Asterisk - The Open Source Telephony Project GIT-master-7988d11
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Data Structures | Macros | Enumerations | Functions | Variables
app_dial.c File Reference
Dial plan applications

dial() & retrydial() - Trivial application to dial a channel and send an URL on answer More...

#include "asterisk.h"
#include <sys/time.h>
#include <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/manager.h"
#include "asterisk/privacy.h"
#include "asterisk/stringfields.h"
#include "asterisk/dsp.h"
#include "asterisk/aoc.h"
#include "asterisk/ccss.h"
#include "asterisk/indications.h"
#include "asterisk/framehook.h"
#include "asterisk/dial.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/bridge_after.h"
#include "asterisk/features_config.h"
#include "asterisk/max_forwards.h"
#include "asterisk/stream.h"
Include dependency graph for app_dial.c:

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
 

Macros

#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_HANGUPCAUSE   (1LLU << 44)
 
#define OPT_HEARPULSING   (1LLU << 45)
 
#define OPT_PEER_H   (1LLU << 35)
 
#define OPT_PREDIAL_CALLEE   (1LLU << 41)
 
#define OPT_PREDIAL_CALLER   (1LLU << 42)
 
#define OPT_RING_WITH_EARLY_MEDIA   (1LLU << 43)
 
#define OPT_TOPOLOGY_PRESERVE   (1LLU << 46)
 

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_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_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_HANGUPCAUSE , OPT_ARG_ARRAY_SIZE
}
 

Functions

static void __reg_module (void)
 
static void __unreg_module (void)
 
struct ast_module * AST_MODULE_SELF_SYM (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 hangupcause)
 
static int load_module (void)
 
static int onedigit_goto (struct ast_channel *chan, const char *context, char exten, int pri)
 
static void publish_dial_end_event (struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
 
static int retrydial_exec (struct ast_channel *chan, const char *data)
 
static void set_duration_var (struct ast_channel *chan, const char *var_base, int64_t duration)
 
static void setup_peer_after_bridge_goto (struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
 
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 void topology_ds_destroy (void *data)
 
static int unload_module (void)
 
static void update_connected_line_from_peer (struct ast_channel *chan, struct ast_channel *peer, int is_caller)
 
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_answer, int *to_progress, struct ast_flags64 *peerflags, char *opt_args[], struct privacy_args *pa, const struct cause_args *num_in, int *result, char *dtmf_progress, char *mf_progress, char *mf_wink, char *sf_progress, char *sf_wink, const int hearpulsing, const int ignore_cc, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid, struct ast_bridge_config *config)
 

Variables

static struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Dialing Application" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, }
 
static const char app [] = "Dial"
 
static const struct ast_module_info * ast_module_info = &__mod_info
 
static const struct ast_app_option dial_exec_options [128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'a' ] = { .flag = OPT_CALLER_ANSWER }, [ 'b' ] = { .flag = OPT_PREDIAL_CALLEE , .arg_index = OPT_ARG_PREDIAL_CALLEE + 1 }, [ 'B' ] = { .flag = OPT_PREDIAL_CALLER , .arg_index = OPT_ARG_PREDIAL_CALLER + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = OPT_CANCEL_ELSEWHERE }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'E' ] = { .flag = OPT_HEARPULSING }, [ 'e' ] = { .flag = OPT_PEER_H }, [ 'f' ] = { .flag = OPT_FORCECLID , .arg_index = OPT_ARG_FORCECLID + 1 }, [ 'F' ] = { .flag = OPT_CALLEE_GO_ON , .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 }, [ 'j' ] = { .flag = OPT_TOPOLOGY_PRESERVE }, [ '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 }, [ '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 }, [ 'Q' ] = { .flag = OPT_HANGUPCAUSE , .arg_index = OPT_ARG_HANGUPCAUSE + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK , .arg_index = OPT_ARG_RINGBACK + 1 }, [ 'R' ] = { .flag = OPT_RING_WITH_EARLY_MEDIA }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 's' ] = { .flag = OPT_FORCE_CID_TAG , .arg_index = OPT_ARG_FORCE_CID_TAG + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'u' ] = { .flag = OPT_FORCE_CID_PRES , .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 = OPT_CANCEL_TIMEOUT }, }
 
static const char rapp [] = "RetryDial"
 
static const struct ast_datastore_info topology_ds_info
 

Detailed Description

dial() & retrydial() - Trivial application to dial a channel and send an URL on answer

Author
Mark Spencer marks.nosp@m.ter@.nosp@m.digiu.nosp@m.m.co.nosp@m.m

Definition in file app_dial.c.

Macro Definition Documentation

◆ AST_MAX_WATCHERS

#define AST_MAX_WATCHERS   256

Definition at line 865 of file app_dial.c.

◆ CAN_EARLY_BRIDGE

#define CAN_EARLY_BRIDGE (   flags,
  chan,
  peer 
)

Definition at line 794 of file app_dial.c.

803 {
804 AST_LIST_ENTRY(chanlist) node;
805 struct ast_channel *chan;
806 /*! Channel interface dialing string (is tech/number). (Stored in stuff[]) */
807 const char *interface;
808 /*! Channel technology name. (Stored in stuff[]) */
809 const char *tech;
810 /*! Channel device addressing. (Stored in stuff[]) */
811 const char *number;
812 /*! Original channel name. Must be freed. Could be NULL if allocation failed. */
813 char *orig_chan_name;
814 uint64_t flags;
815 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
816 struct ast_party_connected_line connected;
817 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
818 unsigned int pending_connected_update:1;
819 struct ast_aoc_decoded *aoc_s_rate_list;
820 /*! The interface, tech, and number strings are stuffed here. */
821 char stuff[0];
822};
823
824AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
825
826static void topology_ds_destroy(void *data) {
827 struct ast_stream_topology *top = data;
828 ast_stream_topology_free(top);
829}
830
831static const struct ast_datastore_info topology_ds_info = {
832 .type = "app_dial_topology_preserve",
833 .destroy = topology_ds_destroy,
834};
835
836static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
837
838static void chanlist_free(struct chanlist *outgoing)
839{
840 ast_party_connected_line_free(&outgoing->connected);
841 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
842 ast_free(outgoing->orig_chan_name);
843 ast_free(outgoing);
844}
845
846static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
847{
848 /* Hang up a tree of stuff */
849 struct chanlist *outgoing;
850
851 while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
852 /* Hangup any existing lines we have open */
853 if (outgoing->chan && (outgoing->chan != exception)) {
854 if (hangupcause >= 0) {
855 /* This is for the channel drivers */
856 ast_channel_hangupcause_set(outgoing->chan, hangupcause);
857 }
858 ast_hangup(outgoing->chan);
859 }
860 chanlist_free(outgoing);
861 }
862}
863
864#define AST_MAX_WATCHERS 256
865
866/*
867 * argument to handle_cause() and other functions.
868 */
869struct cause_args {
870 struct ast_channel *chan;
871 int busy;
872 int congestion;
873 int nochan;
874};
875
876static void handle_cause(int cause, struct cause_args *num)
877{
878 switch(cause) {
879 case AST_CAUSE_BUSY:
880 num->busy++;
881 break;
882 case AST_CAUSE_CONGESTION:
883 num->congestion++;
884 break;
885 case AST_CAUSE_NO_ROUTE_DESTINATION:
886 case AST_CAUSE_UNREGISTERED:
887 num->nochan++;
888 break;
889 case AST_CAUSE_NO_ANSWER:
890 case AST_CAUSE_NORMAL_CLEARING:
891 break;
892 default:
893 num->nochan++;
894 break;
895 }
896}
897
898static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
899{
900 char rexten[2] = { exten, '\0' };
901
902 if (context) {
903 if (!ast_goto_if_exists(chan, context, rexten, pri))
904 return 1;
905 } else {
906 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
907 return 1;
908 }
909 return 0;
910}
911
912/* do not call with chan lock held */
913static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
914{
915 const char *context;
916 const char *exten;
917
918 ast_channel_lock(chan);
919 context = ast_strdupa(ast_channel_context(chan));
920 exten = ast_strdupa(ast_channel_exten(chan));
921 ast_channel_unlock(chan);
922
923 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
924}
925
926/*!
927 * helper function for wait_for_answer()
928 *
929 * \param o Outgoing call channel list.
930 * \param num Incoming call channel cause accumulation
931 * \param peerflags Dial option flags
932 * \param single TRUE if there is only one outgoing call.
933 * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
934 * \param to Remaining call timeout time.
935 * \param forced_clid OPT_FORCECLID caller id to send
936 * \param stored_clid Caller id representing the called party if needed
937 *
938 * XXX this code is highly suspicious, as it essentially overwrites
939 * the outgoing channel without properly deleting it.
940 *
941 * \todo eventually this function should be integrated into and replaced by ast_call_forward()
942 */
943static void do_forward(struct chanlist *o, struct cause_args *num,
944 struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
945 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
946{
947 char tmpchan[256];
948 char forwarder[AST_CHANNEL_NAME];
949 struct ast_channel *original = o->chan;
950 struct ast_channel *c = o->chan; /* the winner */
951 struct ast_channel *in = num->chan; /* the input channel */
952 char *stuff;
953 const char *tech;
954 int cause;
955 struct ast_party_caller caller;
956
957 ast_copy_string(forwarder, ast_channel_name(c), sizeof(forwarder));
958 ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
959 if ((stuff = strchr(tmpchan, '/'))) {
960 *stuff++ = '\0';
961 tech = tmpchan;
962 } else {
963 const char *forward_context;
964 ast_channel_lock(c);
965 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
966 if (ast_strlen_zero(forward_context)) {
967 forward_context = NULL;
968 }
969 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
970 ast_channel_unlock(c);
971 stuff = tmpchan;
972 tech = "Local";
973 }
974 if (!strcasecmp(tech, "Local")) {
975 /*
976 * Drop the connected line update block for local channels since
977 * this is going to run dialplan and the user can change his
978 * mind about what connected line information he wants to send.
979 */
980 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
981 }
982
983 /* Before processing channel, go ahead and check for forwarding */
984 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
985 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
986 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
987 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
988 ast_channel_publish_dial_forward(in, original, NULL, NULL, "CANCEL",
989 ast_channel_call_forward(original));
990 c = o->chan = NULL;
991 cause = AST_CAUSE_BUSY;
992 } else {
993 struct ast_stream_topology *topology;
994
995 ast_channel_lock(in);
996 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(in));
997 ast_channel_unlock(in);
998
999 /* Setup parameters */
1000 c = o->chan = ast_request_with_stream_topology(tech, topology, NULL, in, stuff, &cause);
1001
1002 ast_stream_topology_free(topology);
1003
1004 if (c) {
1005 if (single && !caller_entertained) {
1006 ast_channel_make_compatible(in, o->chan);
1007 }
1008 ast_channel_lock_both(in, o->chan);
1009 ast_channel_inherit_variables(in, o->chan);
1010 ast_channel_datastore_inherit(in, o->chan);
1011 pbx_builtin_setvar_helper(o->chan, "FORWARDERNAME", forwarder);
1012 ast_max_forwards_decrement(o->chan);
1013 ast_channel_unlock(in);
1014 ast_channel_unlock(o->chan);
1015 /* When a call is forwarded, we don't want to track new interfaces
1016 * dialed for CC purposes. Setting the done flag will ensure that
1017 * any Dial operations that happen later won't record CC interfaces.
1018 */
1019 ast_ignore_cc(o->chan);
1020 ast_verb(3, "Not accepting call completion offers from call-forward recipient %s\n",
1021 ast_channel_name(o->chan));
1022 } else
1023 ast_log(LOG_NOTICE,
1024 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
1025 tech, stuff, cause);
1026 }
1027 if (!c) {
1028 ast_channel_publish_dial(in, original, stuff, "BUSY");
1029 ast_clear_flag64(o, DIAL_STILLGOING);
1030 handle_cause(cause, num);
1031 ast_hangup(original);
1032 } else {
1033 ast_channel_lock_both(c, original);
1034 ast_party_redirecting_copy(ast_channel_redirecting(c),
1035 ast_channel_redirecting(original));
1036 ast_channel_unlock(c);
1037 ast_channel_unlock(original);
1038
1039 ast_channel_lock_both(c, in);
1040
1041 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
1042 ast_rtp_instance_early_bridge_make_compatible(c, in);
1043 }
1044
1045 if (!ast_channel_redirecting(c)->from.number.valid
1046 || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
1047 /*
1048 * The call was not previously redirected so it is
1049 * now redirected from this number.
1050 */
1051 ast_party_number_free(&ast_channel_redirecting(c)->from.number);
1052 ast_party_number_init(&ast_channel_redirecting(c)->from.number);
1053 ast_channel_redirecting(c)->from.number.valid = 1;
1054 ast_channel_redirecting(c)->from.number.str =
1055 ast_strdup(ast_channel_exten(in));
1056 }
1057
1058 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
1059
1060 /* Determine CallerID to store in outgoing channel. */
1061 ast_party_caller_set_init(&caller, ast_channel_caller(c));
1062 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1063 caller.id = *stored_clid;
1064 ast_channel_set_caller_event(c, &caller, NULL);
1065 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1066 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
1067 ast_channel_caller(c)->id.number.str, NULL))) {
1068 /*
1069 * The new channel has no preset CallerID number by the channel
1070 * driver. Use the dialplan extension and hint name.
1071 */
1072 caller.id = *stored_clid;
1073 ast_channel_set_caller_event(c, &caller, NULL);
1074 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1075 } else {
1076 ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
1077 }
1078
1079 /* Determine CallerID for outgoing channel to send. */
1080 if (ast_test_flag64(o, OPT_FORCECLID)) {
1081 struct ast_party_connected_line connected;
1082
1083 ast_party_connected_line_init(&connected);
1084 connected.id = *forced_clid;
1085 ast_party_connected_line_copy(ast_channel_connected(c), &connected);
1086 } else {
1087 ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
1088 }
1089
1090 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
1091
1092 ast_channel_appl_set(c, "AppDial");
1093 ast_channel_data_set(c, "(Outgoing Line)");
1094 ast_channel_publish_snapshot(c);
1095
1096 ast_channel_unlock(in);
1097 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1098 struct ast_party_redirecting redirecting;
1099
1100 /*
1101 * Redirecting updates to the caller make sense only on single
1102 * calls.
1103 *
1104 * Need to re-evalute if unlocking is still required here as macro is gone
1105 */
1106 ast_party_redirecting_init(&redirecting);
1107 ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
1108 ast_channel_unlock(c);
1109 if (ast_channel_redirecting_sub(c, in, &redirecting, 0)) {
1110 ast_channel_update_redirecting(in, &redirecting, NULL);
1111 }
1112 ast_party_redirecting_free(&redirecting);
1113 } else {
1114 ast_channel_unlock(c);
1115 }
1116
1117 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1118 *to = -1;
1119 }
1120
1121 if (ast_call(c, stuff, 0)) {
1122 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1123 tech, stuff);
1124 ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1125 ast_clear_flag64(o, DIAL_STILLGOING);
1126 ast_hangup(original);
1127 ast_hangup(c);
1128 c = o->chan = NULL;
1129 num->nochan++;
1130 } else {
1131 ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1132 ast_channel_call_forward(original));
1133
1134 ast_channel_publish_dial(in, c, stuff, NULL);
1135
1136 /* Hangup the original channel now, in case we needed it */
1137 ast_hangup(original);
1138 }
1139 if (single && !caller_entertained) {
1140 ast_indicate(in, -1);
1141 }
1142 }
1143}
1144
1145/* argument used for some functions. */
1146struct privacy_args {
1147 int sentringing;
1148 int privdb_val;
1149 char privcid[256];
1150 char privintro[1024];
1151 char status[256];
1152 int canceled;
1153};
1154
1155static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
1156{
1157 struct chanlist *outgoing;
1158 AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1159 if (!outgoing->chan || outgoing->chan == exception) {
1160 continue;
1161 }
1162 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1163 }
1164}
1165
1166/*!
1167 * \internal
1168 * \brief Update connected line on chan from peer.
1169 * \since 13.6.0
1170 *
1171 * \param chan Channel to get connected line updated.
1172 * \param peer Channel providing connected line information.
1173 * \param is_caller Non-zero if chan is the calling channel.
1174 */
1175static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
1176{
1177 struct ast_party_connected_line connected_caller;
1178
1179 ast_party_connected_line_init(&connected_caller);
1180
1181 ast_channel_lock(peer);
1182 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1183 ast_channel_unlock(peer);
1184 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1185 if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)) {
1186 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1187 }
1188 ast_party_connected_line_free(&connected_caller);
1189}
1190
1191/*!
1192 * \internal
1193 * \pre chan is locked
1194 */
1195static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
1196{
1197 char buf[32];
1198 char full_var_name[128];
1199
1200 snprintf(buf, sizeof(buf), "%" PRId64, duration / 1000);
1201 pbx_builtin_setvar_helper(chan, var_base, buf);
1202
1203 snprintf(full_var_name, sizeof(full_var_name), "%s_MS", var_base);
1204 snprintf(buf, sizeof(buf), "%" PRId64, duration);
1205 pbx_builtin_setvar_helper(chan, full_var_name, buf);
1206}
1207
1208static struct ast_channel *wait_for_answer(struct ast_channel *in,
1209 struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags,
1210 char *opt_args[],
1211 struct privacy_args *pa,
1212 const struct cause_args *num_in, int *result, char *dtmf_progress,
1213 char *mf_progress, char *mf_wink,
1214 char *sf_progress, char *sf_wink,
1215 const int hearpulsing,
1216 const int ignore_cc,
1217 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
1218 struct ast_bridge_config *config)
1219{
1220 struct cause_args num = *num_in;
1221 int prestart = num.busy + num.congestion + num.nochan;
1222 int orig_answer_to = *to_answer;
1223 int orig_progress_to = *to_progress;
1224 struct ast_channel *peer = NULL;
1225 struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1226 /* single is set if only one destination is enabled */
1227 int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1228 int caller_entertained = outgoing
1229 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1230 struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1231 int cc_recall_core_id;
1232 int is_cc_recall;
1233 int cc_frame_received = 0;
1234 int num_ringing = 0;
1235 int sent_ring = 0;
1236 int sent_progress = 0, sent_wink = 0;
1237 struct timeval start = ast_tvnow();
1238 SCOPE_ENTER(3, "%s\n", ast_channel_name(in));
1239
1240 if (single) {
1241 /* Turn off hold music, etc */
1242 if (!caller_entertained) {
1243 ast_deactivate_generator(in);
1244 /* If we are calling a single channel, and not providing ringback or music, */
1245 /* then, make them compatible for in-band tone purpose */
1246 if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1247 /* If these channels can not be made compatible,
1248 * there is no point in continuing. The bridge
1249 * will just fail if it gets that far.
1250 */
1251 *to_answer = -1;
1252 strcpy(pa->status, "CONGESTION");
1253 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1254 SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
1255 ast_channel_name(in), ast_channel_name(outgoing->chan));
1256 }
1257 }
1258
1259 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1260 && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1261 update_connected_line_from_peer(in, outgoing->chan, 1);
1262 }
1263 }
1264
1265 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1266
1267 while ((*to_answer = ast_remaining_ms(start, orig_answer_to)) && (*to_progress = ast_remaining_ms(start, orig_progress_to)) && !peer) {
1268 struct chanlist *o;
1269 int pos = 0; /* how many channels do we handle */
1270 int numlines = prestart;
1271 struct ast_channel *winner;
1272 struct ast_channel *watchers[AST_MAX_WATCHERS];
1273
1274 watchers[pos++] = in;
1275 AST_LIST_TRAVERSE(out_chans, o, node) {
1276 /* Keep track of important channels */
1277 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1278 watchers[pos++] = o->chan;
1279 numlines++;
1280 }
1281 if (pos == 1) { /* only the input channel is available */
1282 if (numlines == (num.busy + num.congestion + num.nochan)) {
1283 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1284 if (num.busy)
1285 strcpy(pa->status, "BUSY");
1286 else if (num.congestion)
1287 strcpy(pa->status, "CONGESTION");
1288 else if (num.nochan)
1289 strcpy(pa->status, "CHANUNAVAIL");
1290 } else {
1291 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1292 }
1293 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1294 if (is_cc_recall) {
1295 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1296 }
1297 SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
1298 }
1299
1300 /* If progress timeout is active, use that if it's the shorter of the 2 timeouts. */
1301 winner = ast_waitfor_n(watchers, pos, *to_progress > 0 && (*to_answer < 0 || *to_progress < *to_answer) ? to_progress : to_answer);
1302
1303 AST_LIST_TRAVERSE(out_chans, o, node) {
1304 int res = 0;
1305 struct ast_frame *f;
1306 struct ast_channel *c = o->chan;
1307
1308 if (c == NULL)
1309 continue;
1310 if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1311 if (!peer) {
1312 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1313 if (o->orig_chan_name
1314 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1315 /*
1316 * The channel name changed so we must generate COLP update.
1317 * Likely because a call pickup channel masqueraded in.
1318 */
1319 update_connected_line_from_peer(in, c, 1);
1320 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1321 if (o->pending_connected_update) {
1322 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1323 ast_channel_update_connected_line(in, &o->connected, NULL);
1324 }
1325 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1326 update_connected_line_from_peer(in, c, 1);
1327 }
1328 }
1329 if (o->aoc_s_rate_list) {
1330 size_t encoded_size;
1331 struct ast_aoc_encoded *encoded;
1332 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1333 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1334 ast_aoc_destroy_encoded(encoded);
1335 }
1336 }
1337 peer = c;
1338 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1339 ast_copy_flags64(peerflags, o,
1340 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1341 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1342 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1343 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1344 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1345 DIAL_NOFORWARDHTML);
1346 ast_channel_dialcontext_set(c, "");
1347 ast_channel_exten_set(c, "");
1348 }
1349 continue;
1350 }
1351 if (c != winner)
1352 continue;
1353 /* here, o->chan == c == winner */
1354 if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1355 pa->sentringing = 0;
1356 if (!ignore_cc && (f = ast_read(c))) {
1357 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1358 /* This channel is forwarding the call, and is capable of CC, so
1359 * be sure to add the new device interface to the list
1360 */
1361 ast_handle_cc_control_frame(in, c, f->data.ptr);
1362 }
1363 ast_frfree(f);
1364 }
1365
1366 if (o->pending_connected_update) {
1367 /*
1368 * Re-seed the chanlist's connected line information with
1369 * previously acquired connected line info from the incoming
1370 * channel. The previously acquired connected line info could
1371 * have been set through the CONNECTED_LINE dialplan function.
1372 */
1373 o->pending_connected_update = 0;
1374 ast_channel_lock(in);
1375 ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1376 ast_channel_unlock(in);
1377 }
1378
1379 do_forward(o, &num, peerflags, single, caller_entertained, &orig_answer_to,
1380 forced_clid, stored_clid);
1381
1382 if (o->chan) {
1383 ast_free(o->orig_chan_name);
1384 o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1385 if (single
1386 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1387 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1388 update_connected_line_from_peer(in, o->chan, 1);
1389 }
1390 }
1391 continue;
1392 }
1393 f = ast_read(winner);
1394 if (!f) {
1395 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1396 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1397 ast_hangup(c);
1398 c = o->chan = NULL;
1399 ast_clear_flag64(o, DIAL_STILLGOING);
1400 handle_cause(ast_channel_hangupcause(in), &num);
1401 continue;
1402 }
1403 switch (f->frametype) {
1404 case AST_FRAME_CONTROL:
1405 switch (f->subclass.integer) {
1406 case AST_CONTROL_ANSWER:
1407 /* This is our guy if someone answered. */
1408 if (!peer) {
1409 ast_trace(-1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1410 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1411 if (o->orig_chan_name
1412 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1413 /*
1414 * The channel name changed so we must generate COLP update.
1415 * Likely because a call pickup channel masqueraded in.
1416 */
1417 update_connected_line_from_peer(in, c, 1);
1418 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1419 if (o->pending_connected_update) {
1420 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1421 ast_channel_update_connected_line(in, &o->connected, NULL);
1422 }
1423 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1424 update_connected_line_from_peer(in, c, 1);
1425 }
1426 }
1427 if (o->aoc_s_rate_list) {
1428 size_t encoded_size;
1429 struct ast_aoc_encoded *encoded;
1430 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1431 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1432 ast_aoc_destroy_encoded(encoded);
1433 }
1434 }
1435 peer = c;
1436 /* Answer can optionally include a topology */
1437 if (f->subclass.topology) {
1438 /*
1439 * We need to bump the refcount on the topology to prevent it
1440 * from being cleaned up when the frame is cleaned up.
1441 */
1442 config->answer_topology = ao2_bump(f->subclass.topology);
1443 ast_trace(-1, "%s Found topology in frame: %p %p %s\n",
1444 ast_channel_name(peer), f, config->answer_topology,
1445 ast_str_tmp(256, ast_stream_topology_to_str(config->answer_topology, &STR_TMP)));
1446 }
1447
1448 /* Inform everyone else that they've been canceled.
1449 * The dial end event for the peer will be sent out after
1450 * other Dial options have been handled.
1451 */
1452 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1453 ast_copy_flags64(peerflags, o,
1454 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1455 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1456 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1457 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1458 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1459 DIAL_NOFORWARDHTML);
1460 ast_channel_dialcontext_set(c, "");
1461 ast_channel_exten_set(c, "");
1462 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1463 /* Setup early bridge if appropriate */
1464 ast_channel_early_bridge(in, peer);
1465 }
1466 }
1467 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1468 ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1469 ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1470 break;
1471 case AST_CONTROL_BUSY:
1472 ast_verb(3, "%s is busy\n", ast_channel_name(c));
1473 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1474 ast_channel_publish_dial(in, c, NULL, "BUSY");
1475 ast_hangup(c);
1476 c = o->chan = NULL;
1477 ast_clear_flag64(o, DIAL_STILLGOING);
1478 handle_cause(AST_CAUSE_BUSY, &num);
1479 break;
1480 case AST_CONTROL_CONGESTION:
1481 ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1482 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1483 ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1484 ast_hangup(c);
1485 c = o->chan = NULL;
1486 ast_clear_flag64(o, DIAL_STILLGOING);
1487 handle_cause(AST_CAUSE_CONGESTION, &num);
1488 break;
1489 case AST_CONTROL_RINGING:
1490 /* This is a tricky area to get right when using a native
1491 * CC agent. The reason is that we do the best we can to send only a
1492 * single ringing notification to the caller.
1493 *
1494 * Call completion complicates the logic used here. CCNR is typically
1495 * offered during a ringing message. Let's say that party A calls
1496 * parties B, C, and D. B and C do not support CC requests, but D
1497 * does. If we were to receive a ringing notification from B before
1498 * the others, then we would end up sending a ringing message to
1499 * A with no CCNR offer present.
1500 *
1501 * The approach that we have taken is that if we receive a ringing
1502 * response from a party and no CCNR offer is present, we need to
1503 * wait. Specifically, we need to wait until either a) a called party
1504 * offers CCNR in its ringing response or b) all called parties have
1505 * responded in some way to our call and none offers CCNR.
1506 *
1507 * The drawback to this is that if one of the parties has a delayed
1508 * response or, god forbid, one just plain doesn't respond to our
1509 * outgoing call, then this will result in a significant delay between
1510 * when the caller places the call and hears ringback.
1511 *
1512 * Note also that if CC is disabled for this call, then it is perfectly
1513 * fine for ringing frames to get sent through.
1514 */
1515 ++num_ringing;
1516 *to_progress = -1;
1517 orig_progress_to = -1;
1518 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1519 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1520 /* Setup early media if appropriate */
1521 if (single && !caller_entertained
1522 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1523 ast_channel_early_bridge(in, c);
1524 }
1525 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1526 ast_indicate(in, AST_CONTROL_RINGING);
1527 pa->sentringing++;
1528 }
1529 if (!sent_ring) {
1530 struct timeval now, then;
1531 int64_t diff;
1532
1533 now = ast_tvnow();
1534
1535 ast_channel_lock(in);
1536 ast_channel_stage_snapshot(in);
1537
1538 then = ast_channel_creationtime(c);
1539 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1540 set_duration_var(in, "RINGTIME", diff);
1541
1542 ast_channel_stage_snapshot_done(in);
1543 ast_channel_unlock(in);
1544 sent_ring = 1;
1545 }
1546 }
1547 ast_channel_publish_dial(in, c, NULL, "RINGING");
1548 break;
1549 case AST_CONTROL_PROGRESS:
1550 ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1551 /* Setup early media if appropriate */
1552 if (single && !caller_entertained
1553 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1554 ast_channel_early_bridge(in, c);
1555 }
1556 if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1557 if (single || (!single && !pa->sentringing)) {
1558 ast_indicate(in, AST_CONTROL_PROGRESS);
1559 }
1560 }
1561 *to_progress = -1;
1562 orig_progress_to = -1;
1563 if (!sent_progress) {
1564 struct timeval now, then;
1565 int64_t diff;
1566
1567 now = ast_tvnow();
1568
1569 ast_channel_lock(in);
1570 ast_channel_stage_snapshot(in);
1571
1572 then = ast_channel_creationtime(c);
1573 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1574 set_duration_var(in, "PROGRESSTIME", diff);
1575
1576 ast_channel_stage_snapshot_done(in);
1577 ast_channel_unlock(in);
1578 sent_progress = 1;
1579
1580 if (!ast_strlen_zero(mf_progress)) {
1581 ast_verb(3,
1582 "Sending MF '%s' to %s as result of "
1583 "receiving a PROGRESS message.\n",
1584 mf_progress, hearpulsing ? "parties" : "called party");
1585 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1586 (hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
1587 }
1588 if (!ast_strlen_zero(sf_progress)) {
1589 ast_verb(3,
1590 "Sending SF '%s' to %s as result of "
1591 "receiving a PROGRESS message.\n",
1592 sf_progress, (hearpulsing ? "parties" : "called party"));
1593 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1594 (hearpulsing ? in : NULL), sf_progress, 0, 0);
1595 }
1596 if (!ast_strlen_zero(dtmf_progress)) {
1597 ast_verb(3,
1598 "Sending DTMF '%s' to the called party as result of "
1599 "receiving a PROGRESS message.\n",
1600 dtmf_progress);
1601 res |= ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1602 }
1603 if (res) {
1604 ast_log(LOG_WARNING, "Called channel %s hung up post-progress before all digits could be sent\n", ast_channel_name(c));
1605 if (ast_channel_state(c) == AST_STATE_UP) {
1606 /* The called channel answered while we were sending it digits, so the answer never got processed by app_dial.
1607 * The channel is dying now, but better to answer late than never? */
1608 ast_debug(1, "Channel %s answered while we were sending it digits, answering %s retroactively\n", ast_channel_name(c), ast_channel_name(in));
1609 /* Indicate answer supervision to the caller before we exit.
1610 * We're not going to bridge, but this way at least the CDRs are correct, etc. */
1611 ast_raw_answer(in);
1612 strcpy(pa->status, "ANSWER");
1613 } else {
1614 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1615 }
1616 goto wait_over;
1617 }
1618 }
1619 ast_channel_publish_dial(in, c, NULL, "PROGRESS");
1620 break;
1621 case AST_CONTROL_WINK:
1622 ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1623 if (!sent_wink) {
1624 sent_wink = 1;
1625 if (!ast_strlen_zero(mf_wink)) {
1626 ast_verb(3,
1627 "Sending MF '%s' to %s as result of "
1628 "receiving a WINK message.\n",
1629 mf_wink, (hearpulsing ? "parties" : "called party"));
1630 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1631 (hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
1632 }
1633 if (!ast_strlen_zero(sf_wink)) {
1634 ast_verb(3,
1635 "Sending SF '%s' to %s as result of "
1636 "receiving a WINK message.\n",
1637 sf_wink, (hearpulsing ? "parties" : "called party"));
1638 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1639 (hearpulsing ? in : NULL), sf_wink, 0, 0);
1640 }
1641 if (res) {
1642 ast_log(LOG_WARNING, "Called channel %s hung up post-wink before all digits could be sent\n", ast_channel_name(c));
1643 if (ast_channel_state(c) == AST_STATE_UP) {
1644 /* Same as in AST_CONTROL_PROGRESS */
1645 ast_debug(1, "Channel %s answered while we were sending it digits, answering %s retroactively\n", ast_channel_name(c), ast_channel_name(in));
1646 ast_raw_answer(in);
1647 strcpy(pa->status, "ANSWER");
1648 } else {
1649 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1650 }
1651 goto wait_over;
1652 }
1653 }
1654 ast_indicate(in, AST_CONTROL_WINK);
1655 break;
1656 case AST_CONTROL_VIDUPDATE:
1657 case AST_CONTROL_SRCUPDATE:
1658 case AST_CONTROL_SRCCHANGE:
1659 if (!single || caller_entertained) {
1660 break;
1661 }
1662 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1663 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1664 ast_indicate(in, f->subclass.integer);
1665 break;
1666 case AST_CONTROL_CONNECTED_LINE:
1667 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1668 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1669 break;
1670 }
1671 if (!single) {
1672 struct ast_party_connected_line connected;
1673
1674 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1675 ast_channel_name(c), ast_channel_name(in));
1676 ast_party_connected_line_set_init(&connected, &o->connected);
1677 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1678 ast_party_connected_line_set(&o->connected, &connected, NULL);
1679 ast_party_connected_line_free(&connected);
1680 o->pending_connected_update = 1;
1681 break;
1682 }
1683 if (ast_channel_connected_line_sub(c, in, f, 1)) {
1684 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1685 }
1686 break;
1687 case AST_CONTROL_AOC:
1688 {
1689 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1690 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1691 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1692 o->aoc_s_rate_list = decoded;
1693 } else {
1694 ast_aoc_destroy_decoded(decoded);
1695 }
1696 }
1697 break;
1698 case AST_CONTROL_REDIRECTING:
1699 if (!single) {
1700 /*
1701 * Redirecting updates to the caller make sense only on single
1702 * calls.
1703 */
1704 break;
1705 }
1706 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1707 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1708 break;
1709 }
1710 ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1711 ast_channel_name(c), ast_channel_name(in));
1712 if (ast_channel_redirecting_sub(c, in, f, 1)) {
1713 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1714 }
1715 pa->sentringing = 0;
1716 break;
1717 case AST_CONTROL_PROCEEDING:
1718 ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1719 if (single && !caller_entertained
1720 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1721 ast_channel_early_bridge(in, c);
1722 }
1723 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1724 ast_indicate(in, AST_CONTROL_PROCEEDING);
1725 ast_channel_publish_dial(in, c, NULL, "PROCEEDING");
1726 break;
1727 case AST_CONTROL_HOLD:
1728 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1729 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1730 ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1731 break;
1732 case AST_CONTROL_UNHOLD:
1733 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1734 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1735 ast_indicate(in, AST_CONTROL_UNHOLD);
1736 break;
1737 case AST_CONTROL_OFFHOOK:
1738 case AST_CONTROL_FLASH:
1739 /* Ignore going off hook and flash */
1740 break;
1741 case AST_CONTROL_CC:
1742 if (!ignore_cc) {
1743 ast_handle_cc_control_frame(in, c, f->data.ptr);
1744 cc_frame_received = 1;
1745 }
1746 break;
1747 case AST_CONTROL_PVT_CAUSE_CODE:
1748 ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1749 break;
1750 case AST_CONTROL_PLAYBACK_BEGIN:
1751 if (!f->data.ptr) {
1752 ast_log(LOG_WARNING, "Got playback begin directive without filename on %s\n", ast_channel_name(c));
1753 } else {
1754 const char *filename = f->data.ptr;
1755 ast_verb(3, "Playing audio file %s on %s\n", filename, ast_channel_name(in));
1756 ast_streamfile(in, filename, ast_channel_language(in));
1757 }
1758 break;
1759 case -1:
1760 if (single && !caller_entertained) {
1761 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1762 ast_indicate(in, -1);
1763 pa->sentringing = 0;
1764 }
1765 break;
1766 default:
1767 ast_debug(1, "Dunno what to do with control type %d on %s\n", f->subclass.integer, ast_channel_name(in));
1768 break;
1769 }
1770 break;
1771 case AST_FRAME_VIDEO:
1772 case AST_FRAME_VOICE:
1773 case AST_FRAME_IMAGE:
1774 case AST_FRAME_DTMF_BEGIN:
1775 case AST_FRAME_DTMF_END:
1776 if (caller_entertained) {
1777 break;
1778 }
1779 *to_progress = -1;
1780 orig_progress_to = -1;
1781 /* Fall through */
1782 case AST_FRAME_TEXT:
1783 if (single && ast_write(in, f)) {
1784 ast_log(LOG_WARNING, "Unable to write frametype %u on %s\n",
1785 f->frametype, ast_channel_name(in));
1786 }
1787 break;
1788 case AST_FRAME_HTML:
1789 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1790 && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1791 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(in));
1792 }
1793 break;
1794 default:
1795 break;
1796 }
1797 ast_frfree(f);
1798 } /* end for */
1799 if (winner == in) {
1800 struct ast_frame *f = ast_read(in);
1801 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1802 /* Got hung up */
1803 *to_answer = -1;
1804 strcpy(pa->status, "CANCEL");
1805 pa->canceled = 1;
1806 publish_dial_end_event(in, out_chans, NULL, pa->status);
1807 if (f) {
1808 if (f->data.uint32) {
1809 ast_channel_hangupcause_set(in, f->data.uint32);
1810 }
1811 ast_frfree(f);
1812 }
1813 if (is_cc_recall) {
1814 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1815 }
1816 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller hung up\n", ast_channel_name(in));
1817 }
1818
1819 /* now f is guaranteed non-NULL */
1820 if (f->frametype == AST_FRAME_DTMF) {
1821 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1822 const char *context;
1823 ast_channel_lock(in);
1824 if ((context = pbx_builtin_getvar_helper(in, "EXITCONTEXT"))) {
1825 context = ast_strdupa(context);
1826 }
1827 ast_channel_unlock(in);
1828 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1829 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1830 *to_answer = 0;
1831 *result = f->subclass.integer;
1832 strcpy(pa->status, "CANCEL");
1833 pa->canceled = 1;
1834 publish_dial_end_event(in, out_chans, NULL, pa->status);
1835 ast_frfree(f);
1836 if (is_cc_recall) {
1837 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1838 }
1839 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller pressed %c to end call\n",
1840 ast_channel_name(in), f->subclass.integer);
1841 }
1842 }
1843
1844 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1845 detect_disconnect(in, f->subclass.integer, &featurecode)) {
1846 ast_verb(3, "User requested call disconnect.\n");
1847 *to_answer = 0;
1848 strcpy(pa->status, "CANCEL");
1849 pa->canceled = 1;
1850 publish_dial_end_event(in, out_chans, NULL, pa->status);
1851 ast_frfree(f);
1852 if (is_cc_recall) {
1853 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1854 }
1855 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller requested disconnect\n",
1856 ast_channel_name(in));
1857 }
1858 }
1859
1860 /* Send the frame from the in channel to all outgoing channels. */
1861 AST_LIST_TRAVERSE(out_chans, o, node) {
1862 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1863 /* This outgoing channel has died so don't send the frame to it. */
1864 continue;
1865 }
1866 switch (f->frametype) {
1867 case AST_FRAME_HTML:
1868 /* Forward HTML stuff */
1869 if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1870 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1871 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(o->chan));
1872 }
1873 break;
1874 case AST_FRAME_VIDEO:
1875 case AST_FRAME_VOICE:
1876 case AST_FRAME_IMAGE:
1877 if (!single || caller_entertained) {
1878 /*
1879 * We are calling multiple parties or caller is being
1880 * entertained and has thus not been made compatible.
1881 * No need to check any other called parties.
1882 */
1883 goto skip_frame;
1884 }
1885 /* Fall through */
1886 case AST_FRAME_TEXT:
1887 case AST_FRAME_DTMF_BEGIN:
1888 case AST_FRAME_DTMF_END:
1889 if (ast_write(o->chan, f)) {
1890 ast_log(LOG_WARNING, "Unable to forward frametype %u on %s\n",
1891 f->frametype, ast_channel_name(o->chan));
1892 }
1893 break;
1894 case AST_FRAME_CONTROL:
1895 switch (f->subclass.integer) {
1896 case AST_CONTROL_HOLD:
1897 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1898 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1899 break;
1900 case AST_CONTROL_UNHOLD:
1901 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1902 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1903 break;
1904 case AST_CONTROL_FLASH:
1905 ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
1906 ast_indicate(o->chan, AST_CONTROL_FLASH);
1907 break;
1908 case AST_CONTROL_VIDUPDATE:
1909 case AST_CONTROL_SRCUPDATE:
1910 case AST_CONTROL_SRCCHANGE:
1911 if (!single || caller_entertained) {
1912 /*
1913 * We are calling multiple parties or caller is being
1914 * entertained and has thus not been made compatible.
1915 * No need to check any other called parties.
1916 */
1917 goto skip_frame;
1918 }
1919 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1920 ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1921 ast_indicate(o->chan, f->subclass.integer);
1922 break;
1923 case AST_CONTROL_CONNECTED_LINE:
1924 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1925 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
1926 break;
1927 }
1928 if (ast_channel_connected_line_sub(in, o->chan, f, 1)) {
1929 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1930 }
1931 break;
1932 case AST_CONTROL_REDIRECTING:
1933 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1934 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
1935 break;
1936 }
1937 if (ast_channel_redirecting_sub(in, o->chan, f, 1)) {
1938 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1939 }
1940 break;
1941 default:
1942 /* We are not going to do anything with this frame. */
1943 goto skip_frame;
1944 }
1945 break;
1946 default:
1947 /* We are not going to do anything with this frame. */
1948 goto skip_frame;
1949 }
1950 }
1951skip_frame:;
1952 ast_frfree(f);
1953 }
1954 }
1955
1956wait_over:
1957 if (!*to_answer || ast_check_hangup(in)) {
1958 if (orig_answer_to != -1) {
1959 ast_verb(3, "Nobody picked up in %d ms\n", orig_answer_to);
1960 } else {
1961 ast_verb(3, "Call terminated without answer\n");
1962 }
1963 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1964 } else if (!*to_progress) {
1965 ast_verb(3, "No early media received in %d ms\n", orig_progress_to);
1966 publish_dial_end_event(in, out_chans, NULL, "CHANUNAVAIL");
1967 strcpy(pa->status, "CHANUNAVAIL");
1968 *to_answer = 0; /* Reset to prevent hangup */
1969 }
1970
1971 if (is_cc_recall) {
1972 ast_cc_completed(in, "Recall completed!");
1973 }
1974 SCOPE_EXIT_RTN_VALUE(peer, "%s: %s%s\n", ast_channel_name(in),
1975 peer ? "Answered by " : "No answer", peer ? ast_channel_name(peer) : "");
1976}
1977
1978static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1979{
1980 char disconnect_code[AST_FEATURE_MAX_LEN];
1981 int res;
1982
1983 ast_str_append(featurecode, 1, "%c", code);
1984
1985 res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1986 if (res) {
1987 ast_str_reset(*featurecode);
1988 return 0;
1989 }
1990
1991 if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1992 /* Could be a partial match, anyway */
1993 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1994 ast_str_reset(*featurecode);
1995 }
1996 return 0;
1997 }
1998
1999 if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
2000 ast_str_reset(*featurecode);
2001 return 0;
2002 }
2003
2004 return 1;
2005}
2006
2007/* returns true if there is a valid privacy reply */
2008static int valid_priv_reply(struct ast_flags64 *opts, int res)
2009{
2010 if (res < '1')
2011 return 0;
2012 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
2013 return 1;
2014 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
2015 return 1;
2016 return 0;
2017}
2018
2019static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
2020 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
2021{
2022
2023 int res2;
2024 int loopcount = 0;
2025
2026 /* Get the user's intro, store it in priv-callerintros/$CID,
2027 unless it is already there-- this should be done before the
2028 call is actually dialed */
2029
2030 /* all ring indications and moh for the caller has been halted as soon as the
2031 target extension was picked up. We are going to have to kill some
2032 time and make the caller believe the peer hasn't picked up yet */
2033
2034 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2035 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2036 ast_indicate(chan, -1);
2037 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2038 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2039 ast_channel_musicclass_set(chan, original_moh);
2040 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2041 ast_indicate(chan, AST_CONTROL_RINGING);
2042 pa->sentringing++;
2043 }
2044
2045 /* Start autoservice on the other chan ?? */
2046 res2 = ast_autoservice_start(chan);
2047 /* Now Stream the File */
2048 for (loopcount = 0; loopcount < 3; loopcount++) {
2049 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
2050 break;
2051 if (!res2) /* on timeout, play the message again */
2052 res2 = ast_play_and_wait(peer, "priv-callpending");
2053 if (!valid_priv_reply(opts, res2))
2054 res2 = 0;
2055 /* priv-callpending script:
2056 "I have a caller waiting, who introduces themselves as:"
2057 */
2058 if (!res2)
2059 res2 = ast_play_and_wait(peer, pa->privintro);
2060 if (!valid_priv_reply(opts, res2))
2061 res2 = 0;
2062 /* now get input from the called party, as to their choice */
2063 if (!res2) {
2064 /* XXX can we have both, or they are mutually exclusive ? */
2065 if (ast_test_flag64(opts, OPT_PRIVACY))
2066 res2 = ast_play_and_wait(peer, "priv-callee-options");
2067 if (ast_test_flag64(opts, OPT_SCREENING))
2068 res2 = ast_play_and_wait(peer, "screen-callee-options");
2069 }
2070
2071 /*! \page DialPrivacy Dial Privacy scripts
2072 * \par priv-callee-options script:
2073 * \li Dial 1 if you wish this caller to reach you directly in the future,
2074 * and immediately connect to their incoming call.
2075 * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
2076 * \li Dial 3 to send this caller to the torture menus, now and forevermore.
2077 * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
2078 * \li Dial 5 to allow this caller to come straight thru to you in the future,
2079 * but right now, just this once, send them to voicemail.
2080 *
2081 * \par screen-callee-options script:
2082 * \li Dial 1 if you wish to immediately connect to the incoming call
2083 * \li Dial 2 if you wish to send this caller to voicemail.
2084 * \li Dial 3 to send this caller to the torture menus.
2085 * \li Dial 4 to send this caller to a simple "go away" menu.
2086 */
2087 if (valid_priv_reply(opts, res2))
2088 break;
2089 /* invalid option */
2090 res2 = ast_play_and_wait(peer, "vm-sorry");
2091 }
2092
2093 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
2094 ast_moh_stop(chan);
2095 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2096 ast_indicate(chan, -1);
2097 pa->sentringing = 0;
2098 }
2099 ast_autoservice_stop(chan);
2100 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
2101 /* map keypresses to various things, the index is res2 - '1' */
2102 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
2103 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
2104 int i = res2 - '1';
2105 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
2106 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
2107 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
2108 }
2109 switch (res2) {
2110 case '1':
2111 break;
2112 case '2':
2113 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2114 break;
2115 case '3':
2116 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2117 break;
2118 case '4':
2119 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2120 break;
2121 case '5':
2122 if (ast_test_flag64(opts, OPT_PRIVACY)) {
2123 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2124 break;
2125 }
2126 /* if not privacy, then 5 is the same as "default" case */
2127 default: /* bad input or -1 if failure to start autoservice */
2128 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
2129 /* well, there seems basically two choices. Just patch the caller thru immediately,
2130 or,... put 'em thru to voicemail. */
2131 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
2132 ast_verb(3, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
2133 /* XXX should we set status to DENY ? */
2134 /* XXX what about the privacy flags ? */
2135 break;
2136 }
2137
2138 if (res2 == '1') { /* the only case where we actually connect */
2139 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
2140 just clog things up, and it's not useful information, not being tied to a CID */
2141 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
2142 ast_filedelete(pa->privintro, NULL);
2143 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2144 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2145 else
2146 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2147 }
2148 return 0; /* the good exit path */
2149 } else {
2150 return -1;
2151 }
2152}
2153
2154/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
2155static int setup_privacy_args(struct privacy_args *pa,
2156 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
2157{
2158 char callerid[60];
2159 int res;
2160 char *l;
2161
2162 if (ast_channel_caller(chan)->id.number.valid
2163 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2164 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2165 ast_shrink_phone_number(l);
2166 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
2167 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
2168 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
2169 } else {
2170 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
2171 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2172 }
2173 } else {
2174 char *tnam, *tn2;
2175
2176 tnam = ast_strdupa(ast_channel_name(chan));
2177 /* clean the channel name so slashes don't try to end up in disk file name */
2178 for (tn2 = tnam; *tn2; tn2++) {
2179 if (*tn2 == '/') /* any other chars to be afraid of? */
2180 *tn2 = '=';
2181 }
2182 ast_verb(3, "Privacy-- callerid is empty\n");
2183
2184 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
2185 l = callerid;
2186 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2187 }
2188
2189 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
2190
2191 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
2192 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
2193 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
2194 pa->privdb_val = AST_PRIVACY_ALLOW;
2195 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
2196 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
2197 }
2198
2199 if (pa->privdb_val == AST_PRIVACY_DENY) {
2200 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
2201 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2202 return 0;
2203 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
2204 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2205 return 0; /* Is this right? */
2206 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
2207 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2208 return 0; /* is this right??? */
2209 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
2210 /* Get the user's intro, store it in priv-callerintros/$CID,
2211 unless it is already there-- this should be done before the
2212 call is actually dialed */
2213
2214 /* make sure the priv-callerintros dir actually exists */
2215 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
2216 if ((res = ast_mkdir(pa->privintro, 0755))) {
2217 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
2218 return -1;
2219 }
2220
2221 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
2222 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
2223 /* the DELUX version of this code would allow this caller the
2224 option to hear and retape their previously recorded intro.
2225 */
2226 } else {
2227 int duration; /* for feedback from play_and_wait */
2228 /* the file doesn't exist yet. Let the caller submit his
2229 vocal intro for posterity */
2230 /* priv-recordintro script:
2231 "At the tone, please say your name:"
2232 */
2233 int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
2234 ast_answer(chan);
2235 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 */
2236 /* don't think we'll need a lock removed, we took care of
2237 conflicts by naming the pa.privintro file */
2238 if (res == -1) {
2239 /* Delete the file regardless since they hung up during recording */
2240 ast_filedelete(pa->privintro, NULL);
2241 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2242 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2243 else
2244 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2245 return -1;
2246 }
2247 if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
2248 ast_waitstream(chan, "");
2249 }
2250 }
2251 return 1; /* success */
2252}
2253
2254static void end_bridge_callback(void *data)
2255{
2256 struct ast_channel *chan = data;
2257
2258 ast_channel_lock(chan);
2259 ast_channel_stage_snapshot(chan);
2260 set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
2261 set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
2262 ast_channel_stage_snapshot_done(chan);
2263 ast_channel_unlock(chan);
2264}
2265
2266static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2267 bconfig->end_bridge_callback_data = originator;
2268}
2269
2270static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2271{
2272 struct ast_tone_zone_sound *ts = NULL;
2273 int res;
2274 const char *str = data;
2275
2276 if (ast_strlen_zero(str)) {
2277 ast_debug(1,"Nothing to play\n");
2278 return -1;
2279 }
2280
2281 ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2282
2283 if (ts && ts->data[0]) {
2284 res = ast_playtones_start(chan, 0, ts->data, 0);
2285 } else {
2286 res = -1;
2287 }
2288
2289 if (ts) {
2290 ts = ast_tone_zone_sound_unref(ts);
2291 }
2292
2293 if (res) {
2294 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2295 }
2296
2297 return res;
2298}
2299
2300/*!
2301 * \internal
2302 * \brief Setup the after bridge goto location on the peer.
2303 * \since 12.0.0
2304 *
2305 * \param chan Calling channel for bridge.
2306 * \param peer Peer channel for bridge.
2307 * \param opts Dialing option flags.
2308 * \param opt_args Dialing option argument strings.
2309 */
2310static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
2311{
2312 const char *context;
2313 const char *extension;
2314 int priority;
2315
2316 if (ast_test_flag64(opts, OPT_PEER_H)) {
2317 ast_channel_lock(chan);
2318 context = ast_strdupa(ast_channel_context(chan));
2319 ast_channel_unlock(chan);
2320 ast_bridge_set_after_h(peer, context);
2321 } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2322 ast_channel_lock(chan);
2323 context = ast_strdupa(ast_channel_context(chan));
2324 extension = ast_strdupa(ast_channel_exten(chan));
2325 priority = ast_channel_priority(chan);
2326 ast_channel_unlock(chan);
2327 ast_bridge_set_after_go_on(peer, context, extension, priority,
2328 opt_args[OPT_ARG_CALLEE_GO_ON]);
2329 }
2330}
2331
2332static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2333{
2334 int res = -1; /* default: error */
2335 char *rest, *cur; /* scan the list of destinations */
2336 struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2337 struct chanlist *outgoing;
2338 struct chanlist *tmp;
2339 struct ast_channel *peer = NULL;
2340 int to_answer, to_progress; /* timeouts */
2341 struct cause_args num = { chan, 0, 0, 0 };
2342 int cause, hanguptreecause = -1;
2343
2344 struct ast_bridge_config config = { { 0, } };
2345 struct timeval calldurationlimit = { 0, };
2346 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
2347 char *mf_progress = NULL, *mf_wink = NULL;
2348 char *sf_progress = NULL, *sf_wink = NULL;
2349 struct privacy_args pa = {
2350 .sentringing = 0,
2351 .privdb_val = 0,
2352 .status = "INVALIDARGS",
2353 .canceled = 0,
2354 };
2355 int sentringing = 0, moh = 0;
2356 const char *outbound_group = NULL;
2357 int result = 0;
2358 char *parse;
2359 int opermode = 0;
2360 int delprivintro = 0;
2361 AST_DECLARE_APP_ARGS(args,
2362 AST_APP_ARG(peers);
2363 AST_APP_ARG(timeout);
2364 AST_APP_ARG(options);
2365 AST_APP_ARG(url);
2366 );
2367 struct ast_flags64 opts = { 0, };
2368 char *opt_args[OPT_ARG_ARRAY_SIZE];
2369 int fulldial = 0, num_dialed = 0;
2370 int ignore_cc = 0;
2371 char device_name[AST_CHANNEL_NAME];
2372 char forced_clid_name[AST_MAX_EXTENSION];
2373 char stored_clid_name[AST_MAX_EXTENSION];
2374 int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2375 /*!
2376 * \brief Forced CallerID party information to send.
2377 * \note This will not have any malloced strings so do not free it.
2378 */
2379 struct ast_party_id forced_clid;
2380 /*!
2381 * \brief Stored CallerID information if needed.
2382 *
2383 * \note If OPT_ORIGINAL_CLID set then this is the o option
2384 * CallerID. Otherwise it is the dialplan extension and hint
2385 * name.
2386 *
2387 * \note This will not have any malloced strings so do not free it.
2388 */
2389 struct ast_party_id stored_clid;
2390 /*!
2391 * \brief CallerID party information to store.
2392 * \note This will not have any malloced strings so do not free it.
2393 */
2394 struct ast_party_caller caller;
2395 int max_forwards;
2396 struct ast_datastore *topology_ds = NULL;
2397 SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
2398
2399 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2400 ast_channel_lock(chan);
2401 ast_channel_stage_snapshot(chan);
2402 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2403 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2404 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2405 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2406 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME_MS", "");
2407 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2408 pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
2409 pbx_builtin_setvar_helper(chan, "RINGTIME", "");
2410 pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
2411 pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
2412 pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
2413 ast_channel_stage_snapshot_done(chan);
2414 max_forwards = ast_max_forwards_get(chan);
2415 ast_channel_unlock(chan);
2416
2417 if (max_forwards <= 0) {
2418 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2419 ast_channel_name(chan));
2420 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2421 SCOPE_EXIT_RTN_VALUE(-1, "%s: Max forwards exceeded\n", ast_channel_name(chan));
2422 }
2423
2424 if (ast_check_hangup_locked(chan)) {
2425 /*
2426 * Caller hung up before we could dial. If dial is executed
2427 * within an AGI then the AGI has likely eaten all queued
2428 * frames before executing the dial in DeadAGI mode. With
2429 * the caller hung up and no pending frames from the caller's
2430 * read queue, dial would not know that the call has hung up
2431 * until a called channel answers. It is rather annoying to
2432 * whoever just answered the non-existent call.
2433 *
2434 * Dial should not continue execution in DeadAGI mode, hangup
2435 * handlers, or the h exten.
2436 */
2437 ast_verb(3, "Caller hung up before dial.\n");
2438 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2439 SCOPE_EXIT_RTN_VALUE(-1, "%s: Caller hung up before dial\n", ast_channel_name(chan));
2440 }
2441
2442 parse = ast_strdupa(data ?: "");
2443
2444 AST_STANDARD_APP_ARGS(args, parse);
2445
2446 if (!ast_strlen_zero(args.options) &&
2447 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2448 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2449 goto done;
2450 }
2451
2452 if (ast_cc_call_init(chan, &ignore_cc)) {
2453 goto done;
2454 }
2455
2456 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2457 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2458
2459 if (delprivintro < 0 || delprivintro > 1) {
2460 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2461 delprivintro = 0;
2462 }
2463 }
2464
2465 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2466 opt_args[OPT_ARG_RINGBACK] = NULL;
2467 }
2468
2469 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2470 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2471 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2472 }
2473
2474 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2475 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2476 if (!calldurationlimit.tv_sec) {
2477 ast_log(LOG_WARNING, "Dial does not accept S(%s)\n", opt_args[OPT_ARG_DURATION_STOP]);
2478 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2479 goto done;
2480 }
2481 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2482 }
2483
2484 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2485 sf_wink = opt_args[OPT_ARG_SENDDTMF];
2486 dtmfcalled = strsep(&sf_wink, ":");
2487 dtmfcalling = strsep(&sf_wink, ":");
2488 dtmf_progress = strsep(&sf_wink, ":");
2489 mf_progress = strsep(&sf_wink, ":");
2490 mf_wink = strsep(&sf_wink, ":");
2491 sf_progress = strsep(&sf_wink, ":");
2492 }
2493
2494 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2495 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2496 goto done;
2497 }
2498
2499 /* Setup the forced CallerID information to send if used. */
2500 ast_party_id_init(&forced_clid);
2501 force_forwards_only = 0;
2502 if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2503 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2504 ast_channel_lock(chan);
2505 forced_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2506 ast_channel_unlock(chan);
2507 forced_clid_name[0] = '\0';
2508 forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2509 sizeof(forced_clid_name), chan);
2510 force_forwards_only = 1;
2511 } else {
2512 /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2513 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2514 &forced_clid.number.str);
2515 }
2516 if (!ast_strlen_zero(forced_clid.name.str)) {
2517 forced_clid.name.valid = 1;
2518 }
2519 if (!ast_strlen_zero(forced_clid.number.str)) {
2520 forced_clid.number.valid = 1;
2521 }
2522 }
2523 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2524 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2525 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2526 }
2527 forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2528 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2529 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2530 int pres;
2531
2532 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2533 if (0 <= pres) {
2534 forced_clid.number.presentation = pres;
2535 }
2536 }
2537
2538 /* Setup the stored CallerID information if needed. */
2539 ast_party_id_init(&stored_clid);
2540 if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2541 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2542 ast_channel_lock(chan);
2543 ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2544 if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2545 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2546 }
2547 if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2548 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2549 }
2550 if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2551 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2552 }
2553 if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2554 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2555 }
2556 ast_channel_unlock(chan);
2557 } else {
2558 /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2559 ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2560 &stored_clid.number.str);
2561 if (!ast_strlen_zero(stored_clid.name.str)) {
2562 stored_clid.name.valid = 1;
2563 }
2564 if (!ast_strlen_zero(stored_clid.number.str)) {
2565 stored_clid.number.valid = 1;
2566 }
2567 }
2568 } else {
2569 /*
2570 * In case the new channel has no preset CallerID number by the
2571 * channel driver, setup the dialplan extension and hint name.
2572 */
2573 stored_clid_name[0] = '\0';
2574 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2575 sizeof(stored_clid_name), chan);
2576 if (ast_strlen_zero(stored_clid.name.str)) {
2577 stored_clid.name.str = NULL;
2578 } else {
2579 stored_clid.name.valid = 1;
2580 }
2581 ast_channel_lock(chan);
2582 stored_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2583 stored_clid.number.valid = 1;
2584 ast_channel_unlock(chan);
2585 }
2586
2587 if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2588 ast_cdr_reset(ast_channel_name(chan), 0);
2589 }
2590 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2591 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2592
2593 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2594 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2595 if (res <= 0)
2596 goto out;
2597 res = -1; /* reset default */
2598 }
2599
2600 if (continue_exec)
2601 *continue_exec = 0;
2602
2603 /* If a channel group has been specified, get it for use when we create peer channels */
2604
2605 ast_channel_lock(chan);
2606 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2607 outbound_group = ast_strdupa(outbound_group);
2608 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2609 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2610 outbound_group = ast_strdupa(outbound_group);
2611 }
2612 ast_channel_unlock(chan);
2613
2614 /* Set per dial instance flags. These flags are also passed back to RetryDial. */
2615 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2616 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2617 | OPT_ANNOUNCE | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2618
2619 /* PREDIAL: Run gosub on the caller's channel */
2620 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2621 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2622 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2623 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2624 }
2625
2626 /* loop through the list of dial destinations */
2627 rest = args.peers;
2628 while ((cur = strsep(&rest, "&"))) {
2629 struct ast_channel *tc; /* channel for this destination */
2630 char *number;
2631 char *tech;
2632 int i;
2633 size_t tech_len;
2634 size_t number_len;
2635 struct ast_stream_topology *topology;
2636 struct ast_stream *stream;
2637
2638 cur = ast_strip(cur);
2639 if (ast_strlen_zero(cur)) {
2640 /* No tech/resource in this position. */
2641 continue;
2642 }
2643
2644 /* Get a technology/resource pair */
2645 number = cur;
2646 tech = strsep(&number, "/");
2647
2648 num_dialed++;
2649 if (ast_strlen_zero(number)) {
2650 ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2651 goto out;
2652 }
2653
2654 tech_len = strlen(tech) + 1;
2655 number_len = strlen(number) + 1;
2656 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2657 if (!tmp) {
2658 goto out;
2659 }
2660
2661 /* Save tech, number, and interface. */
2662 cur = tmp->stuff;
2663 strcpy(cur, tech);
2664 tmp->tech = cur;
2665 cur += tech_len;
2666 strcpy(cur, tech);
2667 cur[tech_len - 1] = '/';
2668 tmp->interface = cur;
2669 cur += tech_len;
2670 strcpy(cur, number);
2671 tmp->number = cur;
2672
2673 if (opts.flags) {
2674 /* Set per outgoing call leg options. */
2675 ast_copy_flags64(tmp, &opts,
2676 OPT_CANCEL_ELSEWHERE |
2677 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2678 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2679 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2680 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2681 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2682 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2683 OPT_RING_WITH_EARLY_MEDIA);
2684 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2685 }
2686
2687 /* Request the peer */
2688
2689 ast_channel_lock(chan);
2690 /*
2691 * Seed the chanlist's connected line information with previously
2692 * acquired connected line info from the incoming channel. The
2693 * previously acquired connected line info could have been set
2694 * through the CONNECTED_LINE dialplan function.
2695 */
2696 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2697
2698 if (ast_test_flag64(&opts, OPT_TOPOLOGY_PRESERVE)) {
2699 topology_ds = ast_channel_datastore_find(chan, &topology_ds_info, NULL);
2700
2701 if (!topology_ds && (topology_ds = ast_datastore_alloc(&topology_ds_info, NULL))) {
2702 topology_ds->data = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2703 ast_channel_datastore_add(chan, topology_ds);
2704 }
2705 }
2706
2707 if (topology_ds) {
2708 ao2_ref(topology_ds->data, +1);
2709 topology = topology_ds->data;
2710 } else {
2711 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2712 }
2713
2714 ast_channel_unlock(chan);
2715
2716 for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
2717 stream = ast_stream_topology_get_stream(topology, i);
2718 /* For both recvonly and sendonly the stream state reflects our state, that is we
2719 * are receiving only and we are sending only. Since we are requesting a
2720 * channel for the peer, we need to swap this to reflect what we will be doing.
2721 * That is, if we are receiving from Alice then we want to be sending to Bob,
2722 * so swap recvonly to sendonly and vice versa.
2723 */
2724 if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
2725 ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
2726 } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
2727 ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
2728 }
2729 }
2730
2731 tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
2732
2733 ast_stream_topology_free(topology);
2734
2735 if (!tc) {
2736 /* If we can't, just go on to the next call */
2737 /* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
2738 ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
2739 tmp->tech, cause, ast_cause2str(cause));
2740 handle_cause(cause, &num);
2741 if (!rest) {
2742 /* we are on the last destination */
2743 ast_channel_hangupcause_set(chan, cause);
2744 }
2745 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2746 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2747 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2748 }
2749 }
2750 chanlist_free(tmp);
2751 continue;
2752 }
2753
2754 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2755 if (!ignore_cc) {
2756 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2757 }
2758
2759 ast_channel_lock_both(tc, chan);
2760 ast_channel_stage_snapshot(tc);
2761
2762 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2763
2764 /* Setup outgoing SDP to match incoming one */
2765 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2766 /* We are on the only destination. */
2767 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2768 }
2769
2770 /* Inherit specially named variables from parent channel */
2771 ast_channel_inherit_variables(chan, tc);
2772 ast_channel_datastore_inherit(chan, tc);
2773 ast_max_forwards_decrement(tc);
2774
2775 ast_channel_appl_set(tc, "AppDial");
2776 ast_channel_data_set(tc, "(Outgoing Line)");
2777
2778 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2779
2780 /* Determine CallerID to store in outgoing channel. */
2781 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2782 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2783 caller.id = stored_clid;
2784 ast_channel_set_caller_event(tc, &caller, NULL);
2785 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2786 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2787 ast_channel_caller(tc)->id.number.str, NULL))) {
2788 /*
2789 * The new channel has no preset CallerID number by the channel
2790 * driver. Use the dialplan extension and hint name.
2791 */
2792 caller.id = stored_clid;
2793 if (!caller.id.name.valid
2794 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2795 ast_channel_connected(chan)->id.name.str, NULL))) {
2796 /*
2797 * No hint name available. We have a connected name supplied by
2798 * the dialplan we can use instead.
2799 */
2800 caller.id.name.valid = 1;
2801 caller.id.name = ast_channel_connected(chan)->id.name;
2802 }
2803 ast_channel_set_caller_event(tc, &caller, NULL);
2804 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2805 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2806 NULL))) {
2807 /* The new channel has no preset CallerID name by the channel driver. */
2808 if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2809 ast_channel_connected(chan)->id.name.str, NULL))) {
2810 /*
2811 * We have a connected name supplied by the dialplan we can
2812 * use instead.
2813 */
2814 caller.id.name.valid = 1;
2815 caller.id.name = ast_channel_connected(chan)->id.name;
2816 ast_channel_set_caller_event(tc, &caller, NULL);
2817 }
2818 }
2819
2820 /* Determine CallerID for outgoing channel to send. */
2821 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2822 struct ast_party_connected_line connected;
2823
2824 ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2825 connected.id = forced_clid;
2826 ast_channel_set_connected_line(tc, &connected, NULL);
2827 } else {
2828 ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2829 }
2830
2831 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2832
2833 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2834
2835 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2836 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2837 ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
2838 }
2839
2840 /* Pass ADSI CPE and transfer capability */
2841 ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
2842 ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
2843
2844 /* If we have an outbound group, set this peer channel to it */
2845 if (outbound_group)
2846 ast_app_group_set_channel(tc, outbound_group);
2847 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2848 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
2849 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2850
2851 /* Check if we're forced by configuration */
2852 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2853 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2854
2855
2856 /* Inherit context and extension */
2857 ast_channel_dialcontext_set(tc, ast_channel_context(chan));
2858 ast_channel_exten_set(tc, ast_channel_exten(chan));
2859
2860 ast_channel_stage_snapshot_done(tc);
2861
2862 /* Save the original channel name to detect call pickup masquerading in. */
2863 tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));
2864
2865 ast_channel_unlock(tc);
2866 ast_channel_unlock(chan);
2867
2868 /* Put channel in the list of outgoing thingies. */
2869 tmp->chan = tc;
2870 AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
2871 }
2872
2873 /* As long as we attempted to dial valid peers, don't throw a warning. */
2874 /* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
2875 if (!num_dialed) {
2876 ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
2877 if (continue_exec) {
2878 /* There is no point in having RetryDial try again */
2879 *continue_exec = 1;
2880 }
2881 strcpy(pa.status, "CHANUNAVAIL");
2882 res = 0;
2883 goto out;
2884 }
2885
2886 /*
2887 * PREDIAL: Run gosub on all of the callee channels
2888 *
2889 * We run the callee predial before ast_call() in case the user
2890 * wishes to do something on the newly created channels before
2891 * the channel does anything important.
2892 *
2893 * Inside the target gosub we will be able to do something with
2894 * the newly created channel name ie: now the calling channel
2895 * can know what channel will be used to call the destination
2896 * ex: now we will know that SIP/abc-123 is calling SIP/def-124
2897 */
2898 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
2899 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
2900 && !AST_LIST_EMPTY(&out_chans)) {
2901 const char *predial_callee;
2902
2903 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
2904 predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
2905 if (predial_callee) {
2906 ast_autoservice_start(chan);
2907 AST_LIST_TRAVERSE(&out_chans, tmp, node) {
2908 ast_pre_call(tmp->chan, predial_callee);
2909 }
2910 ast_autoservice_stop(chan);
2911 ast_free((char *) predial_callee);
2912 }
2913 }
2914
2915 /* Start all outgoing calls */
2916 AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
2917 res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
2918 ast_channel_lock(chan);
2919
2920 /* check the results of ast_call */
2921 if (res) {
2922 /* Again, keep going even if there's an error */
2923 ast_debug(1, "ast call on peer returned %d\n", res);
2924 ast_verb(3, "Couldn't call %s\n", tmp->interface);
2925 if (ast_channel_hangupcause(tmp->chan)) {
2926 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
2927 }
2928 ast_channel_unlock(chan);
2929 ast_cc_call_failed(chan, tmp->chan, tmp->interface);
2930 ast_hangup(tmp->chan);
2931 tmp->chan = NULL;
2932 AST_LIST_REMOVE_CURRENT(node);
2933 chanlist_free(tmp);
2934 continue;
2935 }
2936
2937 ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
2938 ast_channel_unlock(chan);
2939
2940 ast_verb(3, "Called %s\n", tmp->interface);
2941 ast_set_flag64(tmp, DIAL_STILLGOING);
2942
2943 /* If this line is up, don't try anybody else */
2944 if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
2945 break;
2946 }
2947 }
2948 AST_LIST_TRAVERSE_SAFE_END;
2949
2950 if (ast_strlen_zero(args.timeout)) {
2951 to_answer = -1;
2952 to_progress = -1;
2953 } else {
2954 double tmp;
2955 char *anstimeout = strsep(&args.timeout, "^");
2956 if (!ast_strlen_zero(anstimeout)) {
2957 if (sscanf(anstimeout, "%30lf", &tmp) == 1 && tmp > 0) {
2958 to_answer = tmp * 1000;
2959 ast_debug(3, "Dial timeout set to %d ms\n", to_answer);
2960 } else {
2961 ast_log(LOG_WARNING, "Invalid answer timeout specified: '%s'. Setting timeout to infinite\n", anstimeout);
2962 to_answer = -1;
2963 }
2964 } else {
2965 to_answer = -1;
2966 }
2967 if (!ast_strlen_zero(args.timeout)) {
2968 if (sscanf(args.timeout, "%30lf", &tmp) == 1 && tmp > 0) {
2969 to_progress = tmp * 1000;
2970 ast_debug(3, "Dial progress timeout set to %d ms\n", to_progress);
2971 } else {
2972 ast_log(LOG_WARNING, "Invalid progress timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2973 to_progress = -1;
2974 }
2975 } else {
2976 to_progress = -1;
2977 }
2978 }
2979
2980 outgoing = AST_LIST_FIRST(&out_chans);
2981 if (!outgoing) {
2982 strcpy(pa.status, "CHANUNAVAIL");
2983 if (fulldial == num_dialed) {
2984 res = -1;
2985 goto out;
2986 }
2987 } else {
2988 /* Our status will at least be NOANSWER */
2989 strcpy(pa.status, "NOANSWER");
2990 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2991 moh = 1;
2992 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2993 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2994 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2995 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2996 ast_channel_musicclass_set(chan, original_moh);
2997 } else {
2998 ast_moh_start(chan, NULL, NULL);
2999 }
3000 ast_indicate(chan, AST_CONTROL_PROGRESS);
3001 } else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
3002 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
3003 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
3004 ast_indicate(chan, AST_CONTROL_RINGING);
3005 sentringing++;
3006 } else {
3007 ast_indicate(chan, AST_CONTROL_PROGRESS);
3008 }
3009 } else {
3010 ast_indicate(chan, AST_CONTROL_RINGING);
3011 sentringing++;
3012 }
3013 }
3014 }
3015
3016 peer = wait_for_answer(chan, &out_chans, &to_answer, &to_progress, peerflags, opt_args, &pa, &num, &result,
3017 dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
3018 (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
3019 ignore_cc, &forced_clid, &stored_clid, &config);
3020
3021 if (!peer) {
3022 if (result) {
3023 res = result; /* User entered a DTMF digit that matched a context */
3024 } else if (to_answer) { /* Musta gotten hung up */
3025 /* This does not necessarily mean that we dialed without a timeout.
3026 * to_answer is (ab)used by wait_for_answer to to indicate whether or we should continue in the dialplan or exit. */
3027 res = -1;
3028 } else { /* Nobody answered, next please? */
3029 res = 0;
3030 }
3031 } else {
3032 const char *number;
3033 const char *name;
3034 int dial_end_raised = 0;
3035 int cause = -1;
3036
3037 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
3038 ast_answer(chan);
3039 }
3040
3041 /* Ah ha! Someone answered within the desired timeframe. Of course after this
3042 we will always return with -1 so that it is hung up properly after the
3043 conversation. */
3044
3045 if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
3046 && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
3047 cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
3048 if (cause <= 0) {
3049 if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
3050 cause = 0;
3051 } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
3052 || cause < 0) {
3053 ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
3054 opt_args[OPT_ARG_HANGUPCAUSE]);
3055 cause = -1;
3056 }
3057 }
3058 }
3059 hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
3060
3061 /* If appropriate, log that we have a destination channel and set the answer time */
3062
3063 ast_channel_lock(peer);
3064 name = ast_strdupa(ast_channel_name(peer));
3065
3066 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
3067 if (ast_strlen_zero(number)) {
3068 number = NULL;
3069 } else {
3070 number = ast_strdupa(number);
3071 }
3072 ast_channel_unlock(peer);
3073
3074 ast_channel_lock(chan);
3075 ast_channel_stage_snapshot(chan);
3076
3077 strcpy(pa.status, "ANSWER");
3078 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3079
3080 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
3081 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
3082
3083 ast_channel_stage_snapshot_done(chan);
3084 ast_channel_unlock(chan);
3085
3086 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
3087 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
3088 ast_channel_sendurl( peer, args.url );
3089 }
3090 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
3091 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
3092 ast_channel_publish_dial(chan, peer, NULL, pa.status);
3093 /* hang up on the callee -- he didn't want to talk anyway! */
3094 ast_autoservice_chan_hangup_peer(chan, peer);
3095 res = 0;
3096 goto out;
3097 }
3098 }
3099 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
3100 res = 0;
3101 } else {
3102 int digit = 0;
3103 struct ast_channel *chans[2];
3104 struct ast_channel *active_chan;
3105 char *calledfile = NULL, *callerfile = NULL;
3106 int calledstream = 0, callerstream = 0;
3107
3108 chans[0] = chan;
3109 chans[1] = peer;
3110
3111 /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
3112 callerfile = opt_args[OPT_ARG_ANNOUNCE];
3113 calledfile = strsep(&callerfile, ":");
3114
3115 /* stream the file(s) */
3116 if (!ast_strlen_zero(calledfile)) {
3117 res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
3118 if (res) {
3119 res = 0;
3120 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
3121 } else {
3122 calledstream = 1;
3123 }
3124 }
3125 if (!ast_strlen_zero(callerfile)) {
3126 res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
3127 if (res) {
3128 res = 0;
3129 ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
3130 } else {
3131 callerstream = 1;
3132 }
3133 }
3134
3135 /* can't use ast_waitstream, because we're streaming two files at once, and can't block
3136 We'll need to handle both channels at once. */
3137
3138 ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
3139 while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
3140 int mspeer, mschan;
3141
3142 mspeer = ast_sched_wait(ast_channel_sched(peer));
3143 mschan = ast_sched_wait(ast_channel_sched(chan));
3144
3145 if (calledstream) {
3146 if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
3147 ast_stopstream(peer);
3148 calledstream = 0;
3149 }
3150 }
3151 if (callerstream) {
3152 if (mschan < 0 && !ast_channel_timingfunc(chan)) {
3153 ast_stopstream(chan);
3154 callerstream = 0;
3155 }
3156 }
3157
3158 if (!calledstream && !callerstream) {
3159 break;
3160 }
3161
3162 if (mspeer < 0)
3163 mspeer = 1000;
3164
3165 if (mschan < 0)
3166 mschan = 1000;
3167
3168 /* wait for the lowest maximum of the two */
3169 active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
3170 if (active_chan) {
3171 struct ast_channel *other_chan;
3172 struct ast_frame *fr = ast_read(active_chan);
3173
3174 if (!fr) {
3175 ast_autoservice_chan_hangup_peer(chan, peer);
3176 res = -1;
3177 goto done;
3178 }
3179 switch (fr->frametype) {
3180 case AST_FRAME_DTMF_END:
3181 digit = fr->subclass.integer;
3182 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
3183 ast_stopstream(peer);
3184 res = ast_senddigit(chan, digit, 0);
3185 }
3186 break;
3187 case AST_FRAME_CONTROL:
3188 switch (fr->subclass.integer) {
3189 case AST_CONTROL_HANGUP:
3190 ast_frfree(fr);
3191 ast_autoservice_chan_hangup_peer(chan, peer);
3192 res = -1;
3193 goto done;
3194 case AST_CONTROL_CONNECTED_LINE:
3195 /* Pass COLP update to the other channel. */
3196 if (active_chan == chan) {
3197 other_chan = peer;
3198 } else {
3199 other_chan = chan;
3200 }
3201 if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)) {
3202 ast_indicate_data(other_chan, fr->subclass.integer,
3203 fr->data.ptr, fr->datalen);
3204 }
3205 break;
3206 default:
3207 break;
3208 }
3209 break;
3210 default:
3211 /* Ignore all others */
3212 break;
3213 }
3214 ast_frfree(fr);
3215 }
3216 ast_sched_runq(ast_channel_sched(peer));
3217 ast_sched_runq(ast_channel_sched(chan));
3218 }
3219 ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
3220 }
3221
3222 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
3223 /* chan and peer are going into the PBX; as such neither are considered
3224 * outgoing channels any longer */
3225 ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);
3226
3227 ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
3228 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
3229 /* peer goes to the same context and extension as chan, so just copy info from chan*/
3230 ast_channel_lock(peer);
3231 ast_channel_stage_snapshot(peer);
3232 ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
3233 ast_channel_context_set(peer, ast_channel_context(chan));
3234 ast_channel_exten_set(peer, ast_channel_exten(chan));
3235 ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
3236 ast_channel_stage_snapshot_done(peer);
3237 ast_channel_unlock(peer);
3238 if (ast_pbx_start(peer)) {
3239 ast_autoservice_chan_hangup_peer(chan, peer);
3240 }
3241 if (continue_exec)
3242 *continue_exec = 1;
3243 res = 0;
3244 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3245 goto done;
3246 }
3247
3248 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
3249 const char *gosub_result_peer;
3250 char *gosub_argstart;
3251 char *gosub_args = NULL;
3252 int gosub_res = -1;
3253
3254 ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
3255 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
3256 if (gosub_argstart) {
3257 const char *what_is_s = "s";
3258 *gosub_argstart = 0;
3259 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)) &&
3260 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))) {
3261 what_is_s = "~~s~~";
3262 }
3263 if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
3264 gosub_args = NULL;
3265 }
3266 *gosub_argstart = ',';
3267 } else {
3268 const char *what_is_s = "s";
3269 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)) &&
3270 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))) {
3271 what_is_s = "~~s~~";
3272 }
3273 if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
3274 gosub_args = NULL;
3275 }
3276 }
3277 if (gosub_args) {
3278 gosub_res = ast_app_exec_sub(chan, peer, gosub_args, 0);
3279 ast_free(gosub_args);
3280 } else {
3281 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
3282 }
3283
3284 ast_channel_lock_both(chan, peer);
3285
3286 if (!gosub_res && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
3287 char *gosub_transfer_dest;
3288 char *gosub_result = ast_strdupa(gosub_result_peer);
3289 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
3290
3291 /* Inherit return value from the peer, so it can be used in the master */
3292 if (gosub_retval) {
3293 pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
3294 }
3295
3296 ast_channel_unlock(peer);
3297 ast_channel_unlock(chan);
3298
3299 if (!strcasecmp(gosub_result, "BUSY")) {
3300 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3301 ast_set_flag64(peerflags, OPT_GO_ON);
3302 gosub_res = -1;
3303 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
3304 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3305 ast_set_flag64(peerflags, OPT_GO_ON);
3306 gosub_res = -1;
3307 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
3308 /* Hangup peer and continue with the next extension priority. */
3309 ast_set_flag64(peerflags, OPT_GO_ON);
3310 gosub_res = -1;
3311 } else if (!strcasecmp(gosub_result, "ABORT")) {
3312 /* Hangup both ends unless the caller has the g flag */
3313 gosub_res = -1;
3314 } else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
3315 gosub_transfer_dest = gosub_result + 5;
3316 gosub_res = -1;
3317 /* perform a transfer to a new extension */
3318 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
3319 ast_replace_subargument_delimiter(gosub_transfer_dest);
3320 }
3321 if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
3322 ast_set_flag64(peerflags, OPT_GO_ON);
3323 }
3324 }
3325 if (gosub_res) {
3326 res = gosub_res;
3327 if (!dial_end_raised) {
3328 ast_channel_publish_dial(chan, peer, NULL, gosub_result);
3329 dial_end_raised = 1;
3330 }
3331 }
3332 } else {
3333 ast_channel_unlock(peer);
3334 ast_channel_unlock(chan);
3335 }
3336 }
3337
3338 if (!res) {
3339
3340 /* None of the Dial options changed our status; inform
3341 * everyone that this channel answered
3342 */
3343 if (!dial_end_raised) {
3344 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3345 dial_end_raised = 1;
3346 }
3347
3348 if (!ast_tvzero(calldurationlimit)) {
3349 struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
3350 ast_channel_lock(peer);
3351 ast_channel_whentohangup_set(peer, &whentohangup);
3352 ast_channel_unlock(peer);
3353 }
3354 if (!ast_strlen_zero(dtmfcalled)) {
3355 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
3356 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
3357 }
3358 if (!ast_strlen_zero(dtmfcalling)) {
3359 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
3360 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
3361 }
3362 }
3363
3364 if (res) { /* some error */
3365 if (!ast_check_hangup(chan) && ast_check_hangup(peer)) {
3366 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
3367 }
3368 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3369 if (ast_bridge_setup_after_goto(peer)
3370 || ast_pbx_start(peer)) {
3371 ast_autoservice_chan_hangup_peer(chan, peer);
3372 }
3373 res = -1;
3374 } else {
3375 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
3376 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
3377 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
3378 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
3379 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
3380 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
3381 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
3382 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
3383 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
3384 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
3385 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
3386 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
3387 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
3388 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
3389 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
3390 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
3391 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
3392 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
3393 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
3394 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
3395
3396 config.end_bridge_callback = end_bridge_callback;
3397 config.end_bridge_callback_data = chan;
3398 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
3399
3400 if (moh) {
3401 moh = 0;
3402 ast_moh_stop(chan);
3403 } else if (sentringing) {
3404 sentringing = 0;
3405 ast_indicate(chan, -1);
3406 }
3407 /* Be sure no generators are left on it and reset the visible indication */
3408 ast_deactivate_generator(chan);
3409 ast_channel_visible_indication_set(chan, 0);
3410 /* Make sure channels are compatible */
3411 res = ast_channel_make_compatible(chan, peer);
3412 if (res < 0) {
3413 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));
3414 ast_autoservice_chan_hangup_peer(chan, peer);
3415 res = -1;
3416 goto done;
3417 }
3418 if (opermode) {
3419 struct oprmode oprmode;
3420
3421 oprmode.peer = peer;
3422 oprmode.mode = opermode;
3423
3424 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
3425 }
3426 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3427
3428 res = ast_bridge_call(chan, peer, &config);
3429 }
3430 }
3431out:
3432 if (moh) {
3433 moh = 0;
3434 ast_moh_stop(chan);
3435 } else if (sentringing) {
3436 sentringing = 0;
3437 ast_indicate(chan, -1);
3438 }
3439
3440 if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3441 ast_filedelete(pa.privintro, NULL);
3442 if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3443 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
3444 } else {
3445 ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
3446 }
3447 }
3448
3449 ast_channel_early_bridge(chan, NULL);
3450 /* forward 'answered elsewhere' if we received it */
3451 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
3452 hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
3453 } else if (pa.canceled) { /* Caller canceled */
3454 if (ast_channel_hangupcause(chan))
3455 hanguptreecause = ast_channel_hangupcause(chan);
3456 else
3457 hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
3458 }
3459 hanguptree(&out_chans, NULL, hanguptreecause);
3460 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3461 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
3462
3463 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
3464 if (!ast_tvzero(calldurationlimit))
3465 memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
3466 res = 0;
3467 }
3468
3469done:
3470 if (config.answer_topology) {
3471 ast_trace(2, "%s Cleaning up topology: %p %s\n",
3472 peer ? ast_channel_name(peer) : "<no channel>", &config.answer_topology,
3473 ast_str_tmp(256, ast_stream_topology_to_str(config.answer_topology, &STR_TMP)));
3474
3475 /*
3476 * At this point, the channel driver that answered should have bumped the
3477 * topology refcount for itself. Here we're cleaning up the reference we added
3478 * in wait_for_answer().
3479 */
3480 ast_stream_topology_free(config.answer_topology);
3481 }
3482 if (config.warning_sound) {
3483 ast_free((char *)config.warning_sound);
3484 }
3485 if (config.end_sound) {
3486 ast_free((char *)config.end_sound);
3487 }
3488 if (config.start_sound) {
3489 ast_free((char *)config.start_sound);
3490 }
3491 ast_ignore_cc(chan);
3492 SCOPE_EXIT_RTN_VALUE(res, "%s: Done\n", ast_channel_name(chan));
3493}
3494
3495static int dial_exec(struct ast_channel *chan, const char *data)
3496{
3497 struct ast_flags64 peerflags;
3498
3499 memset(&peerflags, 0, sizeof(peerflags));
3500
3501 return dial_exec_full(chan, data, &peerflags, NULL);
3502}
3503
3504static int retrydial_exec(struct ast_channel *chan, const char *data)
3505{
3506 char *parse;
3507 const char *context = NULL;
3508 int sleepms = 0, loops = 0, res = -1;
3509 struct ast_flags64 peerflags = { 0, };
3510 AST_DECLARE_APP_ARGS(args,
3511 AST_APP_ARG(announce);
3512 AST_APP_ARG(sleep);
3513 AST_APP_ARG(retries);
3514 AST_APP_ARG(dialdata);
3515 );
3516
3517 if (ast_strlen_zero(data)) {
3518 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
3519 return -1;
3520 }
3521
3522 parse = ast_strdupa(data);
3523 AST_STANDARD_APP_ARGS(args, parse);
3524
3525 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
3526 sleepms *= 1000;
3527
3528 if (!ast_strlen_zero(args.retries)) {
3529 loops = atoi(args.retries);
3530 }
3531
3532 if (!args.dialdata) {
3533 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
3534 goto done;
3535 }
3536
3537 if (sleepms < 1000)
3538 sleepms = 10000;
3539
3540 if (!loops)
3541 loops = -1; /* run forever */
3542
3543 ast_channel_lock(chan);
3544 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
3545 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
3546 ast_channel_unlock(chan);
3547
3548 res = 0;
3549 while (loops) {
3550 int continue_exec;
3551
3552 ast_channel_data_set(chan, "Retrying");
3553 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3554 ast_moh_stop(chan);
3555
3556 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
3557 if (continue_exec)
3558 break;
3559
3560 if (res == 0) {
3561 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
3562 if (!ast_strlen_zero(args.announce)) {
3563 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3564 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3565 ast_waitstream(chan, AST_DIGIT_ANY);
3566 } else
3567 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3568 }
3569 if (!res && sleepms) {
3570 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3571 ast_moh_start(chan, NULL, NULL);
3572 res = ast_waitfordigit(chan, sleepms);
3573 }
3574 } else {
3575 if (!ast_strlen_zero(args.announce)) {
3576 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3577 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3578 res = ast_waitstream(chan, "");
3579 } else
3580 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3581 }
3582 if (sleepms) {
3583 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3584 ast_moh_start(chan, NULL, NULL);
3585 if (!res)
3586 res = ast_waitfordigit(chan, sleepms);
3587 }
3588 }
3589 }
3590
3591 if (res < 0 || res == AST_PBX_INCOMPLETE) {
3592 break;
3593 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
3594 if (onedigit_goto(chan, context, (char) res, 1)) {
3595 res = 0;
3596 break;
3597 }
3598 }
3599 loops--;
3600 }
3601 if (loops == 0)
3602 res = 0;
3603 else if (res == 1)
3604 res = 0;
3605
3606 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3607 ast_moh_stop(chan);
3608 done:
3609 return res;
3610}
3611
3612static int unload_module(void)
3613{
3614 int res;
3615
3616 res = ast_unregister_application(app);
3617 res |= ast_unregister_application(rapp);
3618
3619 return res;
3620}
3621
3622static int load_module(void)
3623{
3624 int res;
3625
3626 res = ast_register_application_xml(app, dial_exec);
3627 res |= ast_register_application_xml(rapp, retrydial_exec);
3628
3629 return res;
3630}
3631
3632AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application",
3633 .support_level = AST_MODULE_SUPPORT_CORE,
3634 .load = load_module,
3635 .unload = unload_module,
3636);
ast_aoc_destroy_encoded
void * ast_aoc_destroy_encoded(struct ast_aoc_encoded *encoded)
free an ast_aoc_encoded object
Definition aoc.c:322
ast_aoc_get_msg_type
enum ast_aoc_type ast_aoc_get_msg_type(struct ast_aoc_decoded *decoded)
get the message type, AOC-D, AOC-E, or AOC Request
Definition aoc.c:901
ast_aoc_decode
struct ast_aoc_decoded * ast_aoc_decode(struct ast_aoc_encoded *encoded, size_t size, struct ast_channel *chan)
decodes an encoded aoc payload.
Definition aoc.c:458
ast_aoc_destroy_decoded
void * ast_aoc_destroy_decoded(struct ast_aoc_decoded *decoded)
free an ast_aoc_decoded object
Definition aoc.c:316
ast_aoc_encode
struct ast_aoc_encoded * ast_aoc_encode(struct ast_aoc_decoded *decoded, size_t *out_size, struct ast_channel *chan)
encodes a decoded aoc structure so it can be passed on the wire
Definition aoc.c:659
AST_AOC_S
@ AST_AOC_S
Definition aoc.h:64
digit
char digit
Definition app_alarmreceiver.c:131
topology_ds_destroy
static void topology_ds_destroy(void *data)
Definition app_dial.c:827
DIAL_STILLGOING
#define DIAL_STILLGOING
Definition app_dial.c:707
dial_exec_full
static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
Definition app_dial.c:2333
OPT_PREDIAL_CALLER
#define OPT_PREDIAL_CALLER
Definition app_dial.c:718
OPT_CANCEL_ELSEWHERE
#define OPT_CANCEL_ELSEWHERE
Definition app_dial.c:710
get_cid_name
static const char * get_cid_name(char *name, int namelen, struct ast_channel *chan)
Definition app_dial.c:914
app
static const char app[]
Definition app_dial.c:670
dial_exec_options
static const struct ast_app_option dial_exec_options[128]
Definition app_dial.c:792
OPT_PEER_H
#define OPT_PEER_H
Definition app_dial.c:711
do_forward
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)
Definition app_dial.c:944
OPT_PREDIAL_CALLEE
#define OPT_PREDIAL_CALLEE
Definition app_dial.c:717
DIAL_CALLERID_ABSENT
#define DIAL_CALLERID_ABSENT
Definition app_dial.c:709
OPT_FORCE_CID_PRES
#define OPT_FORCE_CID_PRES
Definition app_dial.c:715
setup_peer_after_bridge_goto
static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
Definition app_dial.c:2311
CAN_EARLY_BRIDGE
#define CAN_EARLY_BRIDGE(flags, chan, peer)
Definition app_dial.c:794
OPT_TOPOLOGY_PRESERVE
#define OPT_TOPOLOGY_PRESERVE
Definition app_dial.c:722
OPT_RING_WITH_EARLY_MEDIA
#define OPT_RING_WITH_EARLY_MEDIA
Definition app_dial.c:719
OPT_FORCE_CID_TAG
#define OPT_FORCE_CID_TAG
Definition app_dial.c:714
OPT_HEARPULSING
#define OPT_HEARPULSING
Definition app_dial.c:721
dial_exec
static int dial_exec(struct ast_channel *chan, const char *data)
Definition app_dial.c:3496
DIAL_NOFORWARDHTML
#define DIAL_NOFORWARDHTML
Definition app_dial.c:708
AST_MAX_WATCHERS
#define AST_MAX_WATCHERS
Definition app_dial.c:865
OPT_CANCEL_TIMEOUT
#define OPT_CANCEL_TIMEOUT
Definition app_dial.c:713
chanlist_free
static void chanlist_free(struct chanlist *outgoing)
Definition app_dial.c:839
publish_dial_end_event
static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
Definition app_dial.c:1156
onedigit_goto
static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
Definition app_dial.c:899
rapp
static const char rapp[]
Definition app_dial.c:671
handle_cause
static void handle_cause(int cause, struct cause_args *num)
Definition app_dial.c:877
setup_privacy_args
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'
Definition app_dial.c:2156
set_duration_var
static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
Definition app_dial.c:1196
update_connected_line_from_peer
static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
Definition app_dial.c:1176
OPT_ARG_CALLEE_GO_ON
@ OPT_ARG_CALLEE_GO_ON
Definition app_dial.c:732
OPT_ARG_SENDDTMF
@ OPT_ARG_SENDDTMF
Definition app_dial.c:726
OPT_ARG_DURATION_STOP
@ OPT_ARG_DURATION_STOP
Definition app_dial.c:734
OPT_ARG_PREDIAL_CALLEE
@ OPT_ARG_PREDIAL_CALLEE
Definition app_dial.c:741
OPT_ARG_RINGBACK
@ OPT_ARG_RINGBACK
Definition app_dial.c:730
OPT_ARG_MUSICBACK
@ OPT_ARG_MUSICBACK
Definition app_dial.c:729
OPT_ARG_CALLEE_GOSUB
@ OPT_ARG_CALLEE_GOSUB
Definition app_dial.c:731
OPT_ARG_HANGUPCAUSE
@ OPT_ARG_HANGUPCAUSE
Definition app_dial.c:743
OPT_ARG_FORCE_CID_PRES
@ OPT_ARG_FORCE_CID_PRES
Definition app_dial.c:740
OPT_ARG_ANNOUNCE
@ OPT_ARG_ANNOUNCE
Definition app_dial.c:725
OPT_ARG_GOTO
@ OPT_ARG_GOTO
Definition app_dial.c:727
OPT_ARG_DURATION_LIMIT
@ OPT_ARG_DURATION_LIMIT
Definition app_dial.c:728
OPT_ARG_ORIGINAL_CLID
@ OPT_ARG_ORIGINAL_CLID
Definition app_dial.c:737
OPT_ARG_OPERMODE
@ OPT_ARG_OPERMODE
Definition app_dial.c:735
OPT_ARG_FORCECLID
@ OPT_ARG_FORCECLID
Definition app_dial.c:738
OPT_ARG_PREDIAL_CALLER
@ OPT_ARG_PREDIAL_CALLER
Definition app_dial.c:742
OPT_ARG_ARRAY_SIZE
@ OPT_ARG_ARRAY_SIZE
Definition app_dial.c:745
OPT_ARG_PRIVACY
@ OPT_ARG_PRIVACY
Definition app_dial.c:733
OPT_ARG_SCREEN_NOINTRO
@ OPT_ARG_SCREEN_NOINTRO
Definition app_dial.c:736
OPT_ARG_FORCE_CID_TAG
@ OPT_ARG_FORCE_CID_TAG
Definition app_dial.c:739
detect_disconnect
static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
Definition app_dial.c:1979
OPT_HANGUPCAUSE
#define OPT_HANGUPCAUSE
Definition app_dial.c:720
hanguptree
static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
Definition app_dial.c:847
OPT_RESETCDR
@ OPT_RESETCDR
Definition app_dial.c:675
OPT_SCREEN_NOINTRO
@ OPT_SCREEN_NOINTRO
Definition app_dial.c:685
OPT_DTMF_EXIT
@ OPT_DTMF_EXIT
Definition app_dial.c:676
OPT_ANNOUNCE
@ OPT_ANNOUNCE
Definition app_dial.c:674
OPT_CALLEE_PARK
@ OPT_CALLEE_PARK
Definition app_dial.c:698
OPT_DURATION_LIMIT
@ OPT_DURATION_LIMIT
Definition app_dial.c:683
OPT_SCREEN_NOCALLERID
@ OPT_SCREEN_NOCALLERID
Definition app_dial.c:686
OPT_IGNORE_FORWARDING
@ OPT_IGNORE_FORWARDING
Definition app_dial.c:700
OPT_OPERMODE
@ OPT_OPERMODE
Definition app_dial.c:697
OPT_DURATION_STOP
@ OPT_DURATION_STOP
Definition app_dial.c:691
OPT_GO_ON
@ OPT_GO_ON
Definition app_dial.c:679
OPT_RINGBACK
@ OPT_RINGBACK
Definition app_dial.c:690
OPT_GOTO
@ OPT_GOTO
Definition app_dial.c:696
OPT_IGNORE_CONNECTEDLINE
@ OPT_IGNORE_CONNECTEDLINE
Definition app_dial.c:687
OPT_CALLEE_TRANSFER
@ OPT_CALLEE_TRANSFER
Definition app_dial.c:692
OPT_SENDDTMF
@ OPT_SENDDTMF
Definition app_dial.c:677
OPT_CALLER_MIXMONITOR
@ OPT_CALLER_MIXMONITOR
Definition app_dial.c:703
OPT_CALLER_PARK
@ OPT_CALLER_PARK
Definition app_dial.c:699
OPT_CALLER_MONITOR
@ OPT_CALLER_MONITOR
Definition app_dial.c:695
OPT_CALLEE_MONITOR
@ OPT_CALLEE_MONITOR
Definition app_dial.c:694
OPT_CALLEE_GOSUB
@ OPT_CALLEE_GOSUB
Definition app_dial.c:701
OPT_CALLER_HANGUP
@ OPT_CALLER_HANGUP
Definition app_dial.c:681
OPT_FORCECLID
@ OPT_FORCECLID
Definition app_dial.c:678
OPT_CALLEE_HANGUP
@ OPT_CALLEE_HANGUP
Definition app_dial.c:680
OPT_SCREENING
@ OPT_SCREENING
Definition app_dial.c:688
OPT_MUSICBACK
@ OPT_MUSICBACK
Definition app_dial.c:684
OPT_CALLER_TRANSFER
@ OPT_CALLER_TRANSFER
Definition app_dial.c:693
OPT_CALLEE_MIXMONITOR
@ OPT_CALLEE_MIXMONITOR
Definition app_dial.c:702
OPT_ORIGINAL_CLID
@ OPT_ORIGINAL_CLID
Definition app_dial.c:682
OPT_PRIVACY
@ OPT_PRIVACY
Definition app_dial.c:689
topology_ds_info
static const struct ast_datastore_info topology_ds_info
Definition app_dial.c:832
load_module
static int load_module(void)
Definition app_dial.c:3623
retrydial_exec
static int retrydial_exec(struct ast_channel *chan, const char *data)
Definition app_dial.c:3505
dial_handle_playtones
static int dial_handle_playtones(struct ast_channel *chan, const char *data)
Definition app_dial.c:2271
end_bridge_callback
static void end_bridge_callback(void *data)
Definition app_dial.c:2255
unload_module
static int unload_module(void)
Definition app_dial.c:3613
OPT_CALLER_ANSWER
#define OPT_CALLER_ANSWER
Definition app_dial.c:716
wait_for_answer
static struct ast_channel * wait_for_answer(struct ast_channel *in, struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags, char *opt_args[], struct privacy_args *pa, const struct cause_args *num_in, int *result, char *dtmf_progress, char *mf_progress, char *mf_wink, char *sf_progress, char *sf_wink, const int hearpulsing, const int ignore_cc, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid, struct ast_bridge_config *config)
Definition app_dial.c:1209
end_bridge_callback_data_fixup
static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator)
Definition app_dial.c:2267
valid_priv_reply
static int valid_priv_reply(struct ast_flags64 *opts, int res)
Definition app_dial.c:2009
OPT_CALLEE_GO_ON
#define OPT_CALLEE_GO_ON
Definition app_dial.c:712
status
jack_status_t status
Definition app_jack.c:149
str
const char * str
Definition app_jack.c:150
silencethreshold
static int silencethreshold
Definition app_voicemail.c:1109
strsep
char * strsep(char **str, const char *delims)
ast_free
#define ast_free(a)
Definition astmm.h:180
ast_strdup
#define ast_strdup(str)
A wrapper for strdup()
Definition astmm.h:241
ast_strdupa
#define ast_strdupa(s)
duplicate a string in memory from the stack
Definition astmm.h:298
ast_asprintf
#define ast_asprintf(ret, fmt,...)
A wrapper for asprintf()
Definition astmm.h:267
ast_calloc
#define ast_calloc(num, len)
A wrapper for calloc()
Definition astmm.h:202
ast_log
#define ast_log
Definition astobj2.c:42
ao2_ref
#define ao2_ref(o, delta)
Reference/unreference an object and return the old refcount.
Definition astobj2.h:459
ao2_bump
#define ao2_bump(obj)
Bump refcount on an AO2 object by one, returning the object.
Definition astobj2.h:480
ast_bridge_setup_after_goto
int ast_bridge_setup_after_goto(struct ast_channel *chan)
Setup any after bridge goto location to begin execution.
Definition bridge_after.c:435
ast_bridge_set_after_go_on
void ast_bridge_set_after_go_on(struct ast_channel *chan, const char *context, const char *exten, int priority, const char *parseable_goto)
Set channel to go on in the dialplan after the bridge.
Definition bridge_after.c:622
ast_bridge_set_after_h
void ast_bridge_set_after_h(struct ast_channel *chan, const char *context)
Set channel to run the h exten after the bridge.
Definition bridge_after.c:617
AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN
#define AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN
Definition callerid.h:440
ast_parse_caller_presentation
int ast_parse_caller_presentation(const char *data)
Convert caller ID text code to value (used in config file parsing)
Definition callerid.c:1343
ast_callerid_parse
int ast_callerid_parse(char *instr, char **name, char **location)
Destructively parse inbuf into name and location (or number)
Definition callerid.c:1162
AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER
@ AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER
Definition callerid.h:554
ast_shrink_phone_number
void ast_shrink_phone_number(char *n)
Shrink a phone number in place to just digits (more accurately it just removes ()'s,...
Definition callerid.c:1101
AST_CAUSE_CONGESTION
#define AST_CAUSE_CONGESTION
Definition causes.h:153
AST_CAUSE_ANSWERED_ELSEWHERE
#define AST_CAUSE_ANSWERED_ELSEWHERE
Definition causes.h:114
AST_CAUSE_NO_ROUTE_DESTINATION
#define AST_CAUSE_NO_ROUTE_DESTINATION
Definition causes.h:100
AST_CAUSE_UNREGISTERED
#define AST_CAUSE_UNREGISTERED
Definition causes.h:154
AST_CAUSE_BUSY
#define AST_CAUSE_BUSY
Definition causes.h:149
AST_CAUSE_NO_ANSWER
#define AST_CAUSE_NO_ANSWER
Definition causes.h:109
AST_CAUSE_NORMAL_CLEARING
#define AST_CAUSE_NORMAL_CLEARING
Definition causes.h:106
ast_cc_call_failed
void ast_cc_call_failed(struct ast_channel *incoming, struct ast_channel *outgoing, const char *const dialstring)
Make CCBS available in the case that ast_call fails.
Definition ccss.c:4261
ast_ignore_cc
void ast_ignore_cc(struct ast_channel *chan)
Mark the channel to ignore further CC activity.
Definition ccss.c:3747
ast_cc_is_recall
int ast_cc_is_recall(struct ast_channel *chan, int *core_id, const char *const monitor_type)
Decide if a call to a particular channel is a CC recall.
Definition ccss.c:3459
ast_handle_cc_control_frame
void ast_handle_cc_control_frame(struct ast_channel *inbound, struct ast_channel *outbound, void *frame_data)
Properly react to a CC control frame.
Definition ccss.c:2334
ast_cc_completed
int ast_cc_completed(struct ast_channel *chan, const char *const debug,...)
Indicate recall has been acknowledged.
Definition ccss.c:3885
ast_cc_busy_interface
void ast_cc_busy_interface(struct ast_channel *inbound, struct ast_cc_config_params *cc_params, const char *monitor_type, const char *const device_name, const char *const dialstring, void *private_data)
Callback made from ast_cc_callback for certain channel types.
Definition ccss.c:4296
ast_cc_extension_monitor_add_dialstring
void ast_cc_extension_monitor_add_dialstring(struct ast_channel *incoming, const char *const dialstring, const char *const device_name)
Add a child dialstring to an extension monitor.
Definition ccss.c:2022
ast_cc_call_init
int ast_cc_call_init(struct ast_channel *chan, int *ignore_cc)
Start the CC process on a call.
Definition ccss.c:2429
ast_cc_failed
int ast_cc_failed(int core_id, const char *const debug,...)
Indicate failure has occurred.
Definition ccss.c:3924
ast_cc_callback
int ast_cc_callback(struct ast_channel *inbound, const char *const tech, const char *const dest, ast_cc_callback_fn callback)
Run a callback for potential matching destinations.
Definition ccss.c:4311
ast_cdr_reset
int ast_cdr_reset(const char *channel_name, int keep_variables)
Reset the detail record.
Definition cdr.c:3746
priority
static int priority
Definition cdr_beanstalkd.c:76
connected
static int connected
Definition cdr_pgsql.c:73
result
static PGresult * result
Definition cel_pgsql.c:84
config
static const char config[]
Definition chan_ooh323.c:111
ast_channel_exten_set
void ast_channel_exten_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:240
ast_waitfordigit
int ast_waitfordigit(struct ast_channel *c, int ms)
Waits for a digit.
Definition channel.c:3213
ast_str2cause
int ast_str2cause(const char *name) attribute_pure
Convert the string form of a cause code to a number.
Definition channel.c:626
ast_channel_name
const char * ast_channel_name(const struct ast_channel *chan)
ast_autoservice_stop
int ast_autoservice_stop(struct ast_channel *chan)
Stop servicing a channel for us...
Definition autoservice.c:266
ast_channel_appl_set
void ast_channel_appl_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:192
ast_channel_visible_indication_set
void ast_channel_visible_indication_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:353
ast_channel_get_device_name
int ast_channel_get_device_name(struct ast_channel *chan, char *device_name, size_t name_buffer_length)
Get a device name given its channel structure.
Definition channel.c:10606
ast_party_redirecting_init
void ast_party_redirecting_init(struct ast_party_redirecting *init)
Initialize the given redirecting structure.
Definition channel.c:2109
ast_call
int ast_call(struct ast_channel *chan, const char *addr, int timeout)
Make a call.
Definition channel.c:6518
ast_channel_clear_flag
void ast_channel_clear_flag(struct ast_channel *chan, unsigned int flag)
Clear a flag on a channel.
Definition channel.c:11144
ast_channel_datastore_add
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
Add a datastore to a channel.
Definition channel.c:2376
ast_party_id_init
void ast_party_id_init(struct ast_party_id *init)
Initialize the given party id structure.
Definition channel.c:1744
ast_channel_connected_line_sub
int ast_channel_connected_line_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *connected_info, int frame)
Run a connected line interception subroutine and update a channel's connected line information.
Definition channel.c:10444
ast_party_number_init
void ast_party_number_init(struct ast_party_number *init)
Initialize the given number structure.
Definition channel.c:1631
ast_hangup
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition channel.c:2574
AST_CHANNEL_REQUESTOR_BRIDGE_PEER
@ AST_CHANNEL_REQUESTOR_BRIDGE_PEER
Definition channel.h:1525
ast_channel_set_connected_line
void ast_channel_set_connected_line(struct ast_channel *chan, const struct ast_party_connected_line *connected, const struct ast_set_party_connected_line *update)
Set the connected line information in the Asterisk channel.
Definition channel.c:8414
ast_channel_musicclass
const char * ast_channel_musicclass(const struct ast_channel *chan)
ast_channel_sendhtml
int ast_channel_sendhtml(struct ast_channel *channel, int subclass, const char *data, int datalen)
Sends HTML on given channel Send HTML or URL on link.
Definition channel.c:6685
ast_party_connected_line_free
void ast_party_connected_line_free(struct ast_party_connected_line *doomed)
Destroy the connected line information contents.
Definition channel.c:2059
ast_channel_get_up_time_ms
int64_t ast_channel_get_up_time_ms(struct ast_channel *chan)
Obtain how long it has been since the channel was answered in ms.
Definition channel.c:2869
ast_channel_set_caller_event
void ast_channel_set_caller_event(struct ast_channel *chan, const struct ast_party_caller *caller, const struct ast_set_party_caller *update)
Set the caller id information in the Asterisk channel and generate an AMI event if the caller id name...
Definition channel.c:7446
ast_waitfor_n
struct ast_channel * ast_waitfor_n(struct ast_channel **chan, int n, int *ms)
Waits for input on a group of channels Wait for input on an array of channels for a given # of millis...
Definition channel.c:3195
ast_senddigit
int ast_senddigit(struct ast_channel *chan, char digit, unsigned int duration)
Send a DTMF digit to a channel.
Definition channel.c:5031
ast_channel_lock
#define ast_channel_lock(chan)
Definition channel.h:2989
ast_channel_make_compatible
int ast_channel_make_compatible(struct ast_channel *chan, struct ast_channel *peer)
Make the frame formats of two channels compatible.
Definition channel.c:6777
ast_channel_data_set
void ast_channel_data_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:209
AST_FEATURE_AUTOMIXMON
@ AST_FEATURE_AUTOMIXMON
Definition channel.h:1089
AST_FEATURE_REDIRECT
@ AST_FEATURE_REDIRECT
Definition channel.h:1084
AST_FEATURE_PARKCALL
@ AST_FEATURE_PARKCALL
Definition channel.h:1088
AST_FEATURE_AUTOMON
@ AST_FEATURE_AUTOMON
Definition channel.h:1087
AST_FEATURE_DISCONNECT
@ AST_FEATURE_DISCONNECT
Definition channel.h:1085
ast_channel_redirecting
struct ast_party_redirecting * ast_channel_redirecting(struct ast_channel *chan)
Definition channel_internal_api.c:775
ast_channel_transfercapability
unsigned short ast_channel_transfercapability(const struct ast_channel *chan)
Definition channel_internal_api.c:373
ast_party_connected_line_copy
void ast_party_connected_line_copy(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src)
Copy the source connected line information to the destination connected line.
Definition channel.c:2018
ast_channel_flags
struct ast_flags * ast_channel_flags(struct ast_channel *chan)
Definition channel_internal_api.c:1127
ast_party_connected_line_set
void ast_party_connected_line_set(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src, const struct ast_set_party_connected_line *update)
Set the connected line information based on another connected line source.
Definition channel.c:2041
ast_channel_priority
int ast_channel_priority(const struct ast_channel *chan)
Definition channel_internal_api.c:316
ast_channel_lock_both
#define ast_channel_lock_both(chan1, chan2)
Lock two channels.
Definition channel.h:2996
ast_channel_connected
struct ast_party_connected_line * ast_channel_connected(struct ast_channel *chan)
Definition channel_internal_api.c:759
ast_channel_datastore_inherit
int ast_channel_datastore_inherit(struct ast_channel *from, struct ast_channel *to)
Inherit datastores from a parent to a child.
Definition channel.c:2359
ast_channel_req_accountcodes
void ast_channel_req_accountcodes(struct ast_channel *chan, const struct ast_channel *requestor, enum ast_channel_requestor_relationship relationship)
Setup new channel accountcodes from the requestor channel after ast_request().
Definition channel.c:6491
ast_channel_context
const char * ast_channel_context(const struct ast_channel *chan)
Definition channel_internal_api.c:215
ast_deactivate_generator
void ast_deactivate_generator(struct ast_channel *chan)
Definition channel.c:2926
ast_check_hangup_locked
int ast_check_hangup_locked(struct ast_channel *chan)
Definition channel.c:460
ast_write
int ast_write(struct ast_channel *chan, struct ast_frame *frame)
Write a frame to a channel This function writes the given frame to the indicated channel.
Definition channel.c:5201
ast_autoservice_start
int ast_autoservice_start(struct ast_channel *chan)
Automatically service a channel for us...
Definition autoservice.c:200
ast_read
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition channel.c:4312
ast_channel_update_connected_line
void ast_channel_update_connected_line(struct ast_channel *chan, const struct ast_party_connected_line *connected, const struct ast_set_party_connected_line *update)
Indicate that the connected line information has changed.
Definition channel.c:9199
ast_channel_adsicpe
ast_channel_adsicpe
Definition channel.h:888
ast_party_caller_set_init
void ast_party_caller_set_init(struct ast_party_caller *init, const struct ast_party_caller *guide)
Initialize the given caller structure using the given guide for a set update operation.
Definition channel.c:1986
ast_party_id_set_init
void ast_party_id_set_init(struct ast_party_id *init, const struct ast_party_id *guide)
Initialize the given party id structure using the given guide for a set update operation.
Definition channel.c:1767
ast_channel_creationtime
struct timeval ast_channel_creationtime(struct ast_channel *chan)
Definition channel_internal_api.c:843
ast_channel_redirecting_sub
int ast_channel_redirecting_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *redirecting_info, int is_frame)
Run a redirecting interception subroutine and update a channel's redirecting information.
Definition channel.c:10489
ast_channel_inherit_variables
void ast_channel_inherit_variables(const struct ast_channel *parent, struct ast_channel *child)
Inherits channel variable from parent to child channel.
Definition channel.c:6833
ast_connected_line_parse_data
int ast_connected_line_parse_data(const unsigned char *data, size_t datalen, struct ast_party_connected_line *connected)
Parse connected line indication frame data.
Definition channel.c:8891
ast_request_with_stream_topology
struct ast_channel * ast_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *addr, int *cause)
Requests a channel (specifying stream topology)
Definition channel.c:6416
ast_channel_supports_html
int ast_channel_supports_html(struct ast_channel *channel)
Checks for HTML support on a channel.
Definition channel.c:6680
ast_check_hangup
int ast_check_hangup(struct ast_channel *chan)
Check to see if a channel is needing hang up.
Definition channel.c:446
ast_channel_get_stream_topology
struct ast_stream_topology * ast_channel_get_stream_topology(const struct ast_channel *chan)
Retrieve the topology of streams on a channel.
Definition channel_internal_api.c:1499
ast_channel_hangupcause
int ast_channel_hangupcause(const struct ast_channel *chan)
Definition channel_internal_api.c:298
ast_channel_get_duration_ms
int64_t ast_channel_get_duration_ms(struct ast_channel *chan)
Obtain how long it's been, in milliseconds, since the channel was created.
Definition channel.c:2854
ast_channel_dialed
struct ast_party_dialed * ast_channel_dialed(struct ast_channel *chan)
Definition channel_internal_api.c:771
ast_indicate_data
int ast_indicate_data(struct ast_channel *chan, int condition, const void *data, size_t datalen)
Indicates condition of channel, with payload.
Definition channel.c:4710
ast_channel_zone
struct ast_tone_zone * ast_channel_zone(const struct ast_channel *chan)
Definition channel_internal_api.c:612
ast_channel_set_flag
void ast_channel_set_flag(struct ast_channel *chan, unsigned int flag)
Set a flag on a channel.
Definition channel.c:11137
ast_channel_update_redirecting
void ast_channel_update_redirecting(struct ast_channel *chan, const struct ast_party_redirecting *redirecting, const struct ast_set_party_redirecting *update)
Indicate that the redirecting id has changed.
Definition channel.c:10390
ast_channel_timingfunc
ast_timing_func_t ast_channel_timingfunc(const struct ast_channel *chan)
Definition channel_internal_api.c:1098
AST_CHANNEL_NAME
#define AST_CHANNEL_NAME
Definition channel.h:173
ast_channel_whentohangup
struct timeval * ast_channel_whentohangup(struct ast_channel *chan)
Definition channel_internal_api.c:795
ast_party_number_free
void ast_party_number_free(struct ast_party_number *doomed)
Destroy the party number contents.
Definition channel.c:1678
ast_party_connected_line_init
void ast_party_connected_line_init(struct ast_party_connected_line *init)
Initialize the given connected line structure.
Definition channel.c:2009
ast_channel_sendurl
int ast_channel_sendurl(struct ast_channel *channel, const char *url)
Sends a URL on a given link Send URL on link.
Definition channel.c:6692
ast_channel_language
const char * ast_channel_language(const struct ast_channel *chan)
ast_cause2str
const char * ast_cause2str(int cause) attribute_pure
Gives the string form of a given cause code.
Definition channel.c:613
ast_party_redirecting_free
void ast_party_redirecting_free(struct ast_party_redirecting *doomed)
Destroy the redirecting information contents.
Definition channel.c:2166
ast_channel_context_set
void ast_channel_context_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:223
ast_channel_sched
struct ast_sched_context * ast_channel_sched(const struct ast_channel *chan)
Definition channel_internal_api.c:596
ast_connected_line_copy_from_caller
void ast_connected_line_copy_from_caller(struct ast_party_connected_line *dest, const struct ast_party_caller *src)
Copy the caller information to the connected line information.
Definition channel.c:8399
AST_FLAG_OUTGOING
@ AST_FLAG_OUTGOING
Definition channel.h:1019
AST_FLAG_END_DTMF_ONLY
@ AST_FLAG_END_DTMF_ONLY
Definition channel.h:1027
AST_FLAG_MOH
@ AST_FLAG_MOH
Definition channel.h:1011
ast_channel_call_forward
const char * ast_channel_call_forward(const struct ast_channel *chan)
ast_channel_stream
struct ast_filestream * ast_channel_stream(const struct ast_channel *chan)
Definition channel_internal_api.c:488
ast_pre_call
int ast_pre_call(struct ast_channel *chan, const char *sub_args)
Execute a Gosub call on the channel before a call is placed.
Definition channel.c:6501
ast_channel_setoption
int ast_channel_setoption(struct ast_channel *channel, int option, void *data, int datalen, int block)
Sets an option on a channel.
Definition channel.c:7496
ast_channel_caller
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
Definition channel_internal_api.c:755
ast_party_connected_line_set_init
void ast_party_connected_line_set_init(struct ast_party_connected_line *init, const struct ast_party_connected_line *guide)
Initialize the given connected line structure using the given guide for a set update operation.
Definition channel.c:2032
ast_channel_transfercapability_set
void ast_channel_transfercapability_set(struct ast_channel *chan, unsigned short value)
Definition channel_internal_api.c:377
ast_channel_priority_set
void ast_channel_priority_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:320
ast_channel_hangupcause_set
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:302
ast_autoservice_chan_hangup_peer
void ast_autoservice_chan_hangup_peer(struct ast_channel *chan, struct ast_channel *peer)
Put chan into autoservice while hanging up peer.
Definition autoservice.c:349
ast_channel_whentohangup_set
void ast_channel_whentohangup_set(struct ast_channel *chan, struct timeval *value)
Definition channel_internal_api.c:835
ast_answer
int ast_answer(struct ast_channel *chan)
Answer a channel.
Definition channel.c:2839
ast_channel_adsicpe_set
void ast_channel_adsicpe_set(struct ast_channel *chan, enum ast_channel_adsicpe value)
Definition channel_internal_api.c:652
ast_channel_early_bridge
int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
Bridge two channels together (early)
Definition channel.c:7486
ast_indicate
int ast_indicate(struct ast_channel *chan, int condition)
Indicates condition of channel.
Definition channel.c:4332
ast_channel_exten
const char * ast_channel_exten(const struct ast_channel *chan)
Definition channel_internal_api.c:232
ast_channel_unlock
#define ast_channel_unlock(chan)
Definition channel.h:2990
AST_MAX_EXTENSION
#define AST_MAX_EXTENSION
Definition channel.h:134
ast_raw_answer
int ast_raw_answer(struct ast_channel *chan)
Answer a channel.
Definition channel.c:2724
ast_party_redirecting_copy
void ast_party_redirecting_copy(struct ast_party_redirecting *dest, const struct ast_party_redirecting *src)
Copy the source redirecting information to the destination redirecting.
Definition channel.c:2122
ast_channel_datastore_find
struct ast_datastore * ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
Find a datastore on a channel.
Definition channel.c:2390
ast_channel_state
ast_channel_state
ast_channel states
Definition channelstate.h:35
AST_STATE_UP
@ AST_STATE_UP
Definition channelstate.h:42
ast_datastore_alloc
#define ast_datastore_alloc(info, uid)
Definition datastore.h:85
ast_hangup_cause_to_dial_status
const char * ast_hangup_cause_to_dial_status(int hangup_cause)
Convert a hangup cause to a publishable dial status.
Definition dial.c:752
THRESHOLD_SILENCE
@ THRESHOLD_SILENCE
Definition dsp.h:75
ast_dsp_get_threshold_from_settings
int ast_dsp_get_threshold_from_settings(enum threshold which)
Get silence threshold from dsp.conf.
Definition dsp.c:2196
buf
char buf[BUFSIZE]
Definition eagi_proxy.c:66
ast_bridge_call
int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
Bridge a call, optionally allowing redirection.
Definition features.c:694
ast_bridge_timelimit
int ast_bridge_timelimit(struct ast_channel *chan, struct ast_bridge_config *config, char *parse, struct timeval *calldurationlimit)
parse L option and read associated channel variables to set warning, warning frequency,...
Definition features.c:866
ast_stopstream
int ast_stopstream(struct ast_channel *c)
Stops a stream.
Definition file.c:223
ast_streamfile
int ast_streamfile(struct ast_channel *c, const char *filename, const char *preflang)
Streams a file.
Definition file.c:1312
ast_fileexists
int ast_fileexists(const char *filename, const char *fmt, const char *preflang)
Checks for the existence of a given file.
Definition file.c:1148
ast_filedelete
int ast_filedelete(const char *filename, const char *fmt)
Deletes a file.
Definition file.c:1160
AST_DIGIT_ANY
#define AST_DIGIT_ANY
Definition file.h:48
ast_waitstream
int ast_waitstream(struct ast_channel *c, const char *breakon)
Waits for a stream to stop or digit to be pressed.
Definition file.c:1874
name
static const char name[]
Definition format_mp3.c:68
SCOPE_EXIT_RTN_VALUE
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition include/asterisk/logger.h:1036
SCOPE_ENTER
#define SCOPE_ENTER(level,...)
Definition include/asterisk/logger.h:1008
ast_trace
#define ast_trace(level,...)
Definition include/asterisk/logger.h:999
ast_channel_publish_dial
void ast_channel_publish_dial(struct ast_channel *caller, struct ast_channel *peer, const char *dialstring, const char *dialstatus)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition stasis_channels.c:767
ast_channel_stage_snapshot_done
void ast_channel_stage_snapshot_done(struct ast_channel *chan)
Clear flag to indicate channel snapshot is being staged, and publish snapshot.
Definition stasis_channels.c:1071
ast_channel_publish_snapshot
void ast_channel_publish_snapshot(struct ast_channel *chan)
Publish a ast_channel_snapshot for a channel.
Definition stasis_channels.c:1083
ast_channel_stage_snapshot
void ast_channel_stage_snapshot(struct ast_channel *chan)
Set flag to indicate channel snapshot is being staged.
Definition stasis_channels.c:1066
ast_channel_publish_dial_forward
void ast_channel_publish_dial_forward(struct ast_channel *caller, struct ast_channel *peer, struct ast_channel *forwarded, const char *dialstring, const char *dialstatus, const char *forward)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition stasis_channels.c:717
ast_app_expand_sub_args
const char * ast_app_expand_sub_args(struct ast_channel *chan, const char *args)
Add missing context/exten to subroutine argument string.
Definition main/app.c:278
AST_APP_ARG
#define AST_APP_ARG(name)
Define an application argument.
Definition include/asterisk/app.h:1230
ast_sf_stream
int ast_sf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
Send a string of SF digits to a channel.
Definition main/app.c:1097
ast_play_and_record
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path)
Record a file based on input from a channel. Use default accept and cancel DTMF. This function will p...
Definition main/app.c:2155
ast_app_group_set_channel
int ast_app_group_set_channel(struct ast_channel *chan, const char *data)
Set the group for a channel, splitting the provided data into group and category, if specified.
Definition main/app.c:2194
AST_DECLARE_APP_ARGS
#define AST_DECLARE_APP_ARGS(name, arglist)
Declare a structure to hold an application's arguments.
Definition include/asterisk/app.h:1247
ast_play_and_wait
int ast_play_and_wait(struct ast_channel *chan, const char *fn)
Play a stream and wait for a digit, returning the digit that was pressed.
Definition main/app.c:1617
AST_STANDARD_APP_ARGS
#define AST_STANDARD_APP_ARGS(args, parse)
Performs the 'standard' argument separation process for an application.
Definition include/asterisk/app.h:1283
ast_app_exec_sub
int ast_app_exec_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_args, int ignore_hangup)
Run a subroutine on a channel, placing an optional second channel into autoservice.
Definition main/app.c:297
ast_dtmf_stream
int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration)
Send a string of DTMF digits to a channel.
Definition main/app.c:1127
ast_mf_stream
int ast_mf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
Send a string of MF digits to a channel.
Definition main/app.c:1113
ast_app_parse_options64
int ast_app_parse_options64(const struct ast_app_option *options, struct ast_flags64 *flags, char **args, char *optstr)
Parses a string containing application options and sets flags/arguments.
Definition main/app.c:3072
AST_FEATURE_MAX_LEN
#define AST_FEATURE_MAX_LEN
Definition include/asterisk/features_config.h:231
ast_get_builtin_feature
int ast_get_builtin_feature(struct ast_channel *chan, const char *feature, char *buf, size_t len)
Get the DTMF code for a builtin feature.
Definition features_config.c:1305
AST_FRAME_DTMF
#define AST_FRAME_DTMF
Definition include/asterisk/frame.h:139
AST_OPTION_OPRMODE
#define AST_OPTION_OPRMODE
Definition include/asterisk/frame.h:463
ast_frfree
#define ast_frfree(fr)
Definition include/asterisk/frame.h:565
AST_FRAME_DTMF_END
@ AST_FRAME_DTMF_END
Definition include/asterisk/frame.h:99
AST_FRAME_DTMF_BEGIN
@ AST_FRAME_DTMF_BEGIN
Definition include/asterisk/frame.h:125
AST_FRAME_CONTROL
@ AST_FRAME_CONTROL
Definition include/asterisk/frame.h:105
AST_CONTROL_SRCUPDATE
@ AST_CONTROL_SRCUPDATE
Definition include/asterisk/frame.h:297
AST_CONTROL_PROGRESS
@ AST_CONTROL_PROGRESS
Definition include/asterisk/frame.h:291
AST_CONTROL_OFFHOOK
@ AST_CONTROL_OFFHOOK
Definition include/asterisk/frame.h:284
AST_CONTROL_UNHOLD
@ AST_CONTROL_UNHOLD
Definition include/asterisk/frame.h:294
AST_CONTROL_VIDUPDATE
@ AST_CONTROL_VIDUPDATE
Definition include/asterisk/frame.h:295
AST_CONTROL_PROCEEDING
@ AST_CONTROL_PROCEEDING
Definition include/asterisk/frame.h:292
AST_CONTROL_REDIRECTING
@ AST_CONTROL_REDIRECTING
Definition include/asterisk/frame.h:300
AST_CONTROL_CONGESTION
@ AST_CONTROL_CONGESTION
Definition include/asterisk/frame.h:285
AST_CONTROL_PLAYBACK_BEGIN
@ AST_CONTROL_PLAYBACK_BEGIN
Definition include/asterisk/frame.h:338
AST_CONTROL_ANSWER
@ AST_CONTROL_ANSWER
Definition include/asterisk/frame.h:281
AST_CONTROL_RINGING
@ AST_CONTROL_RINGING
Definition include/asterisk/frame.h:280
AST_CONTROL_HANGUP
@ AST_CONTROL_HANGUP
Definition include/asterisk/frame.h:278
AST_CONTROL_CONNECTED_LINE
@ AST_CONTROL_CONNECTED_LINE
Definition include/asterisk/frame.h:299
AST_CONTROL_FLASH
@ AST_CONTROL_FLASH
Definition include/asterisk/frame.h:286
AST_CONTROL_SRCCHANGE
@ AST_CONTROL_SRCCHANGE
Definition include/asterisk/frame.h:303
AST_CONTROL_PVT_CAUSE_CODE
@ AST_CONTROL_PVT_CAUSE_CODE
Definition include/asterisk/frame.h:310
ast_debug
#define ast_debug(level,...)
Log a DEBUG message.
Definition include/asterisk/logger.h:456
LOG_ERROR
#define LOG_ERROR
Definition include/asterisk/logger.h:291
ast_verb
#define ast_verb(level,...)
Definition include/asterisk/logger.h:467
LOG_NOTICE
#define LOG_NOTICE
Definition include/asterisk/logger.h:269
LOG_WARNING
#define LOG_WARNING
Definition include/asterisk/logger.h:280
ast_tone_zone_sound_unref
static struct ast_tone_zone_sound * ast_tone_zone_sound_unref(struct ast_tone_zone_sound *ts)
Release a reference to an ast_tone_zone_sound.
Definition indications.h:227
ast_playtones_start
int ast_playtones_start(struct ast_channel *chan, int vol, const char *tonelist, int interruptible)
Start playing a list of tones on a channel.
Definition indications.c:302
ast_get_indication_tone
struct ast_tone_zone_sound * ast_get_indication_tone(const struct ast_tone_zone *zone, const char *indication)
Locate a tone zone sound.
Definition indications.c:461
AST_LIST_HEAD_NOLOCK
#define AST_LIST_HEAD_NOLOCK(name, type)
Defines a structure to be used to hold a list of specified type (with no lock).
Definition linkedlists.h:225
AST_LIST_TRAVERSE
#define AST_LIST_TRAVERSE(head, var, field)
Loops over (traverses) the entries in a list.
Definition linkedlists.h:491
AST_LIST_EMPTY
#define AST_LIST_EMPTY(head)
Checks whether the specified list contains any entries.
Definition linkedlists.h:450
AST_LIST_INSERT_TAIL
#define AST_LIST_INSERT_TAIL(head, elm, field)
Appends a list entry to the tail of a list.
Definition linkedlists.h:731
AST_LIST_HEAD_NOLOCK_INIT_VALUE
#define AST_LIST_HEAD_NOLOCK_INIT_VALUE
Defines initial values for a declaration of AST_LIST_HEAD_NOLOCK.
Definition linkedlists.h:252
AST_LIST_ENTRY
#define AST_LIST_ENTRY(type)
Declare a forward link structure inside a list entry.
Definition linkedlists.h:410
AST_LIST_TRAVERSE_SAFE_END
#define AST_LIST_TRAVERSE_SAFE_END
Closes a safe loop traversal block.
Definition linkedlists.h:615
AST_LIST_TRAVERSE_SAFE_BEGIN
#define AST_LIST_TRAVERSE_SAFE_BEGIN(head, var, field)
Loops safely over (traverses) the entries in a list.
Definition linkedlists.h:529
AST_LIST_REMOVE_CURRENT
#define AST_LIST_REMOVE_CURRENT(field)
Removes the current entry from a list during a traversal.
Definition linkedlists.h:557
AST_LIST_REMOVE_HEAD
#define AST_LIST_REMOVE_HEAD(head, field)
Removes and returns the head entry from a list.
Definition linkedlists.h:833
AST_LIST_FIRST
#define AST_LIST_FIRST(head)
Returns the first entry contained in a list.
Definition linkedlists.h:421
AST_LIST_NEXT
#define AST_LIST_NEXT(elm, field)
Returns the next entry in the list after the given entry.
Definition linkedlists.h:439
ast_max_forwards_decrement
int ast_max_forwards_decrement(struct ast_channel *chan)
Decrement the max forwards count for a particular channel.
Definition max_forwards.c:135
ast_max_forwards_get
int ast_max_forwards_get(struct ast_channel *chan)
Get the current max forwards for a particular channel.
Definition max_forwards.c:121
AST_MODFLAG_DEFAULT
@ AST_MODFLAG_DEFAULT
Definition module.h:329
AST_MODULE_INFO
#define AST_MODULE_INFO(keystr, flags_to_set, desc, fields...)
Definition module.h:557
AST_MODULE_SUPPORT_CORE
@ AST_MODULE_SUPPORT_CORE
Definition module.h:121
ASTERISK_GPL_KEY
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition module.h:46
ast_unregister_application
int ast_unregister_application(const char *app)
Unregister an application.
Definition pbx_app.c:404
ast_register_application_xml
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition module.h:640
ast_moh_start
int ast_moh_start(struct ast_channel *chan, const char *mclass, const char *interpclass)
Turn on music on hold on a given channel.
Definition channel.c:7840
ast_moh_stop
void ast_moh_stop(struct ast_channel *chan)
Turn off music on hold on a given channel.
Definition channel.c:7850
ast_config_AST_DATA_DIR
const char * ast_config_AST_DATA_DIR
Definition options.c:159
pbx_builtin_getvar_helper
const char * pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name)
Return a pointer to the value of the corresponding channel variable.
Definition pbx_variables.c:1162
ast_exists_extension
int ast_exists_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid)
Determine whether an extension exists.
Definition pbx.c:4211
AST_PBX_INCOMPLETE
#define AST_PBX_INCOMPLETE
Definition pbx.h:51
pbx_builtin_setvar_helper
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name.
Definition pbx_variables.c:1233
ast_goto_if_exists
int ast_goto_if_exists(struct ast_channel *chan, const char *context, const char *exten, int priority)
Definition pbx.c:8825
ast_pbx_start
enum ast_pbx_result ast_pbx_start(struct ast_channel *c)
Create a new thread and start the PBX.
Definition pbx.c:4744
ast_get_hint
int ast_get_hint(char *hint, int hintsize, char *name, int namesize, struct ast_channel *c, const char *context, const char *exten)
If an extension hint exists, return non-zero.
Definition pbx.c:4173
ast_parseable_goto
int ast_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition pbx.c:8910
AST_PRIVACY_KILL
#define AST_PRIVACY_KILL
Definition privacy.h:32
AST_PRIVACY_ALLOW
#define AST_PRIVACY_ALLOW
Definition privacy.h:31
AST_PRIVACY_DENY
#define AST_PRIVACY_DENY
Definition privacy.h:30
ast_privacy_set
int ast_privacy_set(char *dest, char *cid, int status)
Definition privacy.c:82
ast_privacy_check
int ast_privacy_check(char *dest, char *cid)
Definition privacy.c:46
AST_PRIVACY_UNKNOWN
#define AST_PRIVACY_UNKNOWN
Definition privacy.h:34
AST_PRIVACY_TORTURE
#define AST_PRIVACY_TORTURE
Definition privacy.h:33
url
static char url[512]
Definition res_config_ldap.c:74
args
static struct @522 args
NULL
#define NULL
Definition resample.c:96
ast_rtp_instance_early_bridge_make_compatible
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
Make two channels compatible for early bridging.
Definition rtp_engine.c:2466
ast_sched_runq
int ast_sched_runq(struct ast_sched_context *con)
Runs the queue.
Definition sched.c:786
ast_sched_wait
int ast_sched_wait(struct ast_sched_context *con) attribute_warn_unused_result
Determines number of seconds until the next outstanding event to take place.
Definition sched.c:433
AST_STREAM_STATE_RECVONLY
@ AST_STREAM_STATE_RECVONLY
Set when the stream is receiving media only.
Definition stream.h:90
AST_STREAM_STATE_SENDONLY
@ AST_STREAM_STATE_SENDONLY
Set when the stream is sending media only.
Definition stream.h:86
ast_stream_set_state
void ast_stream_set_state(struct ast_stream *stream, enum ast_stream_state state)
Set the state of a stream.
Definition stream.c:380
ast_stream_topology_to_str
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition stream.c:939
ast_stream_topology_get_stream
struct ast_stream * ast_stream_topology_get_stream(const struct ast_stream_topology *topology, unsigned int position)
Get a specific stream from the topology.
Definition stream.c:791
ast_stream_topology_get_count
int ast_stream_topology_get_count(const struct ast_stream_topology *topology)
Get the number of streams in a topology.
Definition stream.c:768
ast_stream_get_state
enum ast_stream_state ast_stream_get_state(const struct ast_stream *stream)
Get the current state of a stream.
Definition stream.c:373
ast_stream_topology_free
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition stream.c:746
ast_stream_topology_clone
struct ast_stream_topology * ast_stream_topology_clone(const struct ast_stream_topology *topology)
Create a deep clone of an existing stream topology.
Definition stream.c:670
ast_str_append
int ast_str_append(struct ast_str **buf, ssize_t max_len, const char *fmt,...)
Append to a thread local dynamic string.
Definition strings.h:1139
ast_str_strlen
size_t attribute_pure ast_str_strlen(const struct ast_str *buf)
Returns the current length of the string stored within buf.
Definition strings.h:730
S_COR
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition strings.h:87
ast_strlen_zero
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition strings.h:65
ast_str_tmp
#define ast_str_tmp(init_len, __expr)
Provides a temporary ast_str and returns a copy of its buffer.
Definition strings.h:1189
ast_str_alloca
#define ast_str_alloca(init_len)
Definition strings.h:848
ast_str_reset
void ast_str_reset(struct ast_str *buf)
Reset the content of a dynamic string. Useful before a series of ast_str_append.
Definition strings.h:693
ast_str_buffer
char *attribute_pure ast_str_buffer(const struct ast_str *buf)
Returns the string buffer within the ast_str buf.
Definition strings.h:761
ast_copy_string
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition strings.h:425
ast_strip
char * ast_strip(char *s)
Strip leading/trailing whitespace from a string.
Definition strings.h:223
ast_aoc_decoded
Definition aoc.c:214
ast_aoc_encoded
Definition aoc.c:206
ast_bridge_config
bridge configuration
Definition channel.h:1096
ast_bridge_config::end_bridge_callback_data
void * end_bridge_callback_data
Definition channel.h:1111
ast_channel
Main Channel structure associated with a channel.
Definition channel_private.h:48
ast_channel::exten
char exten[AST_MAX_EXTENSION]
Definition channel_private.h:184
ast_channel::data
const char * data
Definition channel_private.h:58
ast_channel::tech
const struct ast_channel_tech * tech
Definition channel_private.h:49
ast_channel::context
char context[AST_MAX_CONTEXT]
Definition channel_private.h:183
ast_channel::flags
struct ast_flags flags
Definition channel_private.h:166
ast_datastore_info
Structure for a data store type.
Definition datastore.h:31
ast_datastore_info::type
const char * type
Definition datastore.h:32
ast_datastore
Structure for a data store object.
Definition datastore.h:64
ast_datastore::data
void * data
Definition datastore.h:66
ast_flags64
Structure used to handle a large number of boolean flags == used only in app_dial?
Definition utils.h:225
ast_flags64::flags
uint64_t flags
Definition utils.h:226
ast_frame_subclass::topology
struct ast_stream_topology * topology
Definition include/asterisk/frame.h:157
ast_frame_subclass::integer
int integer
Definition include/asterisk/frame.h:152
ast_frame
Data structure associated with a single frame of data.
Definition include/asterisk/frame.h:165
ast_frame::subclass
struct ast_frame_subclass subclass
Definition include/asterisk/frame.h:169
ast_frame::uint32
uint32_t uint32
Definition include/asterisk/frame.h:183
ast_frame::frametype
enum ast_frame_type frametype
Definition include/asterisk/frame.h:167
ast_frame::data
union ast_frame::@235 data
ast_frame::datalen
int datalen
Definition include/asterisk/frame.h:171
ast_frame::ptr
void * ptr
Definition include/asterisk/frame.h:183
ast_party_caller
Caller Party information.
Definition channel.h:420
ast_party_connected_line
Connected Line/Party information.
Definition channel.h:458
ast_party_connected_line::id
struct ast_party_id id
Connected party ID.
Definition channel.h:460
ast_party_dialed::transit_network_select
int transit_network_select
Transit Network Select.
Definition channel.h:399
ast_party_id
Information needed to identify an endpoint in a call.
Definition channel.h:340
ast_party_id::subaddress
struct ast_party_subaddress subaddress
Subscriber subaddress.
Definition channel.h:346
ast_party_id::tag
char * tag
User-set "tag".
Definition channel.h:356
ast_party_id::name
struct ast_party_name name
Subscriber name.
Definition channel.h:342
ast_party_id::number
struct ast_party_number number
Subscriber phone number.
Definition channel.h:344
ast_party_name::valid
unsigned char valid
TRUE if the name information is valid/present.
Definition channel.h:281
ast_party_name::str
char * str
Subscriber name (Malloced)
Definition channel.h:266
ast_party_number::presentation
int presentation
Q.931 presentation-indicator and screening-indicator encoded fields.
Definition channel.h:297
ast_party_number::valid
unsigned char valid
TRUE if the number information is valid/present.
Definition channel.h:299
ast_party_number::str
char * str
Subscriber phone number (Malloced)
Definition channel.h:293
ast_party_redirecting
Redirecting Line information. RDNIS (Redirecting Directory Number Information Service) Where a call d...
Definition channel.h:524
ast_party_redirecting::from
struct ast_party_id from
Who is redirecting the call (Sent to the party the call is redirected toward)
Definition channel.h:529
ast_party_redirecting::to
struct ast_party_id to
Call is redirecting to a new party (Sent to the caller)
Definition channel.h:532
ast_party_subaddress::str
char * str
Malloced subaddress string.
Definition channel.h:315
ast_str
Support for dynamic strings.
Definition strings.h:623
ast_stream_topology
Definition stream.c:123
ast_stream
Definition stream.c:81
ast_tone_zone_sound
Description of a tone.
Definition indications.h:35
ast_tone_zone_sound::data
const char * data
Description of a tone.
Definition indications.h:52
cause_args
Definition app_dial.c:870
cause_args::congestion
int congestion
Definition app_dial.c:873
cause_args::nochan
int nochan
Definition app_dial.c:874
cause_args::chan
struct ast_channel * chan
Definition app_dial.c:871
cause_args::busy
int busy
Definition app_dial.c:872
chanlist
List of channel drivers.
Definition app_dial.c:804
chanlist::number
const char * number
Definition app_dial.c:812
chanlist::interface
const char * interface
Definition app_dial.c:808
chanlist::aoc_s_rate_list
struct ast_aoc_decoded * aoc_s_rate_list
Definition app_dial.c:820
chanlist::connected
struct ast_party_connected_line connected
Definition app_dial.c:817
chanlist::orig_chan_name
char * orig_chan_name
Definition app_dial.c:814
chanlist::stuff
char stuff[0]
Definition app_dial.c:822
chanlist::chan
struct ast_channel * chan
Definition app_dial.c:806
chanlist::tech
const char * tech
Definition app_dial.c:810
chanlist::pending_connected_update
unsigned int pending_connected_update
Definition app_dial.c:819
chans
Definition astman.c:88
dial_head
Definition app_dial.c:825
extension
structure to hold extensions
Definition res_phoneprov.c:293
max_forwards
Channel datastore data for max forwards.
Definition max_forwards.c:29
number
Number structure.
Definition app_followme.c:157
oprmode::peer
struct ast_channel * peer
Definition include/asterisk/frame.h:527
privacy_args
Definition app_dial.c:1147
privacy_args::status
char status[256]
Definition app_dial.c:1152
privacy_args::privdb_val
int privdb_val
Definition app_dial.c:1149
privacy_args::canceled
int canceled
Definition app_dial.c:1153
privacy_args::privcid
char privcid[256]
Definition app_dial.c:1150
privacy_args::sentringing
int sentringing
Definition app_dial.c:1148
privacy_args::privintro
char privintro[1024]
Definition app_dial.c:1151
done
int done
Definition test_amihooks.c:48
options
static struct test_options options
Definition test_http_media_cache.c:63
c
static struct test_val c
Definition test_linkedlists.c:48
ast_tvzero
int ast_tvzero(const struct timeval t)
Returns true if the argument is 0,0.
Definition time.h:117
ast_remaining_ms
int ast_remaining_ms(struct timeval start, int max_ms)
Calculate remaining milliseconds given a starting timestamp and upper bound.
Definition utils.c:2315
ast_tvadd
struct timeval ast_tvadd(struct timeval a, struct timeval b)
Returns the sum of two timevals a + b.
Definition extconf.c:2280
ast_tvdiff_ms
int64_t ast_tvdiff_ms(struct timeval end, struct timeval start)
Computes the difference (in milliseconds) between two struct timeval instances.
Definition time.h:107
ast_tvnow
struct timeval ast_tvnow(void)
Returns current timeval. Meant to replace calls to gettimeofday().
Definition time.h:159
out
FILE * out
Definition utils/frame.c:33
in
FILE * in
Definition utils/frame.c:33
ast_test_flag
#define ast_test_flag(p, flag)
Definition utils.h:64
ast_set2_flag64
#define ast_set2_flag64(p, value, flag)
Definition utils.h:171
ast_test_flag64
#define ast_test_flag64(p, flag)
Definition utils.h:140
ast_clear_flag64
#define ast_clear_flag64(p, flag)
Definition utils.h:154
ast_clear_flag
#define ast_clear_flag(p, flag)
Definition utils.h:78
ast_mkdir
int ast_mkdir(const char *path, int mode)
Recursively create directory path.
Definition utils.c:2513
ast_copy_flags64
#define ast_copy_flags64(dest, src, flagz)
Definition utils.h:161
ast_set_flag64
#define ast_set_flag64(p, flag)
Definition utils.h:147
ast_set_flag
#define ast_set_flag(p, flag)
Definition utils.h:71
ast_replace_subargument_delimiter
void ast_replace_subargument_delimiter(char *s)
Replace '^' in a string with ','.
Definition utils.c:2377

◆ DIAL_CALLERID_ABSENT

#define DIAL_CALLERID_ABSENT   (1LLU << 33) /* TRUE if caller id is not available for connected line. */

Definition at line 709 of file app_dial.c.

◆ DIAL_NOFORWARDHTML

#define DIAL_NOFORWARDHTML   (1LLU << 32)

Definition at line 708 of file app_dial.c.

◆ DIAL_STILLGOING

#define DIAL_STILLGOING   (1LLU << 31)

Definition at line 707 of file app_dial.c.

◆ OPT_CALLEE_GO_ON

#define OPT_CALLEE_GO_ON   (1LLU << 36)

Definition at line 712 of file app_dial.c.

◆ OPT_CALLER_ANSWER

#define OPT_CALLER_ANSWER   (1LLU << 40)

Definition at line 716 of file app_dial.c.

◆ OPT_CANCEL_ELSEWHERE

#define OPT_CANCEL_ELSEWHERE   (1LLU << 34)

Definition at line 710 of file app_dial.c.

◆ OPT_CANCEL_TIMEOUT

#define OPT_CANCEL_TIMEOUT   (1LLU << 37)

Definition at line 713 of file app_dial.c.

◆ OPT_FORCE_CID_PRES

#define OPT_FORCE_CID_PRES   (1LLU << 39)

Definition at line 715 of file app_dial.c.

◆ OPT_FORCE_CID_TAG

#define OPT_FORCE_CID_TAG   (1LLU << 38)

Definition at line 714 of file app_dial.c.

◆ OPT_HANGUPCAUSE

#define OPT_HANGUPCAUSE   (1LLU << 44)

Definition at line 720 of file app_dial.c.

◆ OPT_HEARPULSING

#define OPT_HEARPULSING   (1LLU << 45)

Definition at line 721 of file app_dial.c.

◆ OPT_PEER_H

#define OPT_PEER_H   (1LLU << 35)

Definition at line 711 of file app_dial.c.

◆ OPT_PREDIAL_CALLEE

#define OPT_PREDIAL_CALLEE   (1LLU << 41)

Definition at line 717 of file app_dial.c.

◆ OPT_PREDIAL_CALLER

#define OPT_PREDIAL_CALLER   (1LLU << 42)

Definition at line 718 of file app_dial.c.

◆ OPT_RING_WITH_EARLY_MEDIA

#define OPT_RING_WITH_EARLY_MEDIA   (1LLU << 43)

Definition at line 719 of file app_dial.c.

◆ OPT_TOPOLOGY_PRESERVE

#define OPT_TOPOLOGY_PRESERVE   (1LLU << 46)

Definition at line 722 of file app_dial.c.

Enumeration Type Documentation

◆ anonymous enum

anonymous enum
Enumerator
OPT_ANNOUNCE 
OPT_RESETCDR 
OPT_DTMF_EXIT 
OPT_SENDDTMF 
OPT_FORCECLID 
OPT_GO_ON 
OPT_CALLEE_HANGUP 
OPT_CALLER_HANGUP 
OPT_ORIGINAL_CLID 
OPT_DURATION_LIMIT 
OPT_MUSICBACK 
OPT_SCREEN_NOINTRO 
OPT_SCREEN_NOCALLERID 
OPT_IGNORE_CONNECTEDLINE 
OPT_SCREENING 
OPT_PRIVACY 
OPT_RINGBACK 
OPT_DURATION_STOP 
OPT_CALLEE_TRANSFER 
OPT_CALLER_TRANSFER 
OPT_CALLEE_MONITOR 
OPT_CALLER_MONITOR 
OPT_GOTO 
OPT_OPERMODE 
OPT_CALLEE_PARK 
OPT_CALLER_PARK 
OPT_IGNORE_FORWARDING 
OPT_CALLEE_GOSUB 
OPT_CALLEE_MIXMONITOR 
OPT_CALLER_MIXMONITOR 

Definition at line 673 of file app_dial.c.

673 {
674 OPT_ANNOUNCE = (1 << 0),
675 OPT_RESETCDR = (1 << 1),
676 OPT_DTMF_EXIT = (1 << 2),
677 OPT_SENDDTMF = (1 << 3),
678 OPT_FORCECLID = (1 << 4),
679 OPT_GO_ON = (1 << 5),
680 OPT_CALLEE_HANGUP = (1 << 6),
681 OPT_CALLER_HANGUP = (1 << 7),
682 OPT_ORIGINAL_CLID = (1 << 8),
683 OPT_DURATION_LIMIT = (1 << 9),
684 OPT_MUSICBACK = (1 << 10),
685 OPT_SCREEN_NOINTRO = (1 << 12),
686 OPT_SCREEN_NOCALLERID = (1 << 13),
687 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
688 OPT_SCREENING = (1 << 15),
689 OPT_PRIVACY = (1 << 16),
690 OPT_RINGBACK = (1 << 17),
691 OPT_DURATION_STOP = (1 << 18),
692 OPT_CALLEE_TRANSFER = (1 << 19),
693 OPT_CALLER_TRANSFER = (1 << 20),
694 OPT_CALLEE_MONITOR = (1 << 21),
695 OPT_CALLER_MONITOR = (1 << 22),
696 OPT_GOTO = (1 << 23),
697 OPT_OPERMODE = (1 << 24),
698 OPT_CALLEE_PARK = (1 << 25),
699 OPT_CALLER_PARK = (1 << 26),
700 OPT_IGNORE_FORWARDING = (1 << 27),
701 OPT_CALLEE_GOSUB = (1 << 28),
702 OPT_CALLEE_MIXMONITOR = (1 << 29),
703 OPT_CALLER_MIXMONITOR = (1 << 30),
704};

◆ anonymous enum

anonymous enum
Enumerator
OPT_ARG_ANNOUNCE 
OPT_ARG_SENDDTMF 
OPT_ARG_GOTO 
OPT_ARG_DURATION_LIMIT 
OPT_ARG_MUSICBACK 
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_HANGUPCAUSE 
OPT_ARG_ARRAY_SIZE 

Definition at line 724 of file app_dial.c.

724 {
725 OPT_ARG_ANNOUNCE = 0,
726 OPT_ARG_SENDDTMF,
727 OPT_ARG_GOTO,
728 OPT_ARG_DURATION_LIMIT,
729 OPT_ARG_MUSICBACK,
730 OPT_ARG_RINGBACK,
731 OPT_ARG_CALLEE_GOSUB,
732 OPT_ARG_CALLEE_GO_ON,
733 OPT_ARG_PRIVACY,
734 OPT_ARG_DURATION_STOP,
735 OPT_ARG_OPERMODE,
736 OPT_ARG_SCREEN_NOINTRO,
737 OPT_ARG_ORIGINAL_CLID,
738 OPT_ARG_FORCECLID,
739 OPT_ARG_FORCE_CID_TAG,
740 OPT_ARG_FORCE_CID_PRES,
741 OPT_ARG_PREDIAL_CALLEE,
742 OPT_ARG_PREDIAL_CALLER,
743 OPT_ARG_HANGUPCAUSE,
744 /* note: this entry _MUST_ be the last one in the enum */
745 OPT_ARG_ARRAY_SIZE
746};

Function Documentation

◆ __reg_module()

static void __reg_module ( void  )
static

Definition at line 3637 of file app_dial.c.

◆ __unreg_module()

static void __unreg_module ( void  )
static

Definition at line 3637 of file app_dial.c.

◆ AST_MODULE_SELF_SYM()

struct ast_module * AST_MODULE_SELF_SYM ( void  )

Definition at line 3637 of file app_dial.c.

◆ chanlist_free()

static void chanlist_free ( struct chanlist *  outgoing)
static

Definition at line 839 of file app_dial.c.

840{
841 ast_party_connected_line_free(&outgoing->connected);
842 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
843 ast_free(outgoing->orig_chan_name);
844 ast_free(outgoing);
845}

References ast_aoc_destroy_decoded(), ast_free, and ast_party_connected_line_free().

Referenced by dial_exec_full(), and hanguptree().

◆ detect_disconnect()

static int detect_disconnect ( struct ast_channel *  chan,
char  code,
struct ast_str **  featurecode 
)
static

Definition at line 1979 of file app_dial.c.

1980{
1981 char disconnect_code[AST_FEATURE_MAX_LEN];
1982 int res;
1983
1984 ast_str_append(featurecode, 1, "%c", code);
1985
1986 res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1987 if (res) {
1988 ast_str_reset(*featurecode);
1989 return 0;
1990 }
1991
1992 if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1993 /* Could be a partial match, anyway */
1994 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1995 ast_str_reset(*featurecode);
1996 }
1997 return 0;
1998 }
1999
2000 if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
2001 ast_str_reset(*featurecode);
2002 return 0;
2003 }
2004
2005 return 1;
2006}

References AST_FEATURE_MAX_LEN, ast_get_builtin_feature(), ast_str_append(), ast_str_buffer(), ast_str_reset(), and ast_str_strlen().

Referenced by wait_for_answer().

◆ dial_exec()

static int dial_exec ( struct ast_channel *  chan,
const char *  data 
)
static

Definition at line 3496 of file app_dial.c.

3497{
3498 struct ast_flags64 peerflags;
3499
3500 memset(&peerflags, 0, sizeof(peerflags));
3501
3502 return dial_exec_full(chan, data, &peerflags, NULL);
3503}

References dial_exec_full(), and NULL.

Referenced by load_module().

◆ dial_exec_full()

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.

Note
This will not have any malloced strings so do not free it.

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.
This will not have any malloced strings so do not free it.

CallerID party information to store.

Note
This will not have any malloced strings so do not free it.

Definition at line 2333 of file app_dial.c.

2334{
2335 int res = -1; /* default: error */
2336 char *rest, *cur; /* scan the list of destinations */
2337 struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2338 struct chanlist *outgoing;
2339 struct chanlist *tmp;
2340 struct ast_channel *peer = NULL;
2341 int to_answer, to_progress; /* timeouts */
2342 struct cause_args num = { chan, 0, 0, 0 };
2343 int cause, hanguptreecause = -1;
2344
2345 struct ast_bridge_config config = { { 0, } };
2346 struct timeval calldurationlimit = { 0, };
2347 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
2348 char *mf_progress = NULL, *mf_wink = NULL;
2349 char *sf_progress = NULL, *sf_wink = NULL;
2350 struct privacy_args pa = {
2351 .sentringing = 0,
2352 .privdb_val = 0,
2353 .status = "INVALIDARGS",
2354 .canceled = 0,
2355 };
2356 int sentringing = 0, moh = 0;
2357 const char *outbound_group = NULL;
2358 int result = 0;
2359 char *parse;
2360 int opermode = 0;
2361 int delprivintro = 0;
2362 AST_DECLARE_APP_ARGS(args,
2363 AST_APP_ARG(peers);
2364 AST_APP_ARG(timeout);
2365 AST_APP_ARG(options);
2366 AST_APP_ARG(url);
2367 );
2368 struct ast_flags64 opts = { 0, };
2369 char *opt_args[OPT_ARG_ARRAY_SIZE];
2370 int fulldial = 0, num_dialed = 0;
2371 int ignore_cc = 0;
2372 char device_name[AST_CHANNEL_NAME];
2373 char forced_clid_name[AST_MAX_EXTENSION];
2374 char stored_clid_name[AST_MAX_EXTENSION];
2375 int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2376 /*!
2377 * \brief Forced CallerID party information to send.
2378 * \note This will not have any malloced strings so do not free it.
2379 */
2380 struct ast_party_id forced_clid;
2381 /*!
2382 * \brief Stored CallerID information if needed.
2383 *
2384 * \note If OPT_ORIGINAL_CLID set then this is the o option
2385 * CallerID. Otherwise it is the dialplan extension and hint
2386 * name.
2387 *
2388 * \note This will not have any malloced strings so do not free it.
2389 */
2390 struct ast_party_id stored_clid;
2391 /*!
2392 * \brief CallerID party information to store.
2393 * \note This will not have any malloced strings so do not free it.
2394 */
2395 struct ast_party_caller caller;
2396 int max_forwards;
2397 struct ast_datastore *topology_ds = NULL;
2398 SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
2399
2400 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2401 ast_channel_lock(chan);
2402 ast_channel_stage_snapshot(chan);
2403 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2404 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2405 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2406 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2407 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME_MS", "");
2408 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2409 pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
2410 pbx_builtin_setvar_helper(chan, "RINGTIME", "");
2411 pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
2412 pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
2413 pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
2414 ast_channel_stage_snapshot_done(chan);
2415 max_forwards = ast_max_forwards_get(chan);
2416 ast_channel_unlock(chan);
2417
2418 if (max_forwards <= 0) {
2419 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2420 ast_channel_name(chan));
2421 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2422 SCOPE_EXIT_RTN_VALUE(-1, "%s: Max forwards exceeded\n", ast_channel_name(chan));
2423 }
2424
2425 if (ast_check_hangup_locked(chan)) {
2426 /*
2427 * Caller hung up before we could dial. If dial is executed
2428 * within an AGI then the AGI has likely eaten all queued
2429 * frames before executing the dial in DeadAGI mode. With
2430 * the caller hung up and no pending frames from the caller's
2431 * read queue, dial would not know that the call has hung up
2432 * until a called channel answers. It is rather annoying to
2433 * whoever just answered the non-existent call.
2434 *
2435 * Dial should not continue execution in DeadAGI mode, hangup
2436 * handlers, or the h exten.
2437 */
2438 ast_verb(3, "Caller hung up before dial.\n");
2439 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2440 SCOPE_EXIT_RTN_VALUE(-1, "%s: Caller hung up before dial\n", ast_channel_name(chan));
2441 }
2442
2443 parse = ast_strdupa(data ?: "");
2444
2445 AST_STANDARD_APP_ARGS(args, parse);
2446
2447 if (!ast_strlen_zero(args.options) &&
2448 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2449 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2450 goto done;
2451 }
2452
2453 if (ast_cc_call_init(chan, &ignore_cc)) {
2454 goto done;
2455 }
2456
2457 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2458 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2459
2460 if (delprivintro < 0 || delprivintro > 1) {
2461 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2462 delprivintro = 0;
2463 }
2464 }
2465
2466 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2467 opt_args[OPT_ARG_RINGBACK] = NULL;
2468 }
2469
2470 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2471 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2472 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2473 }
2474
2475 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2476 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2477 if (!calldurationlimit.tv_sec) {
2478 ast_log(LOG_WARNING, "Dial does not accept S(%s)\n", opt_args[OPT_ARG_DURATION_STOP]);
2479 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2480 goto done;
2481 }
2482 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2483 }
2484
2485 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2486 sf_wink = opt_args[OPT_ARG_SENDDTMF];
2487 dtmfcalled = strsep(&sf_wink, ":");
2488 dtmfcalling = strsep(&sf_wink, ":");
2489 dtmf_progress = strsep(&sf_wink, ":");
2490 mf_progress = strsep(&sf_wink, ":");
2491 mf_wink = strsep(&sf_wink, ":");
2492 sf_progress = strsep(&sf_wink, ":");
2493 }
2494
2495 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2496 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2497 goto done;
2498 }
2499
2500 /* Setup the forced CallerID information to send if used. */
2501 ast_party_id_init(&forced_clid);
2502 force_forwards_only = 0;
2503 if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2504 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2505 ast_channel_lock(chan);
2506 forced_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2507 ast_channel_unlock(chan);
2508 forced_clid_name[0] = '\0';
2509 forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2510 sizeof(forced_clid_name), chan);
2511 force_forwards_only = 1;
2512 } else {
2513 /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2514 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2515 &forced_clid.number.str);
2516 }
2517 if (!ast_strlen_zero(forced_clid.name.str)) {
2518 forced_clid.name.valid = 1;
2519 }
2520 if (!ast_strlen_zero(forced_clid.number.str)) {
2521 forced_clid.number.valid = 1;
2522 }
2523 }
2524 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2525 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2526 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2527 }
2528 forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2529 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2530 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2531 int pres;
2532
2533 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2534 if (0 <= pres) {
2535 forced_clid.number.presentation = pres;
2536 }
2537 }
2538
2539 /* Setup the stored CallerID information if needed. */
2540 ast_party_id_init(&stored_clid);
2541 if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2542 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2543 ast_channel_lock(chan);
2544 ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2545 if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2546 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2547 }
2548 if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2549 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2550 }
2551 if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2552 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2553 }
2554 if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2555 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2556 }
2557 ast_channel_unlock(chan);
2558 } else {
2559 /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2560 ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2561 &stored_clid.number.str);
2562 if (!ast_strlen_zero(stored_clid.name.str)) {
2563 stored_clid.name.valid = 1;
2564 }
2565 if (!ast_strlen_zero(stored_clid.number.str)) {
2566 stored_clid.number.valid = 1;
2567 }
2568 }
2569 } else {
2570 /*
2571 * In case the new channel has no preset CallerID number by the
2572 * channel driver, setup the dialplan extension and hint name.
2573 */
2574 stored_clid_name[0] = '\0';
2575 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2576 sizeof(stored_clid_name), chan);
2577 if (ast_strlen_zero(stored_clid.name.str)) {
2578 stored_clid.name.str = NULL;
2579 } else {
2580 stored_clid.name.valid = 1;
2581 }
2582 ast_channel_lock(chan);
2583 stored_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2584 stored_clid.number.valid = 1;
2585 ast_channel_unlock(chan);
2586 }
2587
2588 if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2589 ast_cdr_reset(ast_channel_name(chan), 0);
2590 }
2591 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2592 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2593
2594 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2595 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2596 if (res <= 0)
2597 goto out;
2598 res = -1; /* reset default */
2599 }
2600
2601 if (continue_exec)
2602 *continue_exec = 0;
2603
2604 /* If a channel group has been specified, get it for use when we create peer channels */
2605
2606 ast_channel_lock(chan);
2607 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2608 outbound_group = ast_strdupa(outbound_group);
2609 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2610 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2611 outbound_group = ast_strdupa(outbound_group);
2612 }
2613 ast_channel_unlock(chan);
2614
2615 /* Set per dial instance flags. These flags are also passed back to RetryDial. */
2616 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2617 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2618 | OPT_ANNOUNCE | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2619
2620 /* PREDIAL: Run gosub on the caller's channel */
2621 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2622 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2623 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2624 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2625 }
2626
2627 /* loop through the list of dial destinations */
2628 rest = args.peers;
2629 while ((cur = strsep(&rest, "&"))) {
2630 struct ast_channel *tc; /* channel for this destination */
2631 char *number;
2632 char *tech;
2633 int i;
2634 size_t tech_len;
2635 size_t number_len;
2636 struct ast_stream_topology *topology;
2637 struct ast_stream *stream;
2638
2639 cur = ast_strip(cur);
2640 if (ast_strlen_zero(cur)) {
2641 /* No tech/resource in this position. */
2642 continue;
2643 }
2644
2645 /* Get a technology/resource pair */
2646 number = cur;
2647 tech = strsep(&number, "/");
2648
2649 num_dialed++;
2650 if (ast_strlen_zero(number)) {
2651 ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2652 goto out;
2653 }
2654
2655 tech_len = strlen(tech) + 1;
2656 number_len = strlen(number) + 1;
2657 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2658 if (!tmp) {
2659 goto out;
2660 }
2661
2662 /* Save tech, number, and interface. */
2663 cur = tmp->stuff;
2664 strcpy(cur, tech);
2665 tmp->tech = cur;
2666 cur += tech_len;
2667 strcpy(cur, tech);
2668 cur[tech_len - 1] = '/';
2669 tmp->interface = cur;
2670 cur += tech_len;
2671 strcpy(cur, number);
2672 tmp->number = cur;
2673
2674 if (opts.flags) {
2675 /* Set per outgoing call leg options. */
2676 ast_copy_flags64(tmp, &opts,
2677 OPT_CANCEL_ELSEWHERE |
2678 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2679 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2680 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2681 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2682 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2683 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2684 OPT_RING_WITH_EARLY_MEDIA);
2685 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2686 }
2687
2688 /* Request the peer */
2689
2690 ast_channel_lock(chan);
2691 /*
2692 * Seed the chanlist's connected line information with previously
2693 * acquired connected line info from the incoming channel. The
2694 * previously acquired connected line info could have been set
2695 * through the CONNECTED_LINE dialplan function.
2696 */
2697 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2698
2699 if (ast_test_flag64(&opts, OPT_TOPOLOGY_PRESERVE)) {
2700 topology_ds = ast_channel_datastore_find(chan, &topology_ds_info, NULL);
2701
2702 if (!topology_ds && (topology_ds = ast_datastore_alloc(&topology_ds_info, NULL))) {
2703 topology_ds->data = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2704 ast_channel_datastore_add(chan, topology_ds);
2705 }
2706 }
2707
2708 if (topology_ds) {
2709 ao2_ref(topology_ds->data, +1);
2710 topology = topology_ds->data;
2711 } else {
2712 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2713 }
2714
2715 ast_channel_unlock(chan);
2716
2717 for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
2718 stream = ast_stream_topology_get_stream(topology, i);
2719 /* For both recvonly and sendonly the stream state reflects our state, that is we
2720 * are receiving only and we are sending only. Since we are requesting a
2721 * channel for the peer, we need to swap this to reflect what we will be doing.
2722 * That is, if we are receiving from Alice then we want to be sending to Bob,
2723 * so swap recvonly to sendonly and vice versa.
2724 */
2725 if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
2726 ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
2727 } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
2728 ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
2729 }
2730 }
2731
2732 tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
2733
2734 ast_stream_topology_free(topology);
2735
2736 if (!tc) {
2737 /* If we can't, just go on to the next call */
2738 /* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
2739 ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
2740 tmp->tech, cause, ast_cause2str(cause));
2741 handle_cause(cause, &num);
2742 if (!rest) {
2743 /* we are on the last destination */
2744 ast_channel_hangupcause_set(chan, cause);
2745 }
2746 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2747 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2748 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2749 }
2750 }
2751 chanlist_free(tmp);
2752 continue;
2753 }
2754
2755 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2756 if (!ignore_cc) {
2757 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2758 }
2759
2760 ast_channel_lock_both(tc, chan);
2761 ast_channel_stage_snapshot(tc);
2762
2763 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2764
2765 /* Setup outgoing SDP to match incoming one */
2766 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2767 /* We are on the only destination. */
2768 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2769 }
2770
2771 /* Inherit specially named variables from parent channel */
2772 ast_channel_inherit_variables(chan, tc);
2773 ast_channel_datastore_inherit(chan, tc);
2774 ast_max_forwards_decrement(tc);
2775
2776 ast_channel_appl_set(tc, "AppDial");
2777 ast_channel_data_set(tc, "(Outgoing Line)");
2778
2779 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2780
2781 /* Determine CallerID to store in outgoing channel. */
2782 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2783 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2784 caller.id = stored_clid;
2785 ast_channel_set_caller_event(tc, &caller, NULL);
2786 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2787 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2788 ast_channel_caller(tc)->id.number.str, NULL))) {
2789 /*
2790 * The new channel has no preset CallerID number by the channel
2791 * driver. Use the dialplan extension and hint name.
2792 */
2793 caller.id = stored_clid;
2794 if (!caller.id.name.valid
2795 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2796 ast_channel_connected(chan)->id.name.str, NULL))) {
2797 /*
2798 * No hint name available. We have a connected name supplied by
2799 * the dialplan we can use instead.
2800 */
2801 caller.id.name.valid = 1;
2802 caller.id.name = ast_channel_connected(chan)->id.name;
2803 }
2804 ast_channel_set_caller_event(tc, &caller, NULL);
2805 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2806 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2807 NULL))) {
2808 /* The new channel has no preset CallerID name by the channel driver. */
2809 if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2810 ast_channel_connected(chan)->id.name.str, NULL))) {
2811 /*
2812 * We have a connected name supplied by the dialplan we can
2813 * use instead.
2814 */
2815 caller.id.name.valid = 1;
2816 caller.id.name = ast_channel_connected(chan)->id.name;
2817 ast_channel_set_caller_event(tc, &caller, NULL);
2818 }
2819 }
2820
2821 /* Determine CallerID for outgoing channel to send. */
2822 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2823 struct ast_party_connected_line connected;
2824
2825 ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2826 connected.id = forced_clid;
2827 ast_channel_set_connected_line(tc, &connected, NULL);
2828 } else {
2829 ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2830 }
2831
2832 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2833
2834 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2835
2836 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2837 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2838 ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
2839 }
2840
2841 /* Pass ADSI CPE and transfer capability */
2842 ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
2843 ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
2844
2845 /* If we have an outbound group, set this peer channel to it */
2846 if (outbound_group)
2847 ast_app_group_set_channel(tc, outbound_group);
2848 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2849 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
2850 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2851
2852 /* Check if we're forced by configuration */
2853 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2854 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2855
2856
2857 /* Inherit context and extension */
2858 ast_channel_dialcontext_set(tc, ast_channel_context(chan));
2859 ast_channel_exten_set(tc, ast_channel_exten(chan));
2860
2861 ast_channel_stage_snapshot_done(tc);
2862
2863 /* Save the original channel name to detect call pickup masquerading in. */
2864 tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));
2865
2866 ast_channel_unlock(tc);
2867 ast_channel_unlock(chan);
2868
2869 /* Put channel in the list of outgoing thingies. */
2870 tmp->chan = tc;
2871 AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
2872 }
2873
2874 /* As long as we attempted to dial valid peers, don't throw a warning. */
2875 /* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
2876 if (!num_dialed) {
2877 ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
2878 if (continue_exec) {
2879 /* There is no point in having RetryDial try again */
2880 *continue_exec = 1;
2881 }
2882 strcpy(pa.status, "CHANUNAVAIL");
2883 res = 0;
2884 goto out;
2885 }
2886
2887 /*
2888 * PREDIAL: Run gosub on all of the callee channels
2889 *
2890 * We run the callee predial before ast_call() in case the user
2891 * wishes to do something on the newly created channels before
2892 * the channel does anything important.
2893 *
2894 * Inside the target gosub we will be able to do something with
2895 * the newly created channel name ie: now the calling channel
2896 * can know what channel will be used to call the destination
2897 * ex: now we will know that SIP/abc-123 is calling SIP/def-124
2898 */
2899 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
2900 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
2901 && !AST_LIST_EMPTY(&out_chans)) {
2902 const char *predial_callee;
2903
2904 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
2905 predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
2906 if (predial_callee) {
2907 ast_autoservice_start(chan);
2908 AST_LIST_TRAVERSE(&out_chans, tmp, node) {
2909 ast_pre_call(tmp->chan, predial_callee);
2910 }
2911 ast_autoservice_stop(chan);
2912 ast_free((char *) predial_callee);
2913 }
2914 }
2915
2916 /* Start all outgoing calls */
2917 AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
2918 res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
2919 ast_channel_lock(chan);
2920
2921 /* check the results of ast_call */
2922 if (res) {
2923 /* Again, keep going even if there's an error */
2924 ast_debug(1, "ast call on peer returned %d\n", res);
2925 ast_verb(3, "Couldn't call %s\n", tmp->interface);
2926 if (ast_channel_hangupcause(tmp->chan)) {
2927 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
2928 }
2929 ast_channel_unlock(chan);
2930 ast_cc_call_failed(chan, tmp->chan, tmp->interface);
2931 ast_hangup(tmp->chan);
2932 tmp->chan = NULL;
2933 AST_LIST_REMOVE_CURRENT(node);
2934 chanlist_free(tmp);
2935 continue;
2936 }
2937
2938 ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
2939 ast_channel_unlock(chan);
2940
2941 ast_verb(3, "Called %s\n", tmp->interface);
2942 ast_set_flag64(tmp, DIAL_STILLGOING);
2943
2944 /* If this line is up, don't try anybody else */
2945 if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
2946 break;
2947 }
2948 }
2949 AST_LIST_TRAVERSE_SAFE_END;
2950
2951 if (ast_strlen_zero(args.timeout)) {
2952 to_answer = -1;
2953 to_progress = -1;
2954 } else {
2955 double tmp;
2956 char *anstimeout = strsep(&args.timeout, "^");
2957 if (!ast_strlen_zero(anstimeout)) {
2958 if (sscanf(anstimeout, "%30lf", &tmp) == 1 && tmp > 0) {
2959 to_answer = tmp * 1000;
2960 ast_debug(3, "Dial timeout set to %d ms\n", to_answer);
2961 } else {
2962 ast_log(LOG_WARNING, "Invalid answer timeout specified: '%s'. Setting timeout to infinite\n", anstimeout);
2963 to_answer = -1;
2964 }
2965 } else {
2966 to_answer = -1;
2967 }
2968 if (!ast_strlen_zero(args.timeout)) {
2969 if (sscanf(args.timeout, "%30lf", &tmp) == 1 && tmp > 0) {
2970 to_progress = tmp * 1000;
2971 ast_debug(3, "Dial progress timeout set to %d ms\n", to_progress);
2972 } else {
2973 ast_log(LOG_WARNING, "Invalid progress timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2974 to_progress = -1;
2975 }
2976 } else {
2977 to_progress = -1;
2978 }
2979 }
2980
2981 outgoing = AST_LIST_FIRST(&out_chans);
2982 if (!outgoing) {
2983 strcpy(pa.status, "CHANUNAVAIL");
2984 if (fulldial == num_dialed) {
2985 res = -1;
2986 goto out;
2987 }
2988 } else {
2989 /* Our status will at least be NOANSWER */
2990 strcpy(pa.status, "NOANSWER");
2991 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2992 moh = 1;
2993 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2994 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2995 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2996 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2997 ast_channel_musicclass_set(chan, original_moh);
2998 } else {
2999 ast_moh_start(chan, NULL, NULL);
3000 }
3001 ast_indicate(chan, AST_CONTROL_PROGRESS);
3002 } else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
3003 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
3004 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
3005 ast_indicate(chan, AST_CONTROL_RINGING);
3006 sentringing++;
3007 } else {
3008 ast_indicate(chan, AST_CONTROL_PROGRESS);
3009 }
3010 } else {
3011 ast_indicate(chan, AST_CONTROL_RINGING);
3012 sentringing++;
3013 }
3014 }
3015 }
3016
3017 peer = wait_for_answer(chan, &out_chans, &to_answer, &to_progress, peerflags, opt_args, &pa, &num, &result,
3018 dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
3019 (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
3020 ignore_cc, &forced_clid, &stored_clid, &config);
3021
3022 if (!peer) {
3023 if (result) {
3024 res = result; /* User entered a DTMF digit that matched a context */
3025 } else if (to_answer) { /* Musta gotten hung up */
3026 /* This does not necessarily mean that we dialed without a timeout.
3027 * to_answer is (ab)used by wait_for_answer to to indicate whether or we should continue in the dialplan or exit. */
3028 res = -1;
3029 } else { /* Nobody answered, next please? */
3030 res = 0;
3031 }
3032 } else {
3033 const char *number;
3034 const char *name;
3035 int dial_end_raised = 0;
3036 int cause = -1;
3037
3038 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
3039 ast_answer(chan);
3040 }
3041
3042 /* Ah ha! Someone answered within the desired timeframe. Of course after this
3043 we will always return with -1 so that it is hung up properly after the
3044 conversation. */
3045
3046 if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
3047 && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
3048 cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
3049 if (cause <= 0) {
3050 if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
3051 cause = 0;
3052 } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
3053 || cause < 0) {
3054 ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
3055 opt_args[OPT_ARG_HANGUPCAUSE]);
3056 cause = -1;
3057 }
3058 }
3059 }
3060 hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
3061
3062 /* If appropriate, log that we have a destination channel and set the answer time */
3063
3064 ast_channel_lock(peer);
3065 name = ast_strdupa(ast_channel_name(peer));
3066
3067 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
3068 if (ast_strlen_zero(number)) {
3069 number = NULL;
3070 } else {
3071 number = ast_strdupa(number);
3072 }
3073 ast_channel_unlock(peer);
3074
3075 ast_channel_lock(chan);
3076 ast_channel_stage_snapshot(chan);
3077
3078 strcpy(pa.status, "ANSWER");
3079 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3080
3081 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
3082 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
3083
3084 ast_channel_stage_snapshot_done(chan);
3085 ast_channel_unlock(chan);
3086
3087 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
3088 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
3089 ast_channel_sendurl( peer, args.url );
3090 }
3091 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
3092 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
3093 ast_channel_publish_dial(chan, peer, NULL, pa.status);
3094 /* hang up on the callee -- he didn't want to talk anyway! */
3095 ast_autoservice_chan_hangup_peer(chan, peer);
3096 res = 0;
3097 goto out;
3098 }
3099 }
3100 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
3101 res = 0;
3102 } else {
3103 int digit = 0;
3104 struct ast_channel *chans[2];
3105 struct ast_channel *active_chan;
3106 char *calledfile = NULL, *callerfile = NULL;
3107 int calledstream = 0, callerstream = 0;
3108
3109 chans[0] = chan;
3110 chans[1] = peer;
3111
3112 /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
3113 callerfile = opt_args[OPT_ARG_ANNOUNCE];
3114 calledfile = strsep(&callerfile, ":");
3115
3116 /* stream the file(s) */
3117 if (!ast_strlen_zero(calledfile)) {
3118 res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
3119 if (res) {
3120 res = 0;
3121 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
3122 } else {
3123 calledstream = 1;
3124 }
3125 }
3126 if (!ast_strlen_zero(callerfile)) {
3127 res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
3128 if (res) {
3129 res = 0;
3130 ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
3131 } else {
3132 callerstream = 1;
3133 }
3134 }
3135
3136 /* can't use ast_waitstream, because we're streaming two files at once, and can't block
3137 We'll need to handle both channels at once. */
3138
3139 ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
3140 while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
3141 int mspeer, mschan;
3142
3143 mspeer = ast_sched_wait(ast_channel_sched(peer));
3144 mschan = ast_sched_wait(ast_channel_sched(chan));
3145
3146 if (calledstream) {
3147 if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
3148 ast_stopstream(peer);
3149 calledstream = 0;
3150 }
3151 }
3152 if (callerstream) {
3153 if (mschan < 0 && !ast_channel_timingfunc(chan)) {
3154 ast_stopstream(chan);
3155 callerstream = 0;
3156 }
3157 }
3158
3159 if (!calledstream && !callerstream) {
3160 break;
3161 }
3162
3163 if (mspeer < 0)
3164 mspeer = 1000;
3165
3166 if (mschan < 0)
3167 mschan = 1000;
3168
3169 /* wait for the lowest maximum of the two */
3170 active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
3171 if (active_chan) {
3172 struct ast_channel *other_chan;
3173 struct ast_frame *fr = ast_read(active_chan);
3174
3175 if (!fr) {
3176 ast_autoservice_chan_hangup_peer(chan, peer);
3177 res = -1;
3178 goto done;
3179 }
3180 switch (fr->frametype) {
3181 case AST_FRAME_DTMF_END:
3182 digit = fr->subclass.integer;
3183 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
3184 ast_stopstream(peer);
3185 res = ast_senddigit(chan, digit, 0);
3186 }
3187 break;
3188 case AST_FRAME_CONTROL:
3189 switch (fr->subclass.integer) {
3190 case AST_CONTROL_HANGUP:
3191 ast_frfree(fr);
3192 ast_autoservice_chan_hangup_peer(chan, peer);
3193 res = -1;
3194 goto done;
3195 case AST_CONTROL_CONNECTED_LINE:
3196 /* Pass COLP update to the other channel. */
3197 if (active_chan == chan) {
3198 other_chan = peer;
3199 } else {
3200 other_chan = chan;
3201 }
3202 if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)) {
3203 ast_indicate_data(other_chan, fr->subclass.integer,
3204 fr->data.ptr, fr->datalen);
3205 }
3206 break;
3207 default:
3208 break;
3209 }
3210 break;
3211 default:
3212 /* Ignore all others */
3213 break;
3214 }
3215 ast_frfree(fr);
3216 }
3217 ast_sched_runq(ast_channel_sched(peer));
3218 ast_sched_runq(ast_channel_sched(chan));
3219 }
3220 ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
3221 }
3222
3223 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
3224 /* chan and peer are going into the PBX; as such neither are considered
3225 * outgoing channels any longer */
3226 ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);
3227
3228 ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
3229 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
3230 /* peer goes to the same context and extension as chan, so just copy info from chan*/
3231 ast_channel_lock(peer);
3232 ast_channel_stage_snapshot(peer);
3233 ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
3234 ast_channel_context_set(peer, ast_channel_context(chan));
3235 ast_channel_exten_set(peer, ast_channel_exten(chan));
3236 ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
3237 ast_channel_stage_snapshot_done(peer);
3238 ast_channel_unlock(peer);
3239 if (ast_pbx_start(peer)) {
3240 ast_autoservice_chan_hangup_peer(chan, peer);
3241 }
3242 if (continue_exec)
3243 *continue_exec = 1;
3244 res = 0;
3245 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3246 goto done;
3247 }
3248
3249 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
3250 const char *gosub_result_peer;
3251 char *gosub_argstart;
3252 char *gosub_args = NULL;
3253 int gosub_res = -1;
3254
3255 ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
3256 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
3257 if (gosub_argstart) {
3258 const char *what_is_s = "s";
3259 *gosub_argstart = 0;
3260 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)) &&
3261 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))) {
3262 what_is_s = "~~s~~";
3263 }
3264 if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
3265 gosub_args = NULL;
3266 }
3267 *gosub_argstart = ',';
3268 } else {
3269 const char *what_is_s = "s";
3270 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)) &&
3271 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))) {
3272 what_is_s = "~~s~~";
3273 }
3274 if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
3275 gosub_args = NULL;
3276 }
3277 }
3278 if (gosub_args) {
3279 gosub_res = ast_app_exec_sub(chan, peer, gosub_args, 0);
3280 ast_free(gosub_args);
3281 } else {
3282 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
3283 }
3284
3285 ast_channel_lock_both(chan, peer);
3286
3287 if (!gosub_res && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
3288 char *gosub_transfer_dest;
3289 char *gosub_result = ast_strdupa(gosub_result_peer);
3290 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
3291
3292 /* Inherit return value from the peer, so it can be used in the master */
3293 if (gosub_retval) {
3294 pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
3295 }
3296
3297 ast_channel_unlock(peer);
3298 ast_channel_unlock(chan);
3299
3300 if (!strcasecmp(gosub_result, "BUSY")) {
3301 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3302 ast_set_flag64(peerflags, OPT_GO_ON);
3303 gosub_res = -1;
3304 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
3305 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3306 ast_set_flag64(peerflags, OPT_GO_ON);
3307 gosub_res = -1;
3308 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
3309 /* Hangup peer and continue with the next extension priority. */
3310 ast_set_flag64(peerflags, OPT_GO_ON);
3311 gosub_res = -1;
3312 } else if (!strcasecmp(gosub_result, "ABORT")) {
3313 /* Hangup both ends unless the caller has the g flag */
3314 gosub_res = -1;
3315 } else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
3316 gosub_transfer_dest = gosub_result + 5;
3317 gosub_res = -1;
3318 /* perform a transfer to a new extension */
3319 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
3320 ast_replace_subargument_delimiter(gosub_transfer_dest);
3321 }
3322 if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
3323 ast_set_flag64(peerflags, OPT_GO_ON);
3324 }
3325 }
3326 if (gosub_res) {
3327 res = gosub_res;
3328 if (!dial_end_raised) {
3329 ast_channel_publish_dial(chan, peer, NULL, gosub_result);
3330 dial_end_raised = 1;
3331 }
3332 }
3333 } else {
3334 ast_channel_unlock(peer);
3335 ast_channel_unlock(chan);
3336 }
3337 }
3338
3339 if (!res) {
3340
3341 /* None of the Dial options changed our status; inform
3342 * everyone that this channel answered
3343 */
3344 if (!dial_end_raised) {
3345 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3346 dial_end_raised = 1;
3347 }
3348
3349 if (!ast_tvzero(calldurationlimit)) {
3350 struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
3351 ast_channel_lock(peer);
3352 ast_channel_whentohangup_set(peer, &whentohangup);
3353 ast_channel_unlock(peer);
3354 }
3355 if (!ast_strlen_zero(dtmfcalled)) {
3356 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
3357 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
3358 }
3359 if (!ast_strlen_zero(dtmfcalling)) {
3360 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
3361 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
3362 }
3363 }
3364
3365 if (res) { /* some error */
3366 if (!ast_check_hangup(chan) && ast_check_hangup(peer)) {
3367 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
3368 }
3369 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3370 if (ast_bridge_setup_after_goto(peer)
3371 || ast_pbx_start(peer)) {
3372 ast_autoservice_chan_hangup_peer(chan, peer);
3373 }
3374 res = -1;
3375 } else {
3376 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
3377 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
3378 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
3379 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
3380 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
3381 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
3382 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
3383 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
3384 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
3385 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
3386 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
3387 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
3388 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
3389 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
3390 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
3391 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
3392 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
3393 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
3394 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
3395 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
3396
3397 config.end_bridge_callback = end_bridge_callback;
3398 config.end_bridge_callback_data = chan;
3399 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
3400
3401 if (moh) {
3402 moh = 0;
3403 ast_moh_stop(chan);
3404 } else if (sentringing) {
3405 sentringing = 0;
3406 ast_indicate(chan, -1);
3407 }
3408 /* Be sure no generators are left on it and reset the visible indication */
3409 ast_deactivate_generator(chan);
3410 ast_channel_visible_indication_set(chan, 0);
3411 /* Make sure channels are compatible */
3412 res = ast_channel_make_compatible(chan, peer);
3413 if (res < 0) {
3414 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));
3415 ast_autoservice_chan_hangup_peer(chan, peer);
3416 res = -1;
3417 goto done;
3418 }
3419 if (opermode) {
3420 struct oprmode oprmode;
3421
3422 oprmode.peer = peer;
3423 oprmode.mode = opermode;
3424
3425 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
3426 }
3427 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3428
3429 res = ast_bridge_call(chan, peer, &config);
3430 }
3431 }
3432out:
3433 if (moh) {
3434 moh = 0;
3435 ast_moh_stop(chan);
3436 } else if (sentringing) {
3437 sentringing = 0;
3438 ast_indicate(chan, -1);
3439 }
3440
3441 if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3442 ast_filedelete(pa.privintro, NULL);
3443 if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3444 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
3445 } else {
3446 ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
3447 }
3448 }
3449
3450 ast_channel_early_bridge(chan, NULL);
3451 /* forward 'answered elsewhere' if we received it */
3452 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
3453 hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
3454 } else if (pa.canceled) { /* Caller canceled */
3455 if (ast_channel_hangupcause(chan))
3456 hanguptreecause = ast_channel_hangupcause(chan);
3457 else
3458 hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
3459 }
3460 hanguptree(&out_chans, NULL, hanguptreecause);
3461 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3462 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
3463
3464 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
3465 if (!ast_tvzero(calldurationlimit))
3466 memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
3467 res = 0;
3468 }
3469
3470done:
3471 if (config.answer_topology) {
3472 ast_trace(2, "%s Cleaning up topology: %p %s\n",
3473 peer ? ast_channel_name(peer) : "<no channel>", &config.answer_topology,
3474 ast_str_tmp(256, ast_stream_topology_to_str(config.answer_topology, &STR_TMP)));
3475
3476 /*
3477 * At this point, the channel driver that answered should have bumped the
3478 * topology refcount for itself. Here we're cleaning up the reference we added
3479 * in wait_for_answer().
3480 */
3481 ast_stream_topology_free(config.answer_topology);
3482 }
3483 if (config.warning_sound) {
3484 ast_free((char *)config.warning_sound);
3485 }
3486 if (config.end_sound) {
3487 ast_free((char *)config.end_sound);
3488 }
3489 if (config.start_sound) {
3490 ast_free((char *)config.start_sound);
3491 }
3492 ast_ignore_cc(chan);
3493 SCOPE_EXIT_RTN_VALUE(res, "%s: Done\n", ast_channel_name(chan));
3494}

References ao2_ref, args, ast_answer(), AST_APP_ARG, 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_setup_after_goto(), ast_bridge_timelimit(), ast_call(), ast_callerid_parse(), ast_calloc, ast_cause2str(), AST_CAUSE_ANSWERED_ELSEWHERE, AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NORMAL_CLEARING, ast_cc_busy_interface(), ast_cc_call_failed(), ast_cc_call_init(), ast_cc_callback(), ast_cc_extension_monitor_add_dialstring(), ast_cdr_reset(), ast_channel_adsicpe_set(), ast_channel_appl_set(), ast_channel_caller(), ast_channel_clear_flag(), ast_channel_connected(), ast_channel_connected_line_sub(), 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_dialed(), ast_channel_early_bridge(), ast_channel_exten(), ast_channel_exten_set(), ast_channel_flags(), ast_channel_get_device_name(), ast_channel_get_stream_topology(), ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_inherit_variables(), ast_channel_language(), ast_channel_lock, ast_channel_lock_both, ast_channel_make_compatible(), ast_channel_musicclass(), AST_CHANNEL_NAME, ast_channel_name(), ast_channel_priority(), ast_channel_priority_set(), ast_channel_publish_dial(), ast_channel_redirecting(), ast_channel_req_accountcodes(), AST_CHANNEL_REQUESTOR_BRIDGE_PEER, ast_channel_sched(), ast_channel_sendurl(), ast_channel_set_caller_event(), ast_channel_set_connected_line(), ast_channel_set_flag(), ast_channel_setoption(), ast_channel_stage_snapshot(), ast_channel_stage_snapshot_done(), 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_check_hangup_locked(), ast_clear_flag, ast_connected_line_copy_from_caller(), AST_CONTROL_CONNECTED_LINE, AST_CONTROL_HANGUP, AST_CONTROL_PROGRESS, AST_CONTROL_RINGING, ast_copy_flags64, ast_copy_string(), ast_datastore_alloc, 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_PARKCALL, AST_FEATURE_REDIRECT, ast_filedelete(), ast_fileexists(), AST_FLAG_END_DTMF_ONLY, AST_FLAG_OUTGOING, AST_FRAME_CONTROL, AST_FRAME_DTMF_END, ast_free, ast_frfree, ast_hangup(), ast_ignore_cc(), ast_indicate(), ast_indicate_data(), AST_LIST_EMPTY, AST_LIST_FIRST, AST_LIST_HEAD_NOLOCK_INIT_VALUE, AST_LIST_INSERT_TAIL, AST_LIST_REMOVE_CURRENT, AST_LIST_TRAVERSE, AST_LIST_TRAVERSE_SAFE_BEGIN, AST_LIST_TRAVERSE_SAFE_END, ast_log, AST_MAX_EXTENSION, ast_max_forwards_decrement(), ast_max_forwards_get(), 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_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_with_stream_topology(), 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_str2cause(), ast_str_tmp, ast_strdup, ast_strdupa, ast_stream_get_state(), ast_stream_set_state(), AST_STREAM_STATE_RECVONLY, AST_STREAM_STATE_SENDONLY, ast_stream_topology_clone(), ast_stream_topology_free(), ast_stream_topology_get_count(), ast_stream_topology_get_stream(), ast_stream_topology_to_str(), ast_streamfile(), ast_strip(), ast_strlen_zero(), ast_test_flag64, ast_trace, ast_tvadd(), ast_tvnow(), ast_tvzero(), ast_verb, ast_waitfor_n(), CAN_EARLY_BRIDGE, privacy_args::canceled, chanlist::chan, cause_args::chan, chanlist_free(), config, chanlist::connected, connected, ast_datastore::data, ast_frame::data, ast_frame::datalen, DIAL_CALLERID_ABSENT, dial_exec_options, dial_handle_playtones(), DIAL_NOFORWARDHTML, DIAL_STILLGOING, digit, done, end_bridge_callback(), end_bridge_callback_data_fixup(), ast_flags64::flags, ast_frame::frametype, get_cid_name(), handle_cause(), hanguptree(), ast_party_caller::id, ast_party_connected_line::id, ast_frame_subclass::integer, chanlist::interface, LOG_ERROR, LOG_NOTICE, LOG_WARNING, oprmode::mode, name, ast_party_id::name, NULL, chanlist::number, ast_party_id::number, OPT_ANNOUNCE, OPT_ARG_ANNOUNCE, OPT_ARG_ARRAY_SIZE, OPT_ARG_CALLEE_GOSUB, 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_HANGUPCAUSE, OPT_ARG_MUSICBACK, 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_GOSUB, OPT_CALLEE_HANGUP, 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_HANGUPCAUSE, OPT_HEARPULSING, OPT_IGNORE_CONNECTEDLINE, OPT_IGNORE_FORWARDING, OPT_MUSICBACK, OPT_OPERMODE, OPT_ORIGINAL_CLID, OPT_PREDIAL_CALLEE, OPT_PREDIAL_CALLER, OPT_PRIVACY, OPT_RESETCDR, OPT_RING_WITH_EARLY_MEDIA, OPT_RINGBACK, OPT_SCREEN_NOINTRO, OPT_SCREENING, OPT_SENDDTMF, OPT_TOPOLOGY_PRESERVE, options, chanlist::orig_chan_name, out, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), oprmode::peer, ast_party_number::presentation, privacy_args::privdb_val, privacy_args::privintro, ast_frame::ptr, result, S_COR, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, privacy_args::sentringing, setup_peer_after_bridge_goto(), setup_privacy_args(), privacy_args::status, ast_party_name::str, ast_party_number::str, ast_party_subaddress::str, strsep(), chanlist::stuff, ast_party_id::subaddress, ast_frame::subclass, ast_party_id::tag, chanlist::tech, ast_channel::tech, topology_ds_info, ast_party_dialed::transit_network_select, url, ast_party_name::valid, ast_party_number::valid, and wait_for_answer().

Referenced by dial_exec(), and retrydial_exec().

◆ dial_handle_playtones()

static int dial_handle_playtones ( struct ast_channel *  chan,
const char *  data 
)
static

Definition at line 2271 of file app_dial.c.

2272{
2273 struct ast_tone_zone_sound *ts = NULL;
2274 int res;
2275 const char *str = data;
2276
2277 if (ast_strlen_zero(str)) {
2278 ast_debug(1,"Nothing to play\n");
2279 return -1;
2280 }
2281
2282 ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2283
2284 if (ts && ts->data[0]) {
2285 res = ast_playtones_start(chan, 0, ts->data, 0);
2286 } else {
2287 res = -1;
2288 }
2289
2290 if (ts) {
2291 ts = ast_tone_zone_sound_unref(ts);
2292 }
2293
2294 if (res) {
2295 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2296 }
2297
2298 return res;
2299}

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, NULL, and str.

Referenced by dial_exec_full().

◆ do_forward()

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()

Parameters
oOutgoing call channel list.
numIncoming call channel cause accumulation
peerflagsDial option flags
singleTRUE if there is only one outgoing call.
caller_entertainedTRUE if the caller is being entertained by MOH or ringback.
toRemaining call timeout time.
forced_clidOPT_FORCECLID caller id to send
stored_clidCaller id representing the called party if needed

XXX this code is highly suspicious, as it essentially overwrites the outgoing channel without properly deleting it.

Todo:
eventually this function should be integrated into and replaced by ast_call_forward()

Definition at line 944 of file app_dial.c.

947{
948 char tmpchan[256];
949 char forwarder[AST_CHANNEL_NAME];
950 struct ast_channel *original = o->chan;
951 struct ast_channel *c = o->chan; /* the winner */
952 struct ast_channel *in = num->chan; /* the input channel */
953 char *stuff;
954 const char *tech;
955 int cause;
956 struct ast_party_caller caller;
957
958 ast_copy_string(forwarder, ast_channel_name(c), sizeof(forwarder));
959 ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
960 if ((stuff = strchr(tmpchan, '/'))) {
961 *stuff++ = '\0';
962 tech = tmpchan;
963 } else {
964 const char *forward_context;
965 ast_channel_lock(c);
966 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
967 if (ast_strlen_zero(forward_context)) {
968 forward_context = NULL;
969 }
970 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
971 ast_channel_unlock(c);
972 stuff = tmpchan;
973 tech = "Local";
974 }
975 if (!strcasecmp(tech, "Local")) {
976 /*
977 * Drop the connected line update block for local channels since
978 * this is going to run dialplan and the user can change his
979 * mind about what connected line information he wants to send.
980 */
981 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
982 }
983
984 /* Before processing channel, go ahead and check for forwarding */
985 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
986 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
987 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
988 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
989 ast_channel_publish_dial_forward(in, original, NULL, NULL, "CANCEL",
990 ast_channel_call_forward(original));
991 c = o->chan = NULL;
992 cause = AST_CAUSE_BUSY;
993 } else {
994 struct ast_stream_topology *topology;
995
996 ast_channel_lock(in);
997 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(in));
998 ast_channel_unlock(in);
999
1000 /* Setup parameters */
1001 c = o->chan = ast_request_with_stream_topology(tech, topology, NULL, in, stuff, &cause);
1002
1003 ast_stream_topology_free(topology);
1004
1005 if (c) {
1006 if (single && !caller_entertained) {
1007 ast_channel_make_compatible(in, o->chan);
1008 }
1009 ast_channel_lock_both(in, o->chan);
1010 ast_channel_inherit_variables(in, o->chan);
1011 ast_channel_datastore_inherit(in, o->chan);
1012 pbx_builtin_setvar_helper(o->chan, "FORWARDERNAME", forwarder);
1013 ast_max_forwards_decrement(o->chan);
1014 ast_channel_unlock(in);
1015 ast_channel_unlock(o->chan);
1016 /* When a call is forwarded, we don't want to track new interfaces
1017 * dialed for CC purposes. Setting the done flag will ensure that
1018 * any Dial operations that happen later won't record CC interfaces.
1019 */
1020 ast_ignore_cc(o->chan);
1021 ast_verb(3, "Not accepting call completion offers from call-forward recipient %s\n",
1022 ast_channel_name(o->chan));
1023 } else
1024 ast_log(LOG_NOTICE,
1025 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
1026 tech, stuff, cause);
1027 }
1028 if (!c) {
1029 ast_channel_publish_dial(in, original, stuff, "BUSY");
1030 ast_clear_flag64(o, DIAL_STILLGOING);
1031 handle_cause(cause, num);
1032 ast_hangup(original);
1033 } else {
1034 ast_channel_lock_both(c, original);
1035 ast_party_redirecting_copy(ast_channel_redirecting(c),
1036 ast_channel_redirecting(original));
1037 ast_channel_unlock(c);
1038 ast_channel_unlock(original);
1039
1040 ast_channel_lock_both(c, in);
1041
1042 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
1043 ast_rtp_instance_early_bridge_make_compatible(c, in);
1044 }
1045
1046 if (!ast_channel_redirecting(c)->from.number.valid
1047 || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
1048 /*
1049 * The call was not previously redirected so it is
1050 * now redirected from this number.
1051 */
1052 ast_party_number_free(&ast_channel_redirecting(c)->from.number);
1053 ast_party_number_init(&ast_channel_redirecting(c)->from.number);
1054 ast_channel_redirecting(c)->from.number.valid = 1;
1055 ast_channel_redirecting(c)->from.number.str =
1056 ast_strdup(ast_channel_exten(in));
1057 }
1058
1059 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
1060
1061 /* Determine CallerID to store in outgoing channel. */
1062 ast_party_caller_set_init(&caller, ast_channel_caller(c));
1063 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1064 caller.id = *stored_clid;
1065 ast_channel_set_caller_event(c, &caller, NULL);
1066 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1067 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
1068 ast_channel_caller(c)->id.number.str, NULL))) {
1069 /*
1070 * The new channel has no preset CallerID number by the channel
1071 * driver. Use the dialplan extension and hint name.
1072 */
1073 caller.id = *stored_clid;
1074 ast_channel_set_caller_event(c, &caller, NULL);
1075 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1076 } else {
1077 ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
1078 }
1079
1080 /* Determine CallerID for outgoing channel to send. */
1081 if (ast_test_flag64(o, OPT_FORCECLID)) {
1082 struct ast_party_connected_line connected;
1083
1084 ast_party_connected_line_init(&connected);
1085 connected.id = *forced_clid;
1086 ast_party_connected_line_copy(ast_channel_connected(c), &connected);
1087 } else {
1088 ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
1089 }
1090
1091 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
1092
1093 ast_channel_appl_set(c, "AppDial");
1094 ast_channel_data_set(c, "(Outgoing Line)");
1095 ast_channel_publish_snapshot(c);
1096
1097 ast_channel_unlock(in);
1098 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1099 struct ast_party_redirecting redirecting;
1100
1101 /*
1102 * Redirecting updates to the caller make sense only on single
1103 * calls.
1104 *
1105 * Need to re-evalute if unlocking is still required here as macro is gone
1106 */
1107 ast_party_redirecting_init(&redirecting);
1108 ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
1109 ast_channel_unlock(c);
1110 if (ast_channel_redirecting_sub(c, in, &redirecting, 0)) {
1111 ast_channel_update_redirecting(in, &redirecting, NULL);
1112 }
1113 ast_party_redirecting_free(&redirecting);
1114 } else {
1115 ast_channel_unlock(c);
1116 }
1117
1118 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1119 *to = -1;
1120 }
1121
1122 if (ast_call(c, stuff, 0)) {
1123 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1124 tech, stuff);
1125 ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1126 ast_clear_flag64(o, DIAL_STILLGOING);
1127 ast_hangup(original);
1128 ast_hangup(c);
1129 c = o->chan = NULL;
1130 num->nochan++;
1131 } else {
1132 ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1133 ast_channel_call_forward(original));
1134
1135 ast_channel_publish_dial(in, c, stuff, NULL);
1136
1137 /* Hangup the original channel now, in case we needed it */
1138 ast_hangup(original);
1139 }
1140 if (single && !caller_entertained) {
1141 ast_indicate(in, -1);
1142 }
1143 }
1144}

References ast_call(), AST_CAUSE_BUSY, 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_get_stream_topology(), ast_channel_inherit_variables(), ast_channel_lock, ast_channel_lock_both, ast_channel_make_compatible(), AST_CHANNEL_NAME, ast_channel_name(), ast_channel_publish_dial(), ast_channel_publish_dial_forward(), ast_channel_publish_snapshot(), ast_channel_redirecting(), ast_channel_redirecting_sub(), ast_channel_req_accountcodes(), AST_CHANNEL_REQUESTOR_BRIDGE_PEER, 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_max_forwards_decrement(), 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_with_stream_topology(), ast_rtp_instance_early_bridge_make_compatible(), ast_set_flag64, ast_strdup, ast_stream_topology_clone(), ast_stream_topology_free(), ast_strlen_zero(), ast_test_flag64, ast_verb, c, CAN_EARLY_BRIDGE, chanlist::chan, cause_args::chan, connected, DIAL_CALLERID_ABSENT, DIAL_STILLGOING, ast_party_redirecting::from, handle_cause(), ast_party_caller::id, in, LOG_NOTICE, cause_args::nochan, NULL, ast_party_id::number, OPT_CANCEL_TIMEOUT, OPT_FORCECLID, OPT_IGNORE_CONNECTEDLINE, OPT_IGNORE_FORWARDING, OPT_ORIGINAL_CLID, pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), S_COR, ast_party_number::str, ast_channel::tech, ast_party_redirecting::to, ast_party_dialed::transit_network_select, and ast_party_number::valid.

Referenced by wait_for_answer().

◆ end_bridge_callback()

static void end_bridge_callback ( void *  data)
static

Definition at line 2255 of file app_dial.c.

2256{
2257 struct ast_channel *chan = data;
2258
2259 ast_channel_lock(chan);
2260 ast_channel_stage_snapshot(chan);
2261 set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
2262 set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
2263 ast_channel_stage_snapshot_done(chan);
2264 ast_channel_unlock(chan);
2265}

References ast_channel_get_duration_ms(), ast_channel_get_up_time_ms(), ast_channel_lock, ast_channel_stage_snapshot(), ast_channel_stage_snapshot_done(), ast_channel_unlock, ast_channel::data, and set_duration_var().

Referenced by dial_exec_full().

◆ end_bridge_callback_data_fixup()

static void end_bridge_callback_data_fixup ( struct ast_bridge_config *  bconfig,
struct ast_channel *  originator,
struct ast_channel *  terminator 
)
static

Definition at line 2267 of file app_dial.c.

2267 {
2268 bconfig->end_bridge_callback_data = originator;
2269}

References ast_bridge_config::end_bridge_callback_data.

Referenced by dial_exec_full().

◆ get_cid_name()

static const char * get_cid_name ( char *  name,
int  namelen,
struct ast_channel *  chan 
)
static

Definition at line 914 of file app_dial.c.

915{
916 const char *context;
917 const char *exten;
918
919 ast_channel_lock(chan);
920 context = ast_strdupa(ast_channel_context(chan));
921 exten = ast_strdupa(ast_channel_exten(chan));
922 ast_channel_unlock(chan);
923
924 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
925}

References ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_unlock, ast_get_hint(), ast_strdupa, ast_channel::context, ast_channel::exten, name, and NULL.

Referenced by dial_exec_full().

◆ handle_cause()

static void handle_cause ( int  cause,
struct cause_args *  num 
)
static

Definition at line 877 of file app_dial.c.

878{
879 switch(cause) {
880 case AST_CAUSE_BUSY:
881 num->busy++;
882 break;
883 case AST_CAUSE_CONGESTION:
884 num->congestion++;
885 break;
886 case AST_CAUSE_NO_ROUTE_DESTINATION:
887 case AST_CAUSE_UNREGISTERED:
888 num->nochan++;
889 break;
890 case AST_CAUSE_NO_ANSWER:
891 case AST_CAUSE_NORMAL_CLEARING:
892 break;
893 default:
894 num->nochan++;
895 break;
896 }
897}

References AST_CAUSE_BUSY, AST_CAUSE_CONGESTION, AST_CAUSE_NO_ANSWER, AST_CAUSE_NO_ROUTE_DESTINATION, AST_CAUSE_NORMAL_CLEARING, AST_CAUSE_UNREGISTERED, cause_args::busy, cause_args::congestion, and cause_args::nochan.

Referenced by dial_exec_full(), do_forward(), and wait_for_answer().

◆ hanguptree()

static void hanguptree ( struct dial_head *  out_chans,
struct ast_channel *  exception,
int  hangupcause 
)
static

Definition at line 847 of file app_dial.c.

848{
849 /* Hang up a tree of stuff */
850 struct chanlist *outgoing;
851
852 while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
853 /* Hangup any existing lines we have open */
854 if (outgoing->chan && (outgoing->chan != exception)) {
855 if (hangupcause >= 0) {
856 /* This is for the channel drivers */
857 ast_channel_hangupcause_set(outgoing->chan, hangupcause);
858 }
859 ast_hangup(outgoing->chan);
860 }
861 chanlist_free(outgoing);
862 }
863}

References ast_channel_hangupcause_set(), ast_hangup(), AST_LIST_REMOVE_HEAD, and chanlist_free().

Referenced by dial_exec_full().

◆ load_module()

static int load_module ( void  )
static

Definition at line 3623 of file app_dial.c.

3624{
3625 int res;
3626
3627 res = ast_register_application_xml(app, dial_exec);
3628 res |= ast_register_application_xml(rapp, retrydial_exec);
3629
3630 return res;
3631}

References app, ast_register_application_xml, dial_exec(), rapp, and retrydial_exec().

◆ onedigit_goto()

static int onedigit_goto ( struct ast_channel *  chan,
const char *  context,
char  exten,
int  pri 
)
static

Definition at line 899 of file app_dial.c.

900{
901 char rexten[2] = { exten, '\0' };
902
903 if (context) {
904 if (!ast_goto_if_exists(chan, context, rexten, pri))
905 return 1;
906 } else {
907 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
908 return 1;
909 }
910 return 0;
911}

References ast_channel_context(), ast_goto_if_exists(), ast_channel::context, and ast_channel::exten.

Referenced by retrydial_exec(), and wait_for_answer().

◆ publish_dial_end_event()

static void publish_dial_end_event ( struct ast_channel *  in,
struct dial_head *  out_chans,
struct ast_channel *  exception,
const char *  status 
)
static

Definition at line 1156 of file app_dial.c.

1157{
1158 struct chanlist *outgoing;
1159 AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1160 if (!outgoing->chan || outgoing->chan == exception) {
1161 continue;
1162 }
1163 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1164 }
1165}

References ast_channel_publish_dial(), AST_LIST_TRAVERSE, in, NULL, and status.

Referenced by wait_for_answer().

◆ retrydial_exec()

static int retrydial_exec ( struct ast_channel *  chan,
const char *  data 
)
static

Definition at line 3505 of file app_dial.c.

3506{
3507 char *parse;
3508 const char *context = NULL;
3509 int sleepms = 0, loops = 0, res = -1;
3510 struct ast_flags64 peerflags = { 0, };
3511 AST_DECLARE_APP_ARGS(args,
3512 AST_APP_ARG(announce);
3513 AST_APP_ARG(sleep);
3514 AST_APP_ARG(retries);
3515 AST_APP_ARG(dialdata);
3516 );
3517
3518 if (ast_strlen_zero(data)) {
3519 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
3520 return -1;
3521 }
3522
3523 parse = ast_strdupa(data);
3524 AST_STANDARD_APP_ARGS(args, parse);
3525
3526 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
3527 sleepms *= 1000;
3528
3529 if (!ast_strlen_zero(args.retries)) {
3530 loops = atoi(args.retries);
3531 }
3532
3533 if (!args.dialdata) {
3534 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
3535 goto done;
3536 }
3537
3538 if (sleepms < 1000)
3539 sleepms = 10000;
3540
3541 if (!loops)
3542 loops = -1; /* run forever */
3543
3544 ast_channel_lock(chan);
3545 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
3546 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
3547 ast_channel_unlock(chan);
3548
3549 res = 0;
3550 while (loops) {
3551 int continue_exec;
3552
3553 ast_channel_data_set(chan, "Retrying");
3554 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3555 ast_moh_stop(chan);
3556
3557 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
3558 if (continue_exec)
3559 break;
3560
3561 if (res == 0) {
3562 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
3563 if (!ast_strlen_zero(args.announce)) {
3564 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3565 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3566 ast_waitstream(chan, AST_DIGIT_ANY);
3567 } else
3568 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3569 }
3570 if (!res && sleepms) {
3571 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3572 ast_moh_start(chan, NULL, NULL);
3573 res = ast_waitfordigit(chan, sleepms);
3574 }
3575 } else {
3576 if (!ast_strlen_zero(args.announce)) {
3577 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3578 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3579 res = ast_waitstream(chan, "");
3580 } else
3581 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3582 }
3583 if (sleepms) {
3584 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3585 ast_moh_start(chan, NULL, NULL);
3586 if (!res)
3587 res = ast_waitfordigit(chan, sleepms);
3588 }
3589 }
3590 }
3591
3592 if (res < 0 || res == AST_PBX_INCOMPLETE) {
3593 break;
3594 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
3595 if (onedigit_goto(chan, context, (char) res, 1)) {
3596 res = 0;
3597 break;
3598 }
3599 }
3600 loops--;
3601 }
3602 if (loops == 0)
3603 res = 0;
3604 else if (res == 1)
3605 res = 0;
3606
3607 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3608 ast_moh_stop(chan);
3609 done:
3610 return res;
3611}

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_strdupa, ast_streamfile(), ast_strlen_zero(), ast_test_flag, ast_test_flag64, ast_waitfordigit(), ast_waitstream(), dial_exec_full(), done, LOG_ERROR, LOG_WARNING, NULL, onedigit_goto(), OPT_DTMF_EXIT, pbx_builtin_getvar_helper(), and rapp.

Referenced by load_module().

◆ set_duration_var()

static void set_duration_var ( struct ast_channel *  chan,
const char *  var_base,
int64_t  duration 
)
static

Definition at line 1196 of file app_dial.c.

1197{
1198 char buf[32];
1199 char full_var_name[128];
1200
1201 snprintf(buf, sizeof(buf), "%" PRId64, duration / 1000);
1202 pbx_builtin_setvar_helper(chan, var_base, buf);
1203
1204 snprintf(full_var_name, sizeof(full_var_name), "%s_MS", var_base);
1205 snprintf(buf, sizeof(buf), "%" PRId64, duration);
1206 pbx_builtin_setvar_helper(chan, full_var_name, buf);
1207}

References buf, and pbx_builtin_setvar_helper().

Referenced by end_bridge_callback(), and wait_for_answer().

◆ setup_peer_after_bridge_goto()

static void setup_peer_after_bridge_goto ( struct ast_channel *  chan,
struct ast_channel *  peer,
struct ast_flags64 *  opts,
char *  opt_args[] 
)
static

Definition at line 2311 of file app_dial.c.

2312{
2313 const char *context;
2314 const char *extension;
2315 int priority;
2316
2317 if (ast_test_flag64(opts, OPT_PEER_H)) {
2318 ast_channel_lock(chan);
2319 context = ast_strdupa(ast_channel_context(chan));
2320 ast_channel_unlock(chan);
2321 ast_bridge_set_after_h(peer, context);
2322 } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2323 ast_channel_lock(chan);
2324 context = ast_strdupa(ast_channel_context(chan));
2325 extension = ast_strdupa(ast_channel_exten(chan));
2326 priority = ast_channel_priority(chan);
2327 ast_channel_unlock(chan);
2328 ast_bridge_set_after_go_on(peer, context, extension, priority,
2329 opt_args[OPT_ARG_CALLEE_GO_ON]);
2330 }
2331}

References ast_bridge_set_after_go_on(), ast_bridge_set_after_h(), ast_channel_context(), ast_channel_exten(), ast_channel_lock, ast_channel_priority(), ast_channel_unlock, ast_strdupa, ast_test_flag64, OPT_ARG_CALLEE_GO_ON, OPT_CALLEE_GO_ON, OPT_PEER_H, and priority.

Referenced by dial_exec_full().

◆ setup_privacy_args()

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 2156 of file app_dial.c.

2158{
2159 char callerid[60];
2160 int res;
2161 char *l;
2162
2163 if (ast_channel_caller(chan)->id.number.valid
2164 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2165 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2166 ast_shrink_phone_number(l);
2167 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
2168 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
2169 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
2170 } else {
2171 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
2172 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2173 }
2174 } else {
2175 char *tnam, *tn2;
2176
2177 tnam = ast_strdupa(ast_channel_name(chan));
2178 /* clean the channel name so slashes don't try to end up in disk file name */
2179 for (tn2 = tnam; *tn2; tn2++) {
2180 if (*tn2 == '/') /* any other chars to be afraid of? */
2181 *tn2 = '=';
2182 }
2183 ast_verb(3, "Privacy-- callerid is empty\n");
2184
2185 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
2186 l = callerid;
2187 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2188 }
2189
2190 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
2191
2192 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
2193 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
2194 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
2195 pa->privdb_val = AST_PRIVACY_ALLOW;
2196 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
2197 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
2198 }
2199
2200 if (pa->privdb_val == AST_PRIVACY_DENY) {
2201 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
2202 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2203 return 0;
2204 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
2205 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2206 return 0; /* Is this right? */
2207 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
2208 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2209 return 0; /* is this right??? */
2210 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
2211 /* Get the user's intro, store it in priv-callerintros/$CID,
2212 unless it is already there-- this should be done before the
2213 call is actually dialed */
2214
2215 /* make sure the priv-callerintros dir actually exists */
2216 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
2217 if ((res = ast_mkdir(pa->privintro, 0755))) {
2218 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
2219 return -1;
2220 }
2221
2222 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
2223 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
2224 /* the DELUX version of this code would allow this caller the
2225 option to hear and retape their previously recorded intro.
2226 */
2227 } else {
2228 int duration; /* for feedback from play_and_wait */
2229 /* the file doesn't exist yet. Let the caller submit his
2230 vocal intro for posterity */
2231 /* priv-recordintro script:
2232 "At the tone, please say your name:"
2233 */
2234 int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
2235 ast_answer(chan);
2236 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 */
2237 /* don't think we'll need a lock removed, we took care of
2238 conflicts by naming the pa.privintro file */
2239 if (res == -1) {
2240 /* Delete the file regardless since they hung up during recording */
2241 ast_filedelete(pa->privintro, NULL);
2242 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2243 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2244 else
2245 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2246 return -1;
2247 }
2248 if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
2249 ast_waitstream(chan, "");
2250 }
2251 }
2252 return 1; /* success */
2253}

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_strdupa, ast_streamfile(), ast_strlen_zero(), ast_test_flag64, ast_verb, ast_waitstream(), LOG_NOTICE, LOG_WARNING, NULL, 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().

◆ topology_ds_destroy()

static void topology_ds_destroy ( void *  data)
static

Definition at line 827 of file app_dial.c.

827 {
828 struct ast_stream_topology *top = data;
829 ast_stream_topology_free(top);
830}

References ast_stream_topology_free().

◆ unload_module()

static int unload_module ( void  )
static

Definition at line 3613 of file app_dial.c.

3614{
3615 int res;
3616
3617 res = ast_unregister_application(app);
3618 res |= ast_unregister_application(rapp);
3619
3620 return res;
3621}

References app, ast_unregister_application(), and rapp.

◆ update_connected_line_from_peer()

static void update_connected_line_from_peer ( struct ast_channel *  chan,
struct ast_channel *  peer,
int  is_caller 
)
static

Definition at line 1176 of file app_dial.c.

1177{
1178 struct ast_party_connected_line connected_caller;
1179
1180 ast_party_connected_line_init(&connected_caller);
1181
1182 ast_channel_lock(peer);
1183 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1184 ast_channel_unlock(peer);
1185 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1186 if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)) {
1187 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1188 }
1189 ast_party_connected_line_free(&connected_caller);
1190}

References ast_channel_caller(), ast_channel_connected_line_sub(), ast_channel_lock, ast_channel_unlock, ast_channel_update_connected_line(), ast_connected_line_copy_from_caller(), AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER, ast_party_connected_line_free(), ast_party_connected_line_init(), NULL, and ast_party_connected_line::source.

Referenced by wait_for_answer().

◆ valid_priv_reply()

static int valid_priv_reply ( struct ast_flags64 *  opts,
int  res 
)
static

Definition at line 2009 of file app_dial.c.

2010{
2011 if (res < '1')
2012 return 0;
2013 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
2014 return 1;
2015 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
2016 return 1;
2017 return 0;
2018}

References ast_test_flag64, OPT_PRIVACY, and OPT_SCREENING.

◆ wait_for_answer()

static struct ast_channel * wait_for_answer ( struct ast_channel *  in,
struct dial_head *  out_chans,
int *  to_answer,
int *  to_progress,
struct ast_flags64 *  peerflags,
char *  opt_args[],
struct privacy_args *  pa,
const struct cause_args *  num_in,
int *  result,
char *  dtmf_progress,
char *  mf_progress,
char *  mf_wink,
char *  sf_progress,
char *  sf_wink,
const int  hearpulsing,
const int  ignore_cc,
struct ast_party_id *  forced_clid,
struct ast_party_id *  stored_clid,
struct ast_bridge_config *  config 
)
static

Definition at line 1209 of file app_dial.c.

1220{
1221 struct cause_args num = *num_in;
1222 int prestart = num.busy + num.congestion + num.nochan;
1223 int orig_answer_to = *to_answer;
1224 int orig_progress_to = *to_progress;
1225 struct ast_channel *peer = NULL;
1226 struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1227 /* single is set if only one destination is enabled */
1228 int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1229 int caller_entertained = outgoing
1230 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1231 struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1232 int cc_recall_core_id;
1233 int is_cc_recall;
1234 int cc_frame_received = 0;
1235 int num_ringing = 0;
1236 int sent_ring = 0;
1237 int sent_progress = 0, sent_wink = 0;
1238 struct timeval start = ast_tvnow();
1239 SCOPE_ENTER(3, "%s\n", ast_channel_name(in));
1240
1241 if (single) {
1242 /* Turn off hold music, etc */
1243 if (!caller_entertained) {
1244 ast_deactivate_generator(in);
1245 /* If we are calling a single channel, and not providing ringback or music, */
1246 /* then, make them compatible for in-band tone purpose */
1247 if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1248 /* If these channels can not be made compatible,
1249 * there is no point in continuing. The bridge
1250 * will just fail if it gets that far.
1251 */
1252 *to_answer = -1;
1253 strcpy(pa->status, "CONGESTION");
1254 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1255 SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
1256 ast_channel_name(in), ast_channel_name(outgoing->chan));
1257 }
1258 }
1259
1260 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1261 && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1262 update_connected_line_from_peer(in, outgoing->chan, 1);
1263 }
1264 }
1265
1266 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1267
1268 while ((*to_answer = ast_remaining_ms(start, orig_answer_to)) && (*to_progress = ast_remaining_ms(start, orig_progress_to)) && !peer) {
1269 struct chanlist *o;
1270 int pos = 0; /* how many channels do we handle */
1271 int numlines = prestart;
1272 struct ast_channel *winner;
1273 struct ast_channel *watchers[AST_MAX_WATCHERS];
1274
1275 watchers[pos++] = in;
1276 AST_LIST_TRAVERSE(out_chans, o, node) {
1277 /* Keep track of important channels */
1278 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1279 watchers[pos++] = o->chan;
1280 numlines++;
1281 }
1282 if (pos == 1) { /* only the input channel is available */
1283 if (numlines == (num.busy + num.congestion + num.nochan)) {
1284 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1285 if (num.busy)
1286 strcpy(pa->status, "BUSY");
1287 else if (num.congestion)
1288 strcpy(pa->status, "CONGESTION");
1289 else if (num.nochan)
1290 strcpy(pa->status, "CHANUNAVAIL");
1291 } else {
1292 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1293 }
1294 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1295 if (is_cc_recall) {
1296 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1297 }
1298 SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
1299 }
1300
1301 /* If progress timeout is active, use that if it's the shorter of the 2 timeouts. */
1302 winner = ast_waitfor_n(watchers, pos, *to_progress > 0 && (*to_answer < 0 || *to_progress < *to_answer) ? to_progress : to_answer);
1303
1304 AST_LIST_TRAVERSE(out_chans, o, node) {
1305 int res = 0;
1306 struct ast_frame *f;
1307 struct ast_channel *c = o->chan;
1308
1309 if (c == NULL)
1310 continue;
1311 if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1312 if (!peer) {
1313 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1314 if (o->orig_chan_name
1315 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1316 /*
1317 * The channel name changed so we must generate COLP update.
1318 * Likely because a call pickup channel masqueraded in.
1319 */
1320 update_connected_line_from_peer(in, c, 1);
1321 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1322 if (o->pending_connected_update) {
1323 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1324 ast_channel_update_connected_line(in, &o->connected, NULL);
1325 }
1326 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1327 update_connected_line_from_peer(in, c, 1);
1328 }
1329 }
1330 if (o->aoc_s_rate_list) {
1331 size_t encoded_size;
1332 struct ast_aoc_encoded *encoded;
1333 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1334 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1335 ast_aoc_destroy_encoded(encoded);
1336 }
1337 }
1338 peer = c;
1339 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1340 ast_copy_flags64(peerflags, o,
1341 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1342 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1343 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1344 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1345 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1346 DIAL_NOFORWARDHTML);
1347 ast_channel_dialcontext_set(c, "");
1348 ast_channel_exten_set(c, "");
1349 }
1350 continue;
1351 }
1352 if (c != winner)
1353 continue;
1354 /* here, o->chan == c == winner */
1355 if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1356 pa->sentringing = 0;
1357 if (!ignore_cc && (f = ast_read(c))) {
1358 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1359 /* This channel is forwarding the call, and is capable of CC, so
1360 * be sure to add the new device interface to the list
1361 */
1362 ast_handle_cc_control_frame(in, c, f->data.ptr);
1363 }
1364 ast_frfree(f);
1365 }
1366
1367 if (o->pending_connected_update) {
1368 /*
1369 * Re-seed the chanlist's connected line information with
1370 * previously acquired connected line info from the incoming
1371 * channel. The previously acquired connected line info could
1372 * have been set through the CONNECTED_LINE dialplan function.
1373 */
1374 o->pending_connected_update = 0;
1375 ast_channel_lock(in);
1376 ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1377 ast_channel_unlock(in);
1378 }
1379
1380 do_forward(o, &num, peerflags, single, caller_entertained, &orig_answer_to,
1381 forced_clid, stored_clid);
1382
1383 if (o->chan) {
1384 ast_free(o->orig_chan_name);
1385 o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1386 if (single
1387 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1388 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1389 update_connected_line_from_peer(in, o->chan, 1);
1390 }
1391 }
1392 continue;
1393 }
1394 f = ast_read(winner);
1395 if (!f) {
1396 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1397 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1398 ast_hangup(c);
1399 c = o->chan = NULL;
1400 ast_clear_flag64(o, DIAL_STILLGOING);
1401 handle_cause(ast_channel_hangupcause(in), &num);
1402 continue;
1403 }
1404 switch (f->frametype) {
1405 case AST_FRAME_CONTROL:
1406 switch (f->subclass.integer) {
1407 case AST_CONTROL_ANSWER:
1408 /* This is our guy if someone answered. */
1409 if (!peer) {
1410 ast_trace(-1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1411 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1412 if (o->orig_chan_name
1413 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1414 /*
1415 * The channel name changed so we must generate COLP update.
1416 * Likely because a call pickup channel masqueraded in.
1417 */
1418 update_connected_line_from_peer(in, c, 1);
1419 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1420 if (o->pending_connected_update) {
1421 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1422 ast_channel_update_connected_line(in, &o->connected, NULL);
1423 }
1424 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1425 update_connected_line_from_peer(in, c, 1);
1426 }
1427 }
1428 if (o->aoc_s_rate_list) {
1429 size_t encoded_size;
1430 struct ast_aoc_encoded *encoded;
1431 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1432 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1433 ast_aoc_destroy_encoded(encoded);
1434 }
1435 }
1436 peer = c;
1437 /* Answer can optionally include a topology */
1438 if (f->subclass.topology) {
1439 /*
1440 * We need to bump the refcount on the topology to prevent it
1441 * from being cleaned up when the frame is cleaned up.
1442 */
1443 config->answer_topology = ao2_bump(f->subclass.topology);
1444 ast_trace(-1, "%s Found topology in frame: %p %p %s\n",
1445 ast_channel_name(peer), f, config->answer_topology,
1446 ast_str_tmp(256, ast_stream_topology_to_str(config->answer_topology, &STR_TMP)));
1447 }
1448
1449 /* Inform everyone else that they've been canceled.
1450 * The dial end event for the peer will be sent out after
1451 * other Dial options have been handled.
1452 */
1453 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1454 ast_copy_flags64(peerflags, o,
1455 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1456 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1457 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1458 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1459 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1460 DIAL_NOFORWARDHTML);
1461 ast_channel_dialcontext_set(c, "");
1462 ast_channel_exten_set(c, "");
1463 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1464 /* Setup early bridge if appropriate */
1465 ast_channel_early_bridge(in, peer);
1466 }
1467 }
1468 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1469 ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1470 ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1471 break;
1472 case AST_CONTROL_BUSY:
1473 ast_verb(3, "%s is busy\n", ast_channel_name(c));
1474 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1475 ast_channel_publish_dial(in, c, NULL, "BUSY");
1476 ast_hangup(c);
1477 c = o->chan = NULL;
1478 ast_clear_flag64(o, DIAL_STILLGOING);
1479 handle_cause(AST_CAUSE_BUSY, &num);
1480 break;
1481 case AST_CONTROL_CONGESTION:
1482 ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1483 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1484 ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1485 ast_hangup(c);
1486 c = o->chan = NULL;
1487 ast_clear_flag64(o, DIAL_STILLGOING);
1488 handle_cause(AST_CAUSE_CONGESTION, &num);
1489 break;
1490 case AST_CONTROL_RINGING:
1491 /* This is a tricky area to get right when using a native
1492 * CC agent. The reason is that we do the best we can to send only a
1493 * single ringing notification to the caller.
1494 *
1495 * Call completion complicates the logic used here. CCNR is typically
1496 * offered during a ringing message. Let's say that party A calls
1497 * parties B, C, and D. B and C do not support CC requests, but D
1498 * does. If we were to receive a ringing notification from B before
1499 * the others, then we would end up sending a ringing message to
1500 * A with no CCNR offer present.
1501 *
1502 * The approach that we have taken is that if we receive a ringing
1503 * response from a party and no CCNR offer is present, we need to
1504 * wait. Specifically, we need to wait until either a) a called party
1505 * offers CCNR in its ringing response or b) all called parties have
1506 * responded in some way to our call and none offers CCNR.
1507 *
1508 * The drawback to this is that if one of the parties has a delayed
1509 * response or, god forbid, one just plain doesn't respond to our
1510 * outgoing call, then this will result in a significant delay between
1511 * when the caller places the call and hears ringback.
1512 *
1513 * Note also that if CC is disabled for this call, then it is perfectly
1514 * fine for ringing frames to get sent through.
1515 */
1516 ++num_ringing;
1517 *to_progress = -1;
1518 orig_progress_to = -1;
1519 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1520 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1521 /* Setup early media if appropriate */
1522 if (single && !caller_entertained
1523 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1524 ast_channel_early_bridge(in, c);
1525 }
1526 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1527 ast_indicate(in, AST_CONTROL_RINGING);
1528 pa->sentringing++;
1529 }
1530 if (!sent_ring) {
1531 struct timeval now, then;
1532 int64_t diff;
1533
1534 now = ast_tvnow();
1535
1536 ast_channel_lock(in);
1537 ast_channel_stage_snapshot(in);
1538
1539 then = ast_channel_creationtime(c);
1540 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1541 set_duration_var(in, "RINGTIME", diff);
1542
1543 ast_channel_stage_snapshot_done(in);
1544 ast_channel_unlock(in);
1545 sent_ring = 1;
1546 }
1547 }
1548 ast_channel_publish_dial(in, c, NULL, "RINGING");
1549 break;
1550 case AST_CONTROL_PROGRESS:
1551 ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1552 /* Setup early media if appropriate */
1553 if (single && !caller_entertained
1554 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1555 ast_channel_early_bridge(in, c);
1556 }
1557 if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1558 if (single || (!single && !pa->sentringing)) {
1559 ast_indicate(in, AST_CONTROL_PROGRESS);
1560 }
1561 }
1562 *to_progress = -1;
1563 orig_progress_to = -1;
1564 if (!sent_progress) {
1565 struct timeval now, then;
1566 int64_t diff;
1567
1568 now = ast_tvnow();
1569
1570 ast_channel_lock(in);
1571 ast_channel_stage_snapshot(in);
1572
1573 then = ast_channel_creationtime(c);
1574 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1575 set_duration_var(in, "PROGRESSTIME", diff);
1576
1577 ast_channel_stage_snapshot_done(in);
1578 ast_channel_unlock(in);
1579 sent_progress = 1;
1580
1581 if (!ast_strlen_zero(mf_progress)) {
1582 ast_verb(3,
1583 "Sending MF '%s' to %s as result of "
1584 "receiving a PROGRESS message.\n",
1585 mf_progress, hearpulsing ? "parties" : "called party");
1586 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1587 (hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
1588 }
1589 if (!ast_strlen_zero(sf_progress)) {
1590 ast_verb(3,
1591 "Sending SF '%s' to %s as result of "
1592 "receiving a PROGRESS message.\n",
1593 sf_progress, (hearpulsing ? "parties" : "called party"));
1594 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1595 (hearpulsing ? in : NULL), sf_progress, 0, 0);
1596 }
1597 if (!ast_strlen_zero(dtmf_progress)) {
1598 ast_verb(3,
1599 "Sending DTMF '%s' to the called party as result of "
1600 "receiving a PROGRESS message.\n",
1601 dtmf_progress);
1602 res |= ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1603 }
1604 if (res) {
1605 ast_log(LOG_WARNING, "Called channel %s hung up post-progress before all digits could be sent\n", ast_channel_name(c));
1606 if (ast_channel_state(c) == AST_STATE_UP) {
1607 /* The called channel answered while we were sending it digits, so the answer never got processed by app_dial.
1608 * The channel is dying now, but better to answer late than never? */
1609 ast_debug(1, "Channel %s answered while we were sending it digits, answering %s retroactively\n", ast_channel_name(c), ast_channel_name(in));
1610 /* Indicate answer supervision to the caller before we exit.
1611 * We're not going to bridge, but this way at least the CDRs are correct, etc. */
1612 ast_raw_answer(in);
1613 strcpy(pa->status, "ANSWER");
1614 } else {
1615 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1616 }
1617 goto wait_over;
1618 }
1619 }
1620 ast_channel_publish_dial(in, c, NULL, "PROGRESS");
1621 break;
1622 case AST_CONTROL_WINK:
1623 ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1624 if (!sent_wink) {
1625 sent_wink = 1;
1626 if (!ast_strlen_zero(mf_wink)) {
1627 ast_verb(3,
1628 "Sending MF '%s' to %s as result of "
1629 "receiving a WINK message.\n",
1630 mf_wink, (hearpulsing ? "parties" : "called party"));
1631 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1632 (hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
1633 }
1634 if (!ast_strlen_zero(sf_wink)) {
1635 ast_verb(3,
1636 "Sending SF '%s' to %s as result of "
1637 "receiving a WINK message.\n",
1638 sf_wink, (hearpulsing ? "parties" : "called party"));
1639 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1640 (hearpulsing ? in : NULL), sf_wink, 0, 0);
1641 }
1642 if (res) {
1643 ast_log(LOG_WARNING, "Called channel %s hung up post-wink before all digits could be sent\n", ast_channel_name(c));
1644 if (ast_channel_state(c) == AST_STATE_UP) {
1645 /* Same as in AST_CONTROL_PROGRESS */
1646 ast_debug(1, "Channel %s answered while we were sending it digits, answering %s retroactively\n", ast_channel_name(c), ast_channel_name(in));
1647 ast_raw_answer(in);
1648 strcpy(pa->status, "ANSWER");
1649 } else {
1650 *to_answer = 0; /* Continue in the dialplan, since nobody answered */
1651 }
1652 goto wait_over;
1653 }
1654 }
1655 ast_indicate(in, AST_CONTROL_WINK);
1656 break;
1657 case AST_CONTROL_VIDUPDATE:
1658 case AST_CONTROL_SRCUPDATE:
1659 case AST_CONTROL_SRCCHANGE:
1660 if (!single || caller_entertained) {
1661 break;
1662 }
1663 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1664 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1665 ast_indicate(in, f->subclass.integer);
1666 break;
1667 case AST_CONTROL_CONNECTED_LINE:
1668 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1669 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1670 break;
1671 }
1672 if (!single) {
1673 struct ast_party_connected_line connected;
1674
1675 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1676 ast_channel_name(c), ast_channel_name(in));
1677 ast_party_connected_line_set_init(&connected, &o->connected);
1678 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1679 ast_party_connected_line_set(&o->connected, &connected, NULL);
1680 ast_party_connected_line_free(&connected);
1681 o->pending_connected_update = 1;
1682 break;
1683 }
1684 if (ast_channel_connected_line_sub(c, in, f, 1)) {
1685 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1686 }
1687 break;
1688 case AST_CONTROL_AOC:
1689 {
1690 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1691 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1692 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1693 o->aoc_s_rate_list = decoded;
1694 } else {
1695 ast_aoc_destroy_decoded(decoded);
1696 }
1697 }
1698 break;
1699 case AST_CONTROL_REDIRECTING:
1700 if (!single) {
1701 /*
1702 * Redirecting updates to the caller make sense only on single
1703 * calls.
1704 */
1705 break;
1706 }
1707 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1708 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1709 break;
1710 }
1711 ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1712 ast_channel_name(c), ast_channel_name(in));
1713 if (ast_channel_redirecting_sub(c, in, f, 1)) {
1714 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1715 }
1716 pa->sentringing = 0;
1717 break;
1718 case AST_CONTROL_PROCEEDING:
1719 ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1720 if (single && !caller_entertained
1721 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1722 ast_channel_early_bridge(in, c);
1723 }
1724 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1725 ast_indicate(in, AST_CONTROL_PROCEEDING);
1726 ast_channel_publish_dial(in, c, NULL, "PROCEEDING");
1727 break;
1728 case AST_CONTROL_HOLD:
1729 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1730 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1731 ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1732 break;
1733 case AST_CONTROL_UNHOLD:
1734 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1735 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1736 ast_indicate(in, AST_CONTROL_UNHOLD);
1737 break;
1738 case AST_CONTROL_OFFHOOK:
1739 case AST_CONTROL_FLASH:
1740 /* Ignore going off hook and flash */
1741 break;
1742 case AST_CONTROL_CC:
1743 if (!ignore_cc) {
1744 ast_handle_cc_control_frame(in, c, f->data.ptr);
1745 cc_frame_received = 1;
1746 }
1747 break;
1748 case AST_CONTROL_PVT_CAUSE_CODE:
1749 ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1750 break;
1751 case AST_CONTROL_PLAYBACK_BEGIN:
1752 if (!f->data.ptr) {
1753 ast_log(LOG_WARNING, "Got playback begin directive without filename on %s\n", ast_channel_name(c));
1754 } else {
1755 const char *filename = f->data.ptr;
1756 ast_verb(3, "Playing audio file %s on %s\n", filename, ast_channel_name(in));
1757 ast_streamfile(in, filename, ast_channel_language(in));
1758 }
1759 break;
1760 case -1:
1761 if (single && !caller_entertained) {
1762 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1763 ast_indicate(in, -1);
1764 pa->sentringing = 0;
1765 }
1766 break;
1767 default:
1768 ast_debug(1, "Dunno what to do with control type %d on %s\n", f->subclass.integer, ast_channel_name(in));
1769 break;
1770 }
1771 break;
1772 case AST_FRAME_VIDEO:
1773 case AST_FRAME_VOICE:
1774 case AST_FRAME_IMAGE:
1775 case AST_FRAME_DTMF_BEGIN:
1776 case AST_FRAME_DTMF_END:
1777 if (caller_entertained) {
1778 break;
1779 }
1780 *to_progress = -1;
1781 orig_progress_to = -1;
1782 /* Fall through */
1783 case AST_FRAME_TEXT:
1784 if (single && ast_write(in, f)) {
1785 ast_log(LOG_WARNING, "Unable to write frametype %u on %s\n",
1786 f->frametype, ast_channel_name(in));
1787 }
1788 break;
1789 case AST_FRAME_HTML:
1790 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1791 && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1792 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(in));
1793 }
1794 break;
1795 default:
1796 break;
1797 }
1798 ast_frfree(f);
1799 } /* end for */
1800 if (winner == in) {
1801 struct ast_frame *f = ast_read(in);
1802 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1803 /* Got hung up */
1804 *to_answer = -1;
1805 strcpy(pa->status, "CANCEL");
1806 pa->canceled = 1;
1807 publish_dial_end_event(in, out_chans, NULL, pa->status);
1808 if (f) {
1809 if (f->data.uint32) {
1810 ast_channel_hangupcause_set(in, f->data.uint32);
1811 }
1812 ast_frfree(f);
1813 }
1814 if (is_cc_recall) {
1815 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1816 }
1817 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller hung up\n", ast_channel_name(in));
1818 }
1819
1820 /* now f is guaranteed non-NULL */
1821 if (f->frametype == AST_FRAME_DTMF) {
1822 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1823 const char *context;
1824 ast_channel_lock(in);
1825 if ((context = pbx_builtin_getvar_helper(in, "EXITCONTEXT"))) {
1826 context = ast_strdupa(context);
1827 }
1828 ast_channel_unlock(in);
1829 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1830 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1831 *to_answer = 0;
1832 *result = f->subclass.integer;
1833 strcpy(pa->status, "CANCEL");
1834 pa->canceled = 1;
1835 publish_dial_end_event(in, out_chans, NULL, pa->status);
1836 ast_frfree(f);
1837 if (is_cc_recall) {
1838 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1839 }
1840 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller pressed %c to end call\n",
1841 ast_channel_name(in), f->subclass.integer);
1842 }
1843 }
1844
1845 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1846 detect_disconnect(in, f->subclass.integer, &featurecode)) {
1847 ast_verb(3, "User requested call disconnect.\n");
1848 *to_answer = 0;
1849 strcpy(pa->status, "CANCEL");
1850 pa->canceled = 1;
1851 publish_dial_end_event(in, out_chans, NULL, pa->status);
1852 ast_frfree(f);
1853 if (is_cc_recall) {
1854 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1855 }
1856 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller requested disconnect\n",
1857 ast_channel_name(in));
1858 }
1859 }
1860
1861 /* Send the frame from the in channel to all outgoing channels. */
1862 AST_LIST_TRAVERSE(out_chans, o, node) {
1863 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1864 /* This outgoing channel has died so don't send the frame to it. */
1865 continue;
1866 }
1867 switch (f->frametype) {
1868 case AST_FRAME_HTML:
1869 /* Forward HTML stuff */
1870 if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1871 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1872 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(o->chan));
1873 }
1874 break;
1875 case AST_FRAME_VIDEO:
1876 case AST_FRAME_VOICE:
1877 case AST_FRAME_IMAGE:
1878 if (!single || caller_entertained) {
1879 /*
1880 * We are calling multiple parties or caller is being
1881 * entertained and has thus not been made compatible.
1882 * No need to check any other called parties.
1883 */
1884 goto skip_frame;
1885 }
1886 /* Fall through */
1887 case AST_FRAME_TEXT:
1888 case AST_FRAME_DTMF_BEGIN:
1889 case AST_FRAME_DTMF_END:
1890 if (ast_write(o->chan, f)) {
1891 ast_log(LOG_WARNING, "Unable to forward frametype %u on %s\n",
1892 f->frametype, ast_channel_name(o->chan));
1893 }
1894 break;
1895 case AST_FRAME_CONTROL:
1896 switch (f->subclass.integer) {
1897 case AST_CONTROL_HOLD:
1898 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1899 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1900 break;
1901 case AST_CONTROL_UNHOLD:
1902 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1903 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1904 break;
1905 case AST_CONTROL_FLASH:
1906 ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
1907 ast_indicate(o->chan, AST_CONTROL_FLASH);
1908 break;
1909 case AST_CONTROL_VIDUPDATE:
1910 case AST_CONTROL_SRCUPDATE:
1911 case AST_CONTROL_SRCCHANGE:
1912 if (!single || caller_entertained) {
1913 /*
1914 * We are calling multiple parties or caller is being
1915 * entertained and has thus not been made compatible.
1916 * No need to check any other called parties.
1917 */
1918 goto skip_frame;
1919 }
1920 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1921 ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1922 ast_indicate(o->chan, f->subclass.integer);
1923 break;
1924 case AST_CONTROL_CONNECTED_LINE:
1925 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1926 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
1927 break;
1928 }
1929 if (ast_channel_connected_line_sub(in, o->chan, f, 1)) {
1930 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1931 }
1932 break;
1933 case AST_CONTROL_REDIRECTING:
1934 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1935 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
1936 break;
1937 }
1938 if (ast_channel_redirecting_sub(in, o->chan, f, 1)) {
1939 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1940 }
1941 break;
1942 default:
1943 /* We are not going to do anything with this frame. */
1944 goto skip_frame;
1945 }
1946 break;
1947 default:
1948 /* We are not going to do anything with this frame. */
1949 goto skip_frame;
1950 }
1951 }
1952skip_frame:;
1953 ast_frfree(f);
1954 }
1955 }
1956
1957wait_over:
1958 if (!*to_answer || ast_check_hangup(in)) {
1959 if (orig_answer_to != -1) {
1960 ast_verb(3, "Nobody picked up in %d ms\n", orig_answer_to);
1961 } else {
1962 ast_verb(3, "Call terminated without answer\n");
1963 }
1964 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1965 } else if (!*to_progress) {
1966 ast_verb(3, "No early media received in %d ms\n", orig_progress_to);
1967 publish_dial_end_event(in, out_chans, NULL, "CHANUNAVAIL");
1968 strcpy(pa->status, "CHANUNAVAIL");
1969 *to_answer = 0; /* Reset to prevent hangup */
1970 }
1971
1972 if (is_cc_recall) {
1973 ast_cc_completed(in, "Recall completed!");
1974 }
1975 SCOPE_EXIT_RTN_VALUE(peer, "%s: %s%s\n", ast_channel_name(in),
1976 peer ? "Answered by " : "No answer", peer ? ast_channel_name(peer) : "");
1977}

References ao2_bump, 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_channel_call_forward(), ast_channel_connected(), ast_channel_connected_line_sub(), ast_channel_creationtime(), ast_channel_early_bridge(), ast_channel_exten_set(), ast_channel_hangupcause(), ast_channel_hangupcause_set(), ast_channel_language(), ast_channel_lock, ast_channel_make_compatible(), ast_channel_name(), ast_channel_publish_dial(), ast_channel_redirecting_sub(), ast_channel_sendhtml(), ast_channel_stage_snapshot(), ast_channel_stage_snapshot_done(), ast_channel_unlock, ast_channel_update_connected_line(), ast_check_hangup(), ast_clear_flag64, ast_connected_line_parse_data(), 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_PLAYBACK_BEGIN, 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_CONTROL_WINK, ast_copy_flags64, ast_deactivate_generator(), ast_debug, ast_dtmf_stream(), AST_FEATURE_MAX_LEN, 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_VIDEO, AST_FRAME_VOICE, ast_free, ast_frfree, ast_handle_cc_control_frame(), ast_hangup(), ast_hangup_cause_to_dial_status(), ast_indicate(), ast_indicate_data(), AST_LIST_FIRST, AST_LIST_NEXT, AST_LIST_TRAVERSE, ast_log, AST_MAX_WATCHERS, ast_mf_stream(), ast_party_connected_line_copy(), ast_party_connected_line_free(), ast_party_connected_line_set(), ast_party_connected_line_set_init(), ast_raw_answer(), ast_read(), ast_remaining_ms(), ast_sf_stream(), AST_STATE_UP, ast_str_alloca, ast_str_tmp, ast_strdup, ast_strdupa, ast_stream_topology_to_str(), ast_streamfile(), ast_strlen_zero(), ast_test_flag64, ast_trace, ast_tvdiff_ms(), ast_tvnow(), ast_tvzero(), ast_verb, ast_waitfor_n(), ast_write(), cause_args::busy, c, CAN_EARLY_BRIDGE, privacy_args::canceled, chanlist::chan, config, cause_args::congestion, chanlist::connected, connected, ast_frame::data, ast_frame::datalen, detect_disconnect(), DIAL_CALLERID_ABSENT, DIAL_NOFORWARDHTML, DIAL_STILLGOING, do_forward(), ast_frame::frametype, handle_cause(), in, ast_frame_subclass::integer, LOG_WARNING, cause_args::nochan, NULL, 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, chanlist::orig_chan_name, pbx_builtin_getvar_helper(), chanlist::pending_connected_update, ast_frame::ptr, publish_dial_end_event(), result, SCOPE_ENTER, SCOPE_EXIT_RTN_VALUE, privacy_args::sentringing, set_duration_var(), privacy_args::status, ast_frame::subclass, ast_frame_subclass::topology, ast_frame::uint32, and update_connected_line_from_peer().

Referenced by dial_exec_full().

Variable Documentation

◆ __mod_info

struct ast_module_info __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_DEFAULT , .description = "Dialing Application" , .key = ASTERISK_GPL_KEY , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, }
static

Definition at line 3637 of file app_dial.c.

◆ app

const char app[] = "Dial"
static

Definition at line 670 of file app_dial.c.

Referenced by load_module(), and unload_module().

◆ ast_module_info

const struct ast_module_info* ast_module_info = &__mod_info
static

Definition at line 3637 of file app_dial.c.

◆ dial_exec_options

const struct ast_app_option dial_exec_options[128] = { [ 'A' ] = { .flag = OPT_ANNOUNCE , .arg_index = OPT_ARG_ANNOUNCE + 1 }, [ 'a' ] = { .flag = OPT_CALLER_ANSWER }, [ 'b' ] = { .flag = OPT_PREDIAL_CALLEE , .arg_index = OPT_ARG_PREDIAL_CALLEE + 1 }, [ 'B' ] = { .flag = OPT_PREDIAL_CALLER , .arg_index = OPT_ARG_PREDIAL_CALLER + 1 }, [ 'C' ] = { .flag = OPT_RESETCDR }, [ 'c' ] = { .flag = OPT_CANCEL_ELSEWHERE }, [ 'd' ] = { .flag = OPT_DTMF_EXIT }, [ 'D' ] = { .flag = OPT_SENDDTMF , .arg_index = OPT_ARG_SENDDTMF + 1 }, [ 'E' ] = { .flag = OPT_HEARPULSING }, [ 'e' ] = { .flag = OPT_PEER_H }, [ 'f' ] = { .flag = OPT_FORCECLID , .arg_index = OPT_ARG_FORCECLID + 1 }, [ 'F' ] = { .flag = OPT_CALLEE_GO_ON , .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 }, [ 'j' ] = { .flag = OPT_TOPOLOGY_PRESERVE }, [ '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 }, [ '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 }, [ 'Q' ] = { .flag = OPT_HANGUPCAUSE , .arg_index = OPT_ARG_HANGUPCAUSE + 1 }, [ 'r' ] = { .flag = OPT_RINGBACK , .arg_index = OPT_ARG_RINGBACK + 1 }, [ 'R' ] = { .flag = OPT_RING_WITH_EARLY_MEDIA }, [ 'S' ] = { .flag = OPT_DURATION_STOP , .arg_index = OPT_ARG_DURATION_STOP + 1 }, [ 's' ] = { .flag = OPT_FORCE_CID_TAG , .arg_index = OPT_ARG_FORCE_CID_TAG + 1 }, [ 't' ] = { .flag = OPT_CALLEE_TRANSFER }, [ 'T' ] = { .flag = OPT_CALLER_TRANSFER }, [ 'u' ] = { .flag = OPT_FORCE_CID_PRES , .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 = OPT_CANCEL_TIMEOUT }, }
static

Definition at line 792 of file app_dial.c.

Referenced by dial_exec_full().

◆ rapp

const char rapp[] = "RetryDial"
static

Definition at line 671 of file app_dial.c.

Referenced by load_module(), retrydial_exec(), and unload_module().

◆ topology_ds_info

const struct ast_datastore_info topology_ds_info
static
Initial value:
= {
.type = "app_dial_topology_preserve",
.destroy = topology_ds_destroy,
}

Definition at line 832 of file app_dial.c.

832 {
833 .type = "app_dial_topology_preserve",
834 .destroy = topology_ds_destroy,
835};

Referenced by dial_exec_full().

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