Wed Aug 18 22:34:36 2010

Asterisk developer's documentation


Module: Dial plan applications

Applications support the dialplan. They register dynamically with. More...


Files

file  app_adsiprog.c
 Program Asterisk ADSI Scripts into phone.
file  app_alarmreceiver.c
 Central Station Alarm receiver for Ademco Contact ID.
file  app_authenticate.c
 Execute arbitrary authenticate commands.
file  app_cdr.c
 Applications connected with CDR engine.
file  app_chanisavail.c
 Check if Channel is Available.
file  app_channelredirect.c
 ChannelRedirect application.
file  app_chanspy.c
 ChanSpy: Listen in on any channel.
file  app_controlplayback.c
 Trivial application to control playback of a sound file.
file  app_dahdibarge.c
 DAHDI Barge support.
file  app_dahdiras.c
 Execute an ISDN RAS.
file  app_dahdiscan.c
 DAHDI Scanner.
file  app_db.c
 Database access functions.
file  app_dial.c
 dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
file  app_dictate.c
 Virtual Dictation Machine Application For Asterisk.
file  app_directed_pickup.c
 Directed Call Pickup Support.
file  app_directory.c
 Provide a directory of extensions.
file  app_disa.c
 DISA -- Direct Inward System Access Application.
file  app_dumpchan.c
 Application to dump channel variables.
file  app_echo.c
 Echo application -- play back what you hear to evaluate latency.
file  app_exec.c
 Exec application.
file  app_externalivr.c
 External IVR application interface.
file  app_festival.c
 Connect to festival.
file  app_flash.c
 App to flash a DAHDI trunk.
file  app_followme.c
 Find-Me Follow-Me application.
file  app_forkcdr.c
 Fork CDR application.
file  app_getcpeid.c
 Get ADSI CPE ID.
file  app_ices.c
 Stream to an icecast server via ICES (see contrib/asterisk-ices.xml).
file  app_image.c
 App to transmit an image.
file  app_ivrdemo.c
 IVR Demo application.
file  app_jack.c
 Jack Application.
file  app_macro.c
 Dial plan macro Implementation.
file  app_meetme.c
 Meet me conference bridge and Shared Line Appearances.
file  app_milliwatt.c
 Digital Milliwatt Test.
file  app_minivm.c
 MiniVoiceMail - A Minimal Voicemail System for Asterisk.
file  app_mixmonitor.c
 MixMonitor() - Record a call and mix the audio during the recording.
file  app_morsecode.c
 Morsecode application.
file  app_mp3.c
 Silly application to play an MP3 file -- uses mpg123.
file  app_nbscat.c
 Silly application to play an NBScat file -- uses nbscat8k.
file  app_osplookup.c
 Open Settlement Protocol (OSP) Applications.
file  app_page.c
 page() - Paging application
file  app_parkandannounce.c
 ParkAndAnnounce application for Asterisk.
file  app_playback.c
 Trivial application to playback a sound file.
file  app_privacy.c
 Block all calls without Caller*ID, require phone # to be entered.
file  app_queue.c
 True call queues with optional send URL on answer.
file  app_read.c
 Trivial application to read a variable.
file  app_readexten.c
 Trivial application to read an extension into a variable.
file  app_readfile.c
 ReadFile application -- Reads in a File for you.
file  app_record.c
 Trivial application to record a sound file.
file  app_sayunixtime.c
 SayUnixTime application.
file  app_senddtmf.c
 App to send DTMF digits.
file  app_sendtext.c
 App to transmit a text message.
file  app_setcallerid.c
 App to set callerid presentation.
file  app_skel.c
 Skeleton application.
file  app_sms.c
 SMS application - ETSI ES 201 912 protocol 1 implementation.
file  app_softhangup.c
 SoftHangup application.
file  app_speech_utils.c
 Speech Recognition Utility Applications.
file  app_stack.c
 Stack applications Gosub, Return, etc.
file  app_system.c
 Execute arbitrary system commands.
file  app_talkdetect.c
 Playback a file with audio detect.
file  app_test.c
 Applications to test connection and produce report in text file.
file  app_transfer.c
 Transfer a caller.
file  app_url.c
 App to transmit a URL.
file  app_userevent.c
 UserEvent application -- send manager event.
file  app_verbose.c
 Verbose logging application.
file  app_voicemail.c
 Comedian Mail - Voicemail System.
file  app_waitforring.c
 Wait for Ring Application.
file  app_waitforsilence.c
 Wait for Silence
  • Waits for up to 'x' milliseconds of silence, 'y' times
  • WaitForSilence(500,2) will wait for 1/2 second of silence, twice
  • WaitForSilence(1000,1) will wait for 1 second of silence, once
  • WaitForSilence(300,3,10) will wait for 300ms of silence, 3 times, and return after 10sec For Noise The same as Wait For Silence but listenes noise on the chennel that is above
    the pre-configured silence threshold from dsp.conf.

file  app_waituntil.c
 Sleep until the given epoch.
file  app_while.c
 While Loop Implementation.
file  app_zapateller.c
 Playback the special information tone to get rid of telemarketers.
file  res_ael_share.c
 Shareable AEL code -- mainly between internal and external modules.
file  res_realtime.c
 RealTime CLI.

Functions

static int iax2_prov_app (struct ast_channel *chan, void *data)
static int pbx_builtin_answer (struct ast_channel *, void *)
static int pbx_builtin_background (struct ast_channel *, void *)
static int pbx_builtin_busy (struct ast_channel *, void *)
static int pbx_builtin_congestion (struct ast_channel *, void *)
static int pbx_builtin_execiftime (struct ast_channel *, void *)
static int pbx_builtin_goto (struct ast_channel *, void *)
static int pbx_builtin_gotoiftime (struct ast_channel *, void *)
static int pbx_builtin_hangup (struct ast_channel *, void *)
static int pbx_builtin_proceeding (struct ast_channel *, void *)
static int pbx_builtin_progress (struct ast_channel *, void *)
static int pbx_builtin_resetcdr (struct ast_channel *, void *)
static int pbx_builtin_ringing (struct ast_channel *, void *)
static int pbx_builtin_setamaflags (struct ast_channel *, void *)
static int pbx_builtin_wait (struct ast_channel *, void *)
static int pbx_builtin_waitexten (struct ast_channel *, void *)

Detailed Description

Applications support the dialplan. They register dynamically with.

Asterisk Dial Plan Applications

See also:
ast_register_application() and unregister with

ast_unregister_application()

See also

Function Documentation

static int iax2_prov_app ( struct ast_channel chan,
void *  data 
) [static]

iax2provision

Definition at line 11205 of file chan_iax2.c.

References ast_inet_ntoa(), ast_log(), ast_strdupa, ast_strlen_zero(), ast_verb, chan, iax2_provision(), iax2_tech, iaxs, LOG_NOTICE, PTR_TO_CALLNO, create_addr_info::sockfd, ast_channel::tech, and ast_channel::tech_pvt.

Referenced by load_module().

11206 {
11207    int res;
11208    char *sdata;
11209    char *opts;
11210    int force =0;
11211    unsigned short callno = PTR_TO_CALLNO(chan->tech_pvt);
11212    if (ast_strlen_zero(data))
11213       data = "default";
11214    sdata = ast_strdupa(data);
11215    opts = strchr(sdata, '|');
11216    if (opts)
11217       *opts='\0';
11218 
11219    if (chan->tech != &iax2_tech) {
11220       ast_log(LOG_NOTICE, "Can't provision a non-IAX device!\n");
11221       return -1;
11222    } 
11223    if (!callno || !iaxs[callno] || !iaxs[callno]->addr.sin_addr.s_addr) {
11224       ast_log(LOG_NOTICE, "Can't provision something with no IP?\n");
11225       return -1;
11226    }
11227    res = iax2_provision(&iaxs[callno]->addr, iaxs[callno]->sockfd, NULL, sdata, force);
11228    ast_verb(3, "Provisioned IAXY at '%s' with '%s'= %d\n",
11229       ast_inet_ntoa(iaxs[callno]->addr.sin_addr),
11230       sdata, res);
11231    return res;
11232 }

static int pbx_builtin_answer ( struct ast_channel ,
void *   
) [static]

Definition at line 7957 of file pbx.c.

References __ast_answer(), ast_channel::_state, AST_STATE_UP, ast_strlen_zero(), and chan.

07958 {
07959    int delay = 0;
07960 
07961    if ((chan->_state != AST_STATE_UP) && !ast_strlen_zero(data))
07962       delay = atoi(data);
07963 
07964    if (delay < 0) {
07965       delay = 0;
07966    }
07967 
07968    return __ast_answer(chan, delay, 1);
07969 }

static int pbx_builtin_background ( struct ast_channel ,
void *   
) [static]

Definition at line 8226 of file pbx.c.

References ast_channel::_state, ast_answer(), AST_APP_ARG, ast_app_parse_options(), ast_canmatch_extension(), ast_channel_lock, ast_channel_unlock, ast_copy_string(), AST_DECLARE_APP_ARGS, AST_DIGIT_ANY, AST_FLAG_DISABLE_WORKAROUNDS, ast_log(), ast_matchmore_extension(), AST_STANDARD_APP_ARGS, AST_STATE_UP, ast_stopstream(), ast_strdupa, ast_streamfile(), ast_strlen_zero(), ast_test_flag, ast_waitstream(), ast_waitstream_exten(), BACKGROUND_MATCHEXTEN, BACKGROUND_NOANSWER, background_opts, BACKGROUND_PLAYBACK, BACKGROUND_SKIP, chan, ast_channel::cid, ast_callerid::cid_num, ast_channel::context, ast_channel::exten, ast_flags::flags, ast_channel::language, LOG_WARNING, ast_channel::name, parse(), pbx_builtin_getvar_helper(), pbx_builtin_setvar_helper(), ast_channel::priority, and strsep().

08227 {
08228    int res = 0;
08229    int mres = 0;
08230    struct ast_flags flags = {0};
08231    char *parse, exten[2] = "";
08232    AST_DECLARE_APP_ARGS(args,
08233       AST_APP_ARG(filename);
08234       AST_APP_ARG(options);
08235       AST_APP_ARG(lang);
08236       AST_APP_ARG(context);
08237    );
08238 
08239    if (ast_strlen_zero(data)) {
08240       ast_log(LOG_WARNING, "Background requires an argument (filename)\n");
08241       return -1;
08242    }
08243 
08244    parse = ast_strdupa(data);
08245 
08246    AST_STANDARD_APP_ARGS(args, parse);
08247 
08248    if (ast_strlen_zero(args.lang))
08249       args.lang = (char *)chan->language; /* XXX this is const */
08250 
08251    if (ast_strlen_zero(args.context)) {
08252       const char *context;
08253       ast_channel_lock(chan);
08254       if ((context = pbx_builtin_getvar_helper(chan, "MACRO_CONTEXT"))) {
08255          args.context = ast_strdupa(context);
08256       } else {
08257          args.context = chan->context;
08258       }
08259       ast_channel_unlock(chan);
08260    }
08261 
08262    if (args.options) {
08263       if (!strcasecmp(args.options, "skip"))
08264          flags.flags = BACKGROUND_SKIP;
08265       else if (!strcasecmp(args.options, "noanswer"))
08266          flags.flags = BACKGROUND_NOANSWER;
08267       else
08268          ast_app_parse_options(background_opts, &flags, NULL, args.options);
08269    }
08270 
08271    /* Answer if need be */
08272    if (chan->_state != AST_STATE_UP) {
08273       if (ast_test_flag(&flags, BACKGROUND_SKIP)) {
08274          goto done;
08275       } else if (!ast_test_flag(&flags, BACKGROUND_NOANSWER)) {
08276          res = ast_answer(chan);
08277       }
08278    }
08279 
08280    if (!res) {
08281       char *back = args.filename;
08282       char *front;
08283 
08284       ast_stopstream(chan);      /* Stop anything playing */
08285       /* Stream the list of files */
08286       while (!res && (front = strsep(&back, "&")) ) {
08287          if ( (res = ast_streamfile(chan, front, args.lang)) ) {
08288             ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char*)data);
08289             res = 0;
08290             mres = 1;
08291             break;
08292          }
08293          if (ast_test_flag(&flags, BACKGROUND_PLAYBACK)) {
08294             res = ast_waitstream(chan, "");
08295          } else if (ast_test_flag(&flags, BACKGROUND_MATCHEXTEN)) {
08296             res = ast_waitstream_exten(chan, args.context);
08297          } else {
08298             res = ast_waitstream(chan, AST_DIGIT_ANY);
08299          }
08300          ast_stopstream(chan);
08301       }
08302    }
08303 
08304    /*
08305     * If the single digit DTMF is an extension in the specified context, then
08306     * go there and signal no DTMF.  Otherwise, we should exit with that DTMF.
08307     * If we're in Macro, we'll exit and seek that DTMF as the beginning of an
08308     * extension in the Macro's calling context.  If we're not in Macro, then
08309     * we'll simply seek that extension in the calling context.  Previously,
08310     * someone complained about the behavior as it related to the interior of a
08311     * Gosub routine, and the fix (#14011) inadvertently broke FreePBX
08312     * (#14940).  This change should fix both of these situations, but with the
08313     * possible incompatibility that if a single digit extension does not exist
08314     * (but a longer extension COULD have matched), it would have previously
08315     * gone immediately to the "i" extension, but will now need to wait for a
08316     * timeout.
08317     *
08318     * Later, we had to add a flag to disable this workaround, because AGI
08319     * users can EXEC Background and reasonably expect that the DTMF code will
08320     * be returned (see #16434).
08321     */
08322    if (!ast_test_flag(chan, AST_FLAG_DISABLE_WORKAROUNDS) &&
08323          (exten[0] = res) &&
08324          ast_canmatch_extension(chan, args.context, exten, 1, chan->cid.cid_num) &&
08325          !ast_matchmore_extension(chan, args.context, exten, 1, chan->cid.cid_num)) {
08326       snprintf(chan->exten, sizeof(chan->exten), "%c", res);
08327       ast_copy_string(chan->context, args.context, sizeof(chan->context));
08328       chan->priority = 0;
08329       res = 0;
08330    }
08331 done:
08332    pbx_builtin_setvar_helper(chan, "BACKGROUNDSTATUS", mres ? "FAILED" : "SUCCESS");
08333    return res;
08334 }

static int pbx_builtin_busy ( struct ast_channel ,
void *   
) [static]

Definition at line 7927 of file pbx.c.

References ast_channel::_state, ast_cdr_busy(), AST_CONTROL_BUSY, ast_indicate(), ast_setstate(), AST_STATE_BUSY, AST_STATE_UP, chan, and wait_for_hangup().

07928 {
07929    ast_indicate(chan, AST_CONTROL_BUSY);
07930    /* Don't change state of an UP channel, just indicate
07931       busy in audio */
07932    if (chan->_state != AST_STATE_UP) {
07933       ast_setstate(chan, AST_STATE_BUSY);
07934       ast_cdr_busy(chan->cdr);
07935    }
07936    wait_for_hangup(chan, data);
07937    return -1;
07938 }

static int pbx_builtin_congestion ( struct ast_channel ,
void *   
) [static]

Definition at line 7943 of file pbx.c.

References ast_channel::_state, AST_CONTROL_CONGESTION, ast_indicate(), ast_setstate(), AST_STATE_BUSY, AST_STATE_UP, chan, and wait_for_hangup().

07944 {
07945    ast_indicate(chan, AST_CONTROL_CONGESTION);
07946    /* Don't change state of an UP channel, just indicate
07947       congestion in audio */
07948    if (chan->_state != AST_STATE_UP)
07949       ast_setstate(chan, AST_STATE_BUSY);
07950    wait_for_hangup(chan, data);
07951    return -1;
07952 }

static int pbx_builtin_execiftime ( struct ast_channel ,
void *   
) [static]

Definition at line 8093 of file pbx.c.

References app, ast_build_timing(), ast_check_timing(), ast_log(), ast_strdupa, ast_strlen_zero(), chan, LOG_WARNING, pbx_exec(), pbx_findapp(), s, S_OR, and strsep().

08094 {
08095    char *s, *appname;
08096    struct ast_timing timing;
08097    struct ast_app *app;
08098    static const char *usage = "ExecIfTime requires an argument:\n  <time range>,<days of week>,<days of month>,<months>?<appname>[(<appargs>)]";
08099 
08100    if (ast_strlen_zero(data)) {
08101       ast_log(LOG_WARNING, "%s\n", usage);
08102       return -1;
08103    }
08104 
08105    appname = ast_strdupa(data);
08106 
08107    s = strsep(&appname, "?"); /* Separate the timerange and application name/data */
08108    if (!appname) {   /* missing application */
08109       ast_log(LOG_WARNING, "%s\n", usage);
08110       return -1;
08111    }
08112 
08113    if (!ast_build_timing(&timing, s)) {
08114       ast_log(LOG_WARNING, "Invalid Time Spec: %s\nCorrect usage: %s\n", s, usage);
08115       return -1;
08116    }
08117 
08118    if (!ast_check_timing(&timing))  /* outside the valid time window, just return */
08119       return 0;
08120 
08121    /* now split appname(appargs) */
08122    if ((s = strchr(appname, '('))) {
08123       char *e;
08124       *s++ = '\0';
08125       if ((e = strrchr(s, ')')))
08126          *e = '\0';
08127       else
08128          ast_log(LOG_WARNING, "Failed to find closing parenthesis\n");
08129    }
08130       
08131 
08132    if ((app = pbx_findapp(appname))) {
08133       return pbx_exec(chan, app, S_OR(s, ""));
08134    } else {
08135       ast_log(LOG_WARNING, "Cannot locate application %s\n", appname);
08136       return -1;
08137    }
08138 }

static int pbx_builtin_goto ( struct ast_channel chan,
void *  data 
) [static]

Goto

Definition at line 8339 of file pbx.c.

References ast_parseable_goto(), ast_verb, chan, ast_channel::context, ast_channel::exten, and ast_channel::priority.

Referenced by pbx_builtin_gotoif(), and pbx_builtin_gotoiftime().

08340 {
08341    int res = ast_parseable_goto(chan, data);
08342    if (!res)
08343       ast_verb(3, "Goto (%s,%s,%d)\n", chan->context, chan->exten, chan->priority + 1);
08344    return res;
08345 }

static int pbx_builtin_gotoiftime ( struct ast_channel ,
void *   
) [static]

Definition at line 8059 of file pbx.c.

References ast_build_timing(), ast_check_timing(), ast_debug, ast_log(), ast_strdupa, ast_strlen_zero(), chan, LOG_WARNING, pbx_builtin_goto(), s, and strsep().

08060 {
08061    char *s, *ts, *branch1, *branch2, *branch;
08062    struct ast_timing timing;
08063 
08064    if (ast_strlen_zero(data)) {
08065       ast_log(LOG_WARNING, "GotoIfTime requires an argument:\n  <time range>,<days of week>,<days of month>,<months>?'labeliftrue':'labeliffalse'\n");
08066       return -1;
08067    }
08068 
08069    ts = s = ast_strdupa(data);
08070 
08071    /* Separate the Goto path */
08072    strsep(&ts, "?");
08073    branch1 = strsep(&ts,":");
08074    branch2 = strsep(&ts,"");
08075 
08076    /* struct ast_include include contained garbage here, fixed by zeroing it on get_timerange */
08077    if (ast_build_timing(&timing, s) && ast_check_timing(&timing))
08078       branch = branch1;
08079    else
08080       branch = branch2;
08081 
08082    if (ast_strlen_zero(branch)) {
08083       ast_debug(1, "Not taking any branch\n");
08084       return 0;
08085    }
08086 
08087    return pbx_builtin_goto(chan, branch);
08088 }

static int pbx_builtin_hangup ( struct ast_channel ,
void *   
) [static]

Definition at line 8029 of file pbx.c.

References AST_CAUSE_NORMAL_CLEARING, ast_log(), ast_str2cause(), ast_strlen_zero(), chan, ast_channel::hangupcause, and LOG_WARNING.

08030 {
08031    if (!ast_strlen_zero(data)) {
08032       int cause;
08033       char *endptr;
08034 
08035       if ((cause = ast_str2cause(data)) > -1) {
08036          chan->hangupcause = cause;
08037          return -1;
08038       }
08039       
08040       cause = strtol((const char *) data, &endptr, 10);
08041       if (cause != 0 || (data != endptr)) {
08042          chan->hangupcause = cause;
08043          return -1;
08044       }
08045          
08046       ast_log(LOG_WARNING, "Invalid cause given to Hangup(): \"%s\"\n", (char *) data);
08047    }
08048 
08049    if (!chan->hangupcause) {
08050       chan->hangupcause = AST_CAUSE_NORMAL_CLEARING;
08051    }
08052 
08053    return -1;
08054 }

static int pbx_builtin_proceeding ( struct ast_channel ,
void *   
) [static]

Definition at line 7900 of file pbx.c.

References AST_CONTROL_PROCEEDING, ast_indicate(), and chan.

07901 {
07902    ast_indicate(chan, AST_CONTROL_PROCEEDING);
07903    return 0;
07904 }

static int pbx_builtin_progress ( struct ast_channel ,
void *   
) [static]

Definition at line 7909 of file pbx.c.

References AST_CONTROL_PROGRESS, ast_indicate(), and chan.

07910 {
07911    ast_indicate(chan, AST_CONTROL_PROGRESS);
07912    return 0;
07913 }

static int pbx_builtin_resetcdr ( struct ast_channel ,
void *   
) [static]

Definition at line 8001 of file pbx.c.

References ast_app_parse_options(), ast_cdr_reset(), ast_strdupa, ast_strlen_zero(), ast_channel::cdr, chan, ast_flags::flags, and resetcdr_opts.

08002 {
08003    char *args;
08004    struct ast_flags flags = { 0 };
08005 
08006    if (!ast_strlen_zero(data)) {
08007       args = ast_strdupa(data);
08008       ast_app_parse_options(resetcdr_opts, &flags, NULL, args);
08009    }
08010 
08011    ast_cdr_reset(chan->cdr, &flags);
08012 
08013    return 0;
08014 }

static int pbx_builtin_ringing ( struct ast_channel ,
void *   
) [static]

Definition at line 7918 of file pbx.c.

References AST_CONTROL_RINGING, ast_indicate(), and chan.

07919 {
07920    ast_indicate(chan, AST_CONTROL_RINGING);
07921    return 0;
07922 }

static int pbx_builtin_setamaflags ( struct ast_channel ,
void *   
) [static]

Definition at line 8019 of file pbx.c.

References ast_cdr_setamaflags(), and chan.

08020 {
08021    /* Copy the AMA Flags as specified */
08022    ast_cdr_setamaflags(chan, data ? data : "");
08023    return 0;
08024 }

static int pbx_builtin_wait ( struct ast_channel ,
void *   
) [static]

Definition at line 8143 of file pbx.c.

References ast_safe_sleep(), chan, and s.

08144 {
08145    double s;
08146    int ms;
08147 
08148    /* Wait for "n" seconds */
08149    if (data && (s = atof(data)) > 0.0) {
08150       ms = s * 1000.0;
08151       return ast_safe_sleep(chan, ms);
08152    }
08153    return 0;
08154 }

static int pbx_builtin_waitexten ( struct ast_channel ,
void *   
) [static]

Definition at line 8159 of file pbx.c.

References AST_APP_ARG, ast_app_parse_options(), AST_CONTROL_HOLD, AST_CONTROL_UNHOLD, AST_DECLARE_APP_ARGS, ast_exists_extension(), ast_get_indication_tone(), ast_indicate(), ast_indicate_data(), ast_log(), ast_playtones_start(), ast_playtones_stop(), AST_SOFTHANGUP_TIMEOUT, AST_STANDARD_APP_ARGS, ast_strdupa, ast_strlen_zero(), ast_test_flag, ast_tonepair_start(), ast_verb, ast_waitfordigit(), chan, ast_channel::context, tone_zone_sound::data, ast_flags::flags, LOG_WARNING, parse(), ast_channel::pbx, ast_pbx::rtimeoutms, s, S_OR, set_ext_pri(), WAITEXTEN_DIALTONE, WAITEXTEN_MOH, waitexten_opts, and ast_channel::zone.

08160 {
08161    int ms, res;
08162    double s;
08163    struct ast_flags flags = {0};
08164    char *opts[1] = { NULL };
08165    char *parse;
08166    AST_DECLARE_APP_ARGS(args,
08167       AST_APP_ARG(timeout);
08168       AST_APP_ARG(options);
08169    );
08170 
08171    if (!ast_strlen_zero(data)) {
08172       parse = ast_strdupa(data);
08173       AST_STANDARD_APP_ARGS(args, parse);
08174    } else
08175       memset(&args, 0, sizeof(args));
08176 
08177    if (args.options)
08178       ast_app_parse_options(waitexten_opts, &flags, opts, args.options);
08179    
08180    if (ast_test_flag(&flags, WAITEXTEN_MOH) && !opts[0] ) {
08181       ast_log(LOG_WARNING, "The 'm' option has been specified for WaitExten without a class.\n"); 
08182    } else if (ast_test_flag(&flags, WAITEXTEN_MOH)) {
08183       ast_indicate_data(chan, AST_CONTROL_HOLD, S_OR(opts[0], NULL), strlen(opts[0]));
08184    } else if (ast_test_flag(&flags, WAITEXTEN_DIALTONE)) {
08185       const struct tone_zone_sound *ts = ast_get_indication_tone(chan->zone, "dial");
08186       if (ts)
08187          ast_playtones_start(chan, 0, ts->data, 0);
08188       else
08189          ast_tonepair_start(chan, 350, 440, 0, 0);
08190    }
08191    /* Wait for "n" seconds */
08192    if (args.timeout && (s = atof(args.timeout)) > 0)
08193        ms = s * 1000.0;
08194    else if (chan->pbx)
08195       ms = chan->pbx->rtimeoutms;
08196    else
08197       ms = 10000;
08198 
08199    res = ast_waitfordigit(chan, ms);
08200    if (!res) {
08201       if (ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 1, chan->cid.cid_num)) {
08202          ast_verb(3, "Timeout on %s, continuing...\n", chan->name);
08203       } else if (chan->_softhangup == AST_SOFTHANGUP_TIMEOUT) {
08204          ast_verb(3, "Call timeout on %s, checking for 'T'\n", chan->name);
08205          res = -1;
08206       } else if (ast_exists_extension(chan, chan->context, "t", 1, chan->cid.cid_num)) {
08207          ast_verb(3, "Timeout on %s, going to 't'\n", chan->name);
08208          set_ext_pri(chan, "t", 0); /* 0 will become 1, next time through the loop */
08209       } else {
08210          ast_log(LOG_WARNING, "Timeout but no rule 't' in context '%s'\n", chan->context);
08211          res = -1;
08212       }
08213    }
08214 
08215    if (ast_test_flag(&flags, WAITEXTEN_MOH))
08216       ast_indicate(chan, AST_CONTROL_UNHOLD);
08217    else if (ast_test_flag(&flags, WAITEXTEN_DIALTONE))
08218       ast_playtones_stop(chan);
08219 
08220    return res;
08221 }


Generated on Wed Aug 18 22:34:36 2010 for Asterisk - the Open Source PBX by  doxygen 1.4.7