FRAME_TEXT:
case AST_FRAME_DTMF_BEGIN:
case AST_FRAME_DTMF_END:
if (ast_write(o->chan, f)) {
ast_log(LOG_WARNING, "Unable to forward frametype: %d\n",
f->frametype);
}
break;
case AST_FRAME_CONTROL:
switch (f->subclass.integer) {
case AST_CONTROL_HOLD:
ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
break;
case AST_CONTROL_UNHOLD:
ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
ast_indicate(o->chan, AST_CONTROL_UNHOLD);
break;
case AST_CONTROL_VIDUPDATE:
case AST_CONTROL_SRCUPDATE:
case AST_CONTROL_SRCCHANGE:
if (!single || caller_entertained) {
/*
* We are calling multiple parties or caller is being
* entertained and has thus not been made compatible.
* No need to check any other called parties.
*/
goto skip_frame;
}
ast_verb(3, "%s requested media update control %d, passing it to %s\n",
ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
ast_indicate(o->chan, f->subclass.integer);
break;
case AST_CONTROL_CONNECTED_LINE:
if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
}
break;
case AST_CONTROL_REDIRECTING:
if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
}
break;
default:
/* We are not going to do anything with this frame. */
goto skip_frame;
}
break;
default:
/* We are not going to do anything with this frame. */
goto skip_frame;
}
}
skip_frame:;
ast_frfree(f);
}
}
if (!*to) {
ast_verb(3, "Nobody picked up in %d ms\n", orig);
}
if (!*to || ast_check_hangup(in)) {
ast_cdr_noanswer(ast_channel_cdr(in));
}
#ifdef HAVE_EPOLL
AST_LIST_TRAVERSE(out_chans, epollo, node) {
if (epollo->chan)
ast_poll_channel_del(in, epollo->chan);
}
#endif
if (is_cc_recall) {
ast_cc_completed(in, "Recall completed!");
}
return peer;
}
struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Dialing Application" , .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, .load_pri = AST_MODPRI_DEFAULT, } [static] |
Definition at line 3261 of file app_dial.c.
const char app[] = "Dial" [static] |
Definition at line 562 of file app_dial.c.
struct ast_module_info* ast_module_info = &__mod_info [static] |
Definition at line 3261 of file app_dial.c.
struct ast_app_option dial_exec_options[128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'a' ] = { .flag = (1LLU << 40) }, [ 'b' ] = { .flag = (1LLU << 41) , .arg_index = OPT_ARG_PREDIAL_CALLEE + 1 }, [ 'B' ] = { .flag = (1LLU << 42) , .arg_index = OPT_ARG_PREDIAL_CALLER + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = (1LLU << 34) }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'e' ] = { .flag = (1LLU << 35) }, [ 'f' ] = { .flag = OPT_FORCECLID , .arg_index = OPT_ARG_FORCECLID + 1 }, [ 'F' ] = { .flag = (1LLU << 36) , .arg_index = OPT_ARG_CALLEE_GO_ON + 1 }, [ 'g' ] = { .flag = OPT_GO_ON }, [ 'G' ] = { .flag = OPT_GOTO , .arg_index = OPT_ARG_GOTO + 1 }, [ 'h' ] = { .flag = OPT_CALLEE_HANGUP }, [ 'H' ] = { .flag = OPT_CALLER_HANGUP }, [ 'i' ] = { .flag = OPT_IGNORE_FORWARDING }, [ 'I' ] = { .flag = OPT_IGNORE_CONNECTEDLINE }, [ 'k' ] = { .flag = OPT_CALLEE_PARK }, [ 'K' ] = { .flag = OPT_CALLER_PARK }, [ 'L' ] = { .flag = OPT_DURATION_LIMIT , .arg_index = OPT_ARG_DURATION_LIMIT + 1 }, [ 'm' ] = { .flag = OPT_MUSICBACK , .arg_index = OPT_ARG_MUSICBACK + 1 }, [ 'M' ] = { .flag = OPT_CALLEE_MACRO , .arg_index = OPT_ARG_CALLEE_MACRO + 1 }, [ 'n' ] = { .flag = OPT_SCREEN_NOINTRO , .arg_index = OPT_ARG_SCREEN_NOINTRO + 1 }, [ 'N' ] = { .flag = OPT_SCREEN_NOCALLERID }, [ 'o' ] = { .flag = OPT_ORIGINAL_CLID , .arg_index = OPT_ARG_ORIGINAL_CLID + 1 }, [ 'O' ] = { .flag = OPT_OPERMODE , .arg_index = OPT_ARG_OPERMODE + 1 }, [ 'p' ] = { .flag = OPT_SCREENING }, [ 'P' ] = { .flag = OPT_PRIVACY , .arg_index = OPT_ARG_PRIVACY + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK , .arg_index = OPT_ARG_RINGBACK + 1 }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 's' ] = { .flag = (1LLU << 38) , .arg_index = OPT_ARG_FORCE_CID_TAG + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'u' ] = { .flag = (1LLU << 39) , .arg_index = OPT_ARG_FORCE_CID_PRES + 1 }, [ 'U' ] = { .flag = OPT_CALLEE_GOSUB , .arg_index = OPT_ARG_CALLEE_GOSUB + 1 }, [ 'w' ] = { .flag = OPT_CALLEE_MONITOR }, [ 'W' ] = { .flag = OPT_CALLER_MONITOR }, [ 'x' ] = { .flag = OPT_CALLEE_MIXMONITOR }, [ 'X' ] = { .flag = OPT_CALLER_MIXMONITOR }, [ 'z' ] = { .flag = (1LLU << 37) }, } [static] |
Definition at line 678 of file app_dial.c.
Referenced by dial_exec_full().
const char rapp[] = "RetryDial" [static] |
Definition at line 563 of file app_dial.c.