2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2008, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
21 * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
23 * \author Mark Spencer <markster@digium.com>
25 * \ingroup applications
29 <depend>chan_local</depend>
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
38 #include <sys/signal.h>
40 #include <netinet/in.h>
42 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
43 #include "asterisk/lock.h"
44 #include "asterisk/file.h"
45 #include "asterisk/channel.h"
46 #include "asterisk/pbx.h"
47 #include "asterisk/module.h"
48 #include "asterisk/translate.h"
49 #include "asterisk/say.h"
50 #include "asterisk/config.h"
51 #include "asterisk/features.h"
52 #include "asterisk/musiconhold.h"
53 #include "asterisk/callerid.h"
54 #include "asterisk/utils.h"
55 #include "asterisk/app.h"
56 #include "asterisk/causes.h"
57 #include "asterisk/rtp_engine.h"
58 #include "asterisk/cdr.h"
59 #include "asterisk/manager.h"
60 #include "asterisk/privacy.h"
61 #include "asterisk/stringfields.h"
62 #include "asterisk/global_datastores.h"
63 #include "asterisk/dsp.h"
64 #include "asterisk/cel.h"
65 #include "asterisk/aoc.h"
66 #include "asterisk/ccss.h"
67 #include "asterisk/indications.h"
70 <application name="Dial" language="en_US">
72 Attempt to connect to another device or endpoint and bridge the call.
75 <parameter name="Technology/Resource" required="true" argsep="&">
76 <argument name="Technology/Resource" required="true">
77 <para>Specification of the device(s) to dial. These must be in the format of
78 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
79 represents a particular channel driver, and <replaceable>Resource</replaceable>
80 represents a resource available to that particular channel driver.</para>
82 <argument name="Technology2/Resource2" required="false" multiple="true">
83 <para>Optional extra devices to dial in parallel</para>
84 <para>If you need more then one enter them as
85 Technology2/Resource2&Technology3/Resourse3&.....</para>
88 <parameter name="timeout" required="false">
89 <para>Specifies the number of seconds we attempt to dial the specified devices</para>
90 <para>If not specified, this defaults to 136 years.</para>
92 <parameter name="options" required="false">
95 <argument name="x" required="true">
96 <para>The file to play to the called party</para>
98 <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
101 <para>Immediately answer the calling channel when the called channel answers in
102 all cases. Normally, the calling channel is answered when the called channel
103 answers, but when options such as A() and M() are used, the calling channel is
104 not answered until all actions on the called channel (such as playing an
105 announcement) are completed. This option can be used to answer the calling
106 channel before doing anything on the called channel. You will rarely need to use
107 this option, the default behavior is adequate in most cases.</para>
110 <para>Reset the call detail record (CDR) for this call.</para>
113 <para>If the Dial() application cancels this call, always set the flag to tell the channel
114 driver that the call is answered elsewhere.</para>
117 <para>Allow the calling user to dial a 1 digit extension while waiting for
118 a call to be answered. Exit to that extension if it exists in the
119 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
122 <option name="D" argsep=":">
123 <argument name="called" />
124 <argument name="calling" />
125 <argument name="progress" />
126 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
127 party has answered, but before the call gets bridged. The
128 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
129 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
130 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
131 immediately after receiving a PROGRESS message.</para>
134 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
137 <argument name="x" required="false" />
138 <para>If <replaceable>x</replaceable> is not provided, force the callerid of the <emphasis>calling</emphasis>
139 channel to be set as the extension associated with the channel using a dialplan <literal>hint</literal>.
140 For example, some PSTNs do not allow CallerID to be set to anything
141 other than the number assigned to the caller. If <replaceable>x</replaceable> is provided, though, then
142 this option behaves quite differently. Any outgoing channel created will have its connected party information
143 set to <replaceable>x</replaceable></para>
145 <option name="F" argsep="^">
146 <argument name="context" required="false" />
147 <argument name="exten" required="false" />
148 <argument name="priority" required="true" />
149 <para>When the caller hangs up, transfer the called party
150 to the specified destination and continue execution at that location.</para>
153 <para>Proceed with dialplan execution at the next priority in the current extension if the
154 source channel hangs up.</para>
157 <para>Proceed with dialplan execution at the next priority in the current extension if the
158 destination channel hangs up.</para>
160 <option name="G" argsep="^">
161 <argument name="context" required="false" />
162 <argument name="exten" required="false" />
163 <argument name="priority" required="true" />
164 <para>If the call is answered, transfer the calling party to
165 the specified <replaceable>priority</replaceable> and the called party to the specified
166 <replaceable>priority</replaceable> plus one.</para>
168 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
172 <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
175 <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
178 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
181 <para>Asterisk will ignore any connected line update requests or redirecting party update
182 requests it may receiveon this dial attempt.</para>
185 <para>Allow the called party to enable parking of the call by sending
186 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
189 <para>Allow the calling party to enable parking of the call by sending
190 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
192 <option name="L" argsep=":">
193 <argument name="x" required="true">
194 <para>Maximum call time, in milliseconds</para>
197 <para>Warning time, in milliseconds</para>
200 <para>Repeat time, in milliseconds</para>
202 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
203 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
204 <para>This option is affected by the following variables:</para>
206 <variable name="LIMIT_PLAYAUDIO_CALLER">
207 <value name="yes" default="true" />
209 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
211 <variable name="LIMIT_PLAYAUDIO_CALLEE">
213 <value name="no" default="true"/>
214 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
216 <variable name="LIMIT_TIMEOUT_FILE">
217 <value name="filename"/>
218 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
219 If not set, the time remaining will be announced.</para>
221 <variable name="LIMIT_CONNECT_FILE">
222 <value name="filename"/>
223 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
224 If not set, the time remaining will be announced.</para>
226 <variable name="LIMIT_WARNING_FILE">
227 <value name="filename"/>
228 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
229 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
234 <argument name="class" required="false"/>
235 <para>Provide hold music to the calling party until a requested
236 channel answers. A specific music on hold <replaceable>class</replaceable>
237 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
239 <option name="M" argsep="^">
240 <argument name="macro" required="true">
241 <para>Name of the macro that should be executed.</para>
243 <argument name="arg" multiple="true">
244 <para>Macro arguments</para>
246 <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
247 before connecting to the calling channel. Arguments can be specified to the Macro
248 using <literal>^</literal> as a delimiter. The macro can set the variable
249 <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
250 finished executing:</para>
252 <variable name="MACRO_RESULT">
253 <para>If set, this action will be taken after the macro finished executing.</para>
255 Hangup both legs of the call
257 <value name="CONGESTION">
258 Behave as if line congestion was encountered
261 Behave as if a busy signal was encountered
263 <value name="CONTINUE">
264 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
266 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
267 <value name="GOTO:<context>^<exten>^<priority>">
268 Transfer the call to the specified destination.
273 <para>You cannot use any additional action post answer options in conjunction
274 with this option. Also, pbx services are not run on the peer (called) channel,
275 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
277 <warning><para>Be aware of the limitations that macros have, specifically with regards to use of
278 the <literal>WaitExten</literal> application. For more information, see the documentation for
279 Macro()</para></warning>
282 <argument name="delete">
283 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
284 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
286 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
287 always be deleted.</para>
289 <para>This option is a modifier for the call screening/privacy mode. (See the
290 <literal>p</literal> and <literal>P</literal> options.) It specifies
291 that no introductions are to be saved in the <directory>priv-callerintros</directory>
295 <para>This option is a modifier for the call screening/privacy mode. It specifies
296 that if Caller*ID is present, do not screen the call.</para>
299 <para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
300 be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
301 behavior of Asterisk 1.0 and earlier.</para>
304 <argument name="mode">
305 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
306 the originator hanging up will cause the phone to ring back immediately.</para>
307 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
308 flashes the trunk, it will ring their phone back.</para>
310 <para>Enables <emphasis>operator services</emphasis> mode. This option only
311 works when bridging a DAHDI channel to another DAHDI channel
312 only. if specified on non-DAHDI interfaces, it will be ignored.
313 When the destination answers (presumably an operator services
314 station), the originator no longer has control of their line.
315 They may hang up, but the switch will not release their line
316 until the destination party (the operator) hangs up.</para>
319 <para>This option enables screening mode. This is basically Privacy mode
320 without memory.</para>
323 <argument name="x" />
324 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
325 it is provided. The current extension is used if a database family/key is not specified.</para>
328 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
329 party until the called channel has answered.</para>
330 <argument name="tone" required="false">
331 <para>Indicate progress to calling party. Send audio 'tone' from indications.conf</para>
335 <argument name="x" required="true" />
336 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
337 answered the call.</para>
340 <argument name="x" required="true" />
341 <para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable></para>
344 <para>Allow the called party to transfer the calling party by sending the
345 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
346 transfers initiated by other methods.</para>
349 <para>Allow the calling party to transfer the called party by sending the
350 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
351 transfers initiated by other methods.</para>
353 <option name="U" argsep="^">
354 <argument name="x" required="true">
355 <para>Name of the subroutine to execute via Gosub</para>
357 <argument name="arg" multiple="true" required="false">
358 <para>Arguments for the Gosub routine</para>
360 <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
361 to the calling channel. Arguments can be specified to the Gosub
362 using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
363 <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
365 <variable name="GOSUB_RESULT">
367 Hangup both legs of the call.
369 <value name="CONGESTION">
370 Behave as if line congestion was encountered.
373 Behave as if a busy signal was encountered.
375 <value name="CONTINUE">
376 Hangup the called party and allow the calling party
377 to continue dialplan execution at the next priority.
379 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
380 <value name="GOTO:<context>^<exten>^<priority>">
381 Transfer the call to the specified priority. Optionally, an extension, or
382 extension and priority can be specified.
387 <para>You cannot use any additional action post answer options in conjunction
388 with this option. Also, pbx services are not run on the peer (called) channel,
389 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
393 <argument name = "x" required="true">
394 <para>Force the outgoing callerid presentation indicator parameter to be set
395 to one of the values passed in <replaceable>x</replaceable>:
396 <literal>allowed_not_screened</literal>
397 <literal>allowed_passed_screen</literal>
398 <literal>allowed_failed_screen</literal>
399 <literal>allowed</literal>
400 <literal>prohib_not_screened</literal>
401 <literal>prohib_passed_screen</literal>
402 <literal>prohib_failed_screen</literal>
403 <literal>prohib</literal>
404 <literal>unavailable</literal></para>
408 <para>Allow the called party to enable recording of the call by sending
409 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
412 <para>Allow the calling party to enable recording of the call by sending
413 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
416 <para>Allow the called party to enable recording of the call by sending
417 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
420 <para>Allow the calling party to enable recording of the call by sending
421 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
424 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
428 <parameter name="URL">
429 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
433 <para>This application will place calls to one or more specified channels. As soon
434 as one of the requested channels answers, the originating channel will be
435 answered, if it has not already been answered. These two channels will then
436 be active in a bridged call. All other channels that were requested will then
439 <para>Unless there is a timeout specified, the Dial application will wait
440 indefinitely until one of the called channels answers, the user hangs up, or
441 if all of the called channels are busy or unavailable. Dialplan executing will
442 continue if no requested channels can be called, or if the timeout expires.
443 This application will report normal termination if the originating channel
444 hangs up, or if the call is bridged and either of the parties in the bridge
445 ends the call.</para>
446 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
447 application will be put into that group (as in Set(GROUP()=...).
448 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
449 application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
450 however, the variable will be unset after use.</para>
452 <para>This application sets the following channel variables:</para>
454 <variable name="DIALEDTIME">
455 <para>This is the time from dialing a channel until when it is disconnected.</para>
457 <variable name="ANSWEREDTIME">
458 <para>This is the amount of time for actual call.</para>
460 <variable name="DIALSTATUS">
461 <para>This is the status of the call</para>
462 <value name="CHANUNAVAIL" />
463 <value name="CONGESTION" />
464 <value name="NOANSWER" />
465 <value name="BUSY" />
466 <value name="ANSWER" />
467 <value name="CANCEL" />
468 <value name="DONTCALL">
469 For the Privacy and Screening Modes.
470 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
472 <value name="TORTURE">
473 For the Privacy and Screening Modes.
474 Will be set if the called party chooses to send the calling party to the 'torture' script.
476 <value name="INVALIDARGS" />
481 <application name="RetryDial" language="en_US">
483 Place a call, retrying on failure allowing an optional exit extension.
486 <parameter name="announce" required="true">
487 <para>Filename of sound that will be played when no channel can be reached</para>
489 <parameter name="sleep" required="true">
490 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
492 <parameter name="retries" required="true">
493 <para>Number of retries</para>
494 <para>When this is reached flow will continue at the next priority in the dialplan</para>
496 <parameter name="dialargs" required="true">
497 <para>Same format as arguments provided to the Dial application</para>
501 <para>This application will attempt to place a call using the normal Dial application.
502 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
503 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
504 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
505 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
506 While waiting to retry a call, a 1 digit extension may be dialed. If that
507 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
508 one, The call will jump to that extension immediately.
509 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
510 to the Dial application.</para>
515 static const char app[] = "Dial";
516 static const char rapp[] = "RetryDial";
519 OPT_ANNOUNCE = (1 << 0),
520 OPT_RESETCDR = (1 << 1),
521 OPT_DTMF_EXIT = (1 << 2),
522 OPT_SENDDTMF = (1 << 3),
523 OPT_FORCECLID = (1 << 4),
524 OPT_GO_ON = (1 << 5),
525 OPT_CALLEE_HANGUP = (1 << 6),
526 OPT_CALLER_HANGUP = (1 << 7),
527 OPT_ORIGINAL_CLID = (1 << 8),
528 OPT_DURATION_LIMIT = (1 << 9),
529 OPT_MUSICBACK = (1 << 10),
530 OPT_CALLEE_MACRO = (1 << 11),
531 OPT_SCREEN_NOINTRO = (1 << 12),
532 OPT_SCREEN_NOCALLERID = (1 << 13),
533 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
534 OPT_SCREENING = (1 << 15),
535 OPT_PRIVACY = (1 << 16),
536 OPT_RINGBACK = (1 << 17),
537 OPT_DURATION_STOP = (1 << 18),
538 OPT_CALLEE_TRANSFER = (1 << 19),
539 OPT_CALLER_TRANSFER = (1 << 20),
540 OPT_CALLEE_MONITOR = (1 << 21),
541 OPT_CALLER_MONITOR = (1 << 22),
542 OPT_GOTO = (1 << 23),
543 OPT_OPERMODE = (1 << 24),
544 OPT_CALLEE_PARK = (1 << 25),
545 OPT_CALLER_PARK = (1 << 26),
546 OPT_IGNORE_FORWARDING = (1 << 27),
547 OPT_CALLEE_GOSUB = (1 << 28),
548 OPT_CALLEE_MIXMONITOR = (1 << 29),
549 OPT_CALLER_MIXMONITOR = (1 << 30),
550 OPT_CALLER_ANSWER = (1 << 31),
553 #define DIAL_STILLGOING (1 << 31)
554 #define DIAL_NOFORWARDHTML ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
555 #define DIAL_CALLERID_ABSENT ((uint64_t)1 << 33) /* TRUE if caller id is not available for connected line. */
556 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 34)
557 #define OPT_PEER_H ((uint64_t)1 << 35)
558 #define OPT_CALLEE_GO_ON ((uint64_t)1 << 36)
559 #define OPT_CANCEL_TIMEOUT ((uint64_t)1 << 37)
560 #define OPT_FORCE_CID_TAG ((uint64_t)1 << 38)
561 #define OPT_FORCE_CID_PRES ((uint64_t)1 << 39)
564 OPT_ARG_ANNOUNCE = 0,
567 OPT_ARG_DURATION_LIMIT,
569 OPT_ARG_CALLEE_MACRO,
571 OPT_ARG_CALLEE_GOSUB,
572 OPT_ARG_CALLEE_GO_ON,
574 OPT_ARG_DURATION_STOP,
576 OPT_ARG_SCREEN_NOINTRO,
578 OPT_ARG_FORCE_CID_TAG,
579 OPT_ARG_FORCE_CID_PRES,
580 /* note: this entry _MUST_ be the last one in the enum */
584 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
585 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
586 AST_APP_OPTION('a', OPT_CALLER_ANSWER),
587 AST_APP_OPTION('C', OPT_RESETCDR),
588 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
589 AST_APP_OPTION('d', OPT_DTMF_EXIT),
590 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
591 AST_APP_OPTION('e', OPT_PEER_H),
592 AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
593 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
594 AST_APP_OPTION('g', OPT_GO_ON),
595 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
596 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
597 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
598 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
599 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
600 AST_APP_OPTION('k', OPT_CALLEE_PARK),
601 AST_APP_OPTION('K', OPT_CALLER_PARK),
602 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
603 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
604 AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
605 AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
606 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
607 AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
608 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
609 AST_APP_OPTION('p', OPT_SCREENING),
610 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
611 AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
612 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
613 AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
614 AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
615 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
616 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
617 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
618 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
619 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
620 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
621 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
622 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
625 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
626 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
627 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
628 OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
629 !chan->audiohooks && !peer->audiohooks)
632 * The list of active channels
635 struct chanlist *next;
636 struct ast_channel *chan;
638 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
639 struct ast_party_connected_line connected;
640 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
641 unsigned int pending_connected_update:1;
642 struct ast_aoc_decoded *aoc_s_rate_list;
645 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
647 static void chanlist_free(struct chanlist *outgoing)
649 ast_party_connected_line_free(&outgoing->connected);
650 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
654 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
656 /* Hang up a tree of stuff */
659 /* Hangup any existing lines we have open */
660 if (outgoing->chan && (outgoing->chan != exception)) {
661 if (answered_elsewhere) {
662 /* The flag is used for local channel inheritance and stuff */
663 ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
664 /* This is for the channel drivers */
665 outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
667 ast_hangup(outgoing->chan);
670 outgoing = outgoing->next;
675 #define AST_MAX_WATCHERS 256
678 * argument to handle_cause() and other functions.
681 struct ast_channel *chan;
687 static void handle_cause(int cause, struct cause_args *num)
689 struct ast_cdr *cdr = num->chan->cdr;
698 case AST_CAUSE_CONGESTION:
704 case AST_CAUSE_NO_ROUTE_DESTINATION:
705 case AST_CAUSE_UNREGISTERED:
711 case AST_CAUSE_NO_ANSWER:
713 ast_cdr_noanswer(cdr);
716 case AST_CAUSE_NORMAL_CLEARING:
725 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
727 char rexten[2] = { exten, '\0' };
730 if (!ast_goto_if_exists(chan, context, rexten, pri))
733 if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
735 else if (!ast_strlen_zero(chan->macrocontext)) {
736 if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
743 /* do not call with chan lock held */
744 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
749 ast_channel_lock(chan);
750 context = ast_strdupa(S_OR(chan->macrocontext, chan->context));
751 exten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
752 ast_channel_unlock(chan);
754 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
757 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
759 struct ast_channel *chans[] = { src, dst };
760 ast_manager_event_multichan(EVENT_FLAG_CALL, "Dial", 2, chans,
761 "SubEvent: Begin\r\n"
763 "Destination: %s\r\n"
764 "CallerIDNum: %s\r\n"
765 "CallerIDName: %s\r\n"
767 "DestUniqueID: %s\r\n"
768 "Dialstring: %s\r\n",
769 src->name, dst->name,
770 S_COR(src->caller.id.number.valid, src->caller.id.number.str, "<unknown>"),
771 S_COR(src->caller.id.name.valid, src->caller.id.name.str, "<unknown>"),
772 src->uniqueid, dst->uniqueid,
773 dialstring ? dialstring : "");
776 static void senddialendevent(struct ast_channel *src, const char *dialstatus)
778 ast_manager_event(src, EVENT_FLAG_CALL, "Dial",
782 "DialStatus: %s\r\n",
783 src->name, src->uniqueid, dialstatus);
787 * helper function for wait_for_answer()
789 * XXX this code is highly suspicious, as it essentially overwrites
790 * the outgoing channel without properly deleting it.
792 * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
794 static void do_forward(struct chanlist *o,
795 struct cause_args *num, struct ast_flags64 *peerflags, int single, int *to)
798 struct ast_channel *original = o->chan;
799 struct ast_channel *c = o->chan; /* the winner */
800 struct ast_channel *in = num->chan; /* the input channel */
805 ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
806 if ((stuff = strchr(tmpchan, '/'))) {
810 const char *forward_context;
812 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
813 if (ast_strlen_zero(forward_context)) {
814 forward_context = NULL;
816 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
817 ast_channel_unlock(c);
822 ast_cel_report_event(in, AST_CEL_FORWARD, NULL, c->call_forward, NULL);
824 /* Before processing channel, go ahead and check for forwarding */
825 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
826 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
827 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
828 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
830 cause = AST_CAUSE_BUSY;
832 /* Setup parameters */
833 c = o->chan = ast_request(tech, in->nativeformats, in, stuff, &cause);
836 ast_channel_make_compatible(o->chan, in);
837 ast_channel_inherit_variables(in, o->chan);
838 ast_channel_datastore_inherit(in, o->chan);
839 /* When a call is forwarded, we don't want to track new interfaces
840 * dialed for CC purposes. Setting the done flag will ensure that
841 * any Dial operations that happen later won't record CC interfaces.
843 ast_ignore_cc(o->chan);
844 ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", o->chan->name);
847 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
851 ast_clear_flag64(o, DIAL_STILLGOING);
852 handle_cause(cause, num);
853 ast_hangup(original);
855 struct ast_party_redirecting redirecting;
857 if (single && CAN_EARLY_BRIDGE(peerflags, c, in)) {
858 ast_rtp_instance_early_bridge_make_compatible(c, in);
861 ast_channel_set_redirecting(c, &original->redirecting, NULL);
863 while (ast_channel_trylock(in)) {
864 CHANNEL_DEADLOCK_AVOIDANCE(c);
866 if (!c->redirecting.from.number.valid
867 || ast_strlen_zero(c->redirecting.from.number.str)) {
869 * The call was not previously redirected so it is
870 * now redirected from this number.
872 ast_party_number_free(&c->redirecting.from.number);
873 ast_party_number_init(&c->redirecting.from.number);
874 c->redirecting.from.number.valid = 1;
875 c->redirecting.from.number.str =
876 ast_strdup(S_OR(in->macroexten, in->exten));
879 c->dialed.transit_network_select = in->dialed.transit_network_select;
881 if (ast_test_flag64(o, OPT_FORCECLID)) {
882 ast_party_id_free(&c->caller.id);
883 ast_party_id_init(&c->caller.id);
884 c->caller.id.number.valid = 1;
885 c->caller.id.number.str = ast_strdup(S_OR(in->macroexten, in->exten));
886 ast_string_field_set(c, accountcode, c->accountcode);
888 ast_party_caller_copy(&c->caller, &in->caller);
889 ast_string_field_set(c, accountcode, in->accountcode);
891 ast_party_connected_line_copy(&c->connected, &original->connected);
893 c->data = "(Outgoing Line)";
895 * We must unlock c before calling ast_channel_redirecting_macro, because
896 * we put c into autoservice there. That is pretty much a guaranteed
897 * deadlock. This is why the handling of c's lock may seem a bit unusual
900 ast_party_redirecting_init(&redirecting);
901 ast_party_redirecting_copy(&redirecting, &c->redirecting);
902 ast_channel_unlock(c);
903 if (ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
904 ast_channel_update_redirecting(in, &redirecting, NULL);
906 ast_party_redirecting_free(&redirecting);
907 ast_channel_unlock(in);
909 ast_clear_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE);
910 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
914 if (ast_call(c, stuff, 0)) {
915 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
917 ast_clear_flag64(o, DIAL_STILLGOING);
918 ast_hangup(original);
924 while (ast_channel_trylock(in)) {
925 CHANNEL_DEADLOCK_AVOIDANCE(c);
927 senddialevent(in, c, stuff);
928 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
929 char cidname[AST_MAX_EXTENSION] = "";
930 const char *tmpexten;
931 tmpexten = ast_strdupa(S_OR(in->macroexten, in->exten));
932 ast_channel_unlock(in);
933 ast_channel_unlock(c);
934 ast_set_callerid(c, tmpexten, get_cid_name(cidname, sizeof(cidname), in), NULL);
936 ast_channel_unlock(in);
937 ast_channel_unlock(c);
939 /* Hangup the original channel now, in case we needed it */
940 ast_hangup(original);
943 ast_indicate(in, -1);
948 /* argument used for some functions. */
949 struct privacy_args {
953 char privintro[1024];
957 static struct ast_channel *wait_for_answer(struct ast_channel *in,
958 struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
960 struct privacy_args *pa,
961 const struct cause_args *num_in, int *result, char *dtmf_progress,
964 struct cause_args num = *num_in;
965 int prestart = num.busy + num.congestion + num.nochan;
967 struct ast_channel *peer = NULL;
968 /* single is set if only one destination is enabled */
969 int single = outgoing && !outgoing->next;
971 struct chanlist *epollo;
973 struct ast_party_connected_line connected_caller;
974 struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
975 int cc_recall_core_id;
977 int cc_frame_received = 0;
980 ast_party_connected_line_init(&connected_caller);
982 /* Turn off hold music, etc */
983 if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK)) {
984 ast_deactivate_generator(in);
985 /* If we are calling a single channel, and not providing ringback or music, */
986 /* then, make them compatible for in-band tone purpose */
987 if (ast_channel_make_compatible(outgoing->chan, in) < 0) {
988 /* If these channels can not be made compatible,
989 * there is no point in continuing. The bridge
990 * will just fail if it gets that far.
993 strcpy(pa->status, "CONGESTION");
994 ast_cdr_failed(in->cdr);
999 if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1000 ast_channel_lock(outgoing->chan);
1001 ast_connected_line_copy_from_caller(&connected_caller, &outgoing->chan->caller);
1002 ast_channel_unlock(outgoing->chan);
1003 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1004 ast_channel_update_connected_line(in, &connected_caller, NULL);
1005 ast_party_connected_line_free(&connected_caller);
1009 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1012 for (epollo = outgoing; epollo; epollo = epollo->next)
1013 ast_poll_channel_add(in, epollo->chan);
1016 while (*to && !peer) {
1018 int pos = 0; /* how many channels do we handle */
1019 int numlines = prestart;
1020 struct ast_channel *winner;
1021 struct ast_channel *watchers[AST_MAX_WATCHERS];
1023 watchers[pos++] = in;
1024 for (o = outgoing; o; o = o->next) {
1025 /* Keep track of important channels */
1026 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1027 watchers[pos++] = o->chan;
1030 if (pos == 1) { /* only the input channel is available */
1031 if (numlines == (num.busy + num.congestion + num.nochan)) {
1032 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1034 strcpy(pa->status, "BUSY");
1035 else if (num.congestion)
1036 strcpy(pa->status, "CONGESTION");
1037 else if (num.nochan)
1038 strcpy(pa->status, "CHANUNAVAIL");
1040 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1044 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1048 winner = ast_waitfor_n(watchers, pos, to);
1049 for (o = outgoing; o; o = o->next) {
1050 struct ast_frame *f;
1051 struct ast_channel *c = o->chan;
1055 if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
1057 ast_verb(3, "%s answered %s\n", c->name, in->name);
1058 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1059 if (o->pending_connected_update) {
1060 if (ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1061 ast_channel_update_connected_line(in, &o->connected, NULL);
1063 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1064 ast_channel_lock(c);
1065 ast_connected_line_copy_from_caller(&connected_caller, &c->caller);
1066 ast_channel_unlock(c);
1067 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1068 ast_channel_update_connected_line(in, &connected_caller, NULL);
1069 ast_party_connected_line_free(&connected_caller);
1072 if (o->aoc_s_rate_list) {
1073 size_t encoded_size;
1074 struct ast_aoc_encoded *encoded;
1075 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1076 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1077 ast_aoc_destroy_encoded(encoded);
1081 ast_copy_flags64(peerflags, o,
1082 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1083 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1084 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1085 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1086 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1087 DIAL_NOFORWARDHTML);
1088 ast_string_field_set(c, dialcontext, "");
1089 ast_copy_string(c->exten, "", sizeof(c->exten));
1095 /* here, o->chan == c == winner */
1096 if (!ast_strlen_zero(c->call_forward)) {
1097 pa->sentringing = 0;
1098 if (!ignore_cc && (f = ast_read(c))) {
1099 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1100 /* This channel is forwarding the call, and is capable of CC, so
1101 * be sure to add the new device interface to the list
1103 ast_handle_cc_control_frame(in, c, f->data.ptr);
1107 do_forward(o, &num, peerflags, single, to);
1110 f = ast_read(winner);
1112 in->hangupcause = c->hangupcause;
1114 ast_poll_channel_del(in, c);
1118 ast_clear_flag64(o, DIAL_STILLGOING);
1119 handle_cause(in->hangupcause, &num);
1122 if (f->frametype == AST_FRAME_CONTROL) {
1123 switch (f->subclass.integer) {
1124 case AST_CONTROL_ANSWER:
1125 /* This is our guy if someone answered. */
1127 ast_verb(3, "%s answered %s\n", c->name, in->name);
1128 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1129 if (o->pending_connected_update) {
1130 if (ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1131 ast_channel_update_connected_line(in, &o->connected, NULL);
1133 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1134 ast_channel_lock(c);
1135 ast_connected_line_copy_from_caller(&connected_caller, &c->caller);
1136 ast_channel_unlock(c);
1137 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1138 ast_channel_update_connected_line(in, &connected_caller, NULL);
1139 ast_party_connected_line_free(&connected_caller);
1142 if (o->aoc_s_rate_list) {
1143 size_t encoded_size;
1144 struct ast_aoc_encoded *encoded;
1145 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1146 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1147 ast_aoc_destroy_encoded(encoded);
1152 peer->cdr->answer = ast_tvnow();
1153 peer->cdr->disposition = AST_CDR_ANSWERED;
1155 ast_copy_flags64(peerflags, o,
1156 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1157 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1158 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1159 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1160 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1161 DIAL_NOFORWARDHTML);
1162 ast_string_field_set(c, dialcontext, "");
1163 ast_copy_string(c->exten, "", sizeof(c->exten));
1164 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
1165 /* Setup early bridge if appropriate */
1166 ast_channel_early_bridge(in, peer);
1168 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1169 in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1170 c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1172 case AST_CONTROL_BUSY:
1173 ast_verb(3, "%s is busy\n", c->name);
1174 in->hangupcause = c->hangupcause;
1177 ast_clear_flag64(o, DIAL_STILLGOING);
1178 handle_cause(AST_CAUSE_BUSY, &num);
1180 case AST_CONTROL_CONGESTION:
1181 ast_verb(3, "%s is circuit-busy\n", c->name);
1182 in->hangupcause = c->hangupcause;
1185 ast_clear_flag64(o, DIAL_STILLGOING);
1186 handle_cause(AST_CAUSE_CONGESTION, &num);
1188 case AST_CONTROL_RINGING:
1189 /* This is a tricky area to get right when using a native
1190 * CC agent. The reason is that we do the best we can to send only a
1191 * single ringing notification to the caller.
1193 * Call completion complicates the logic used here. CCNR is typically
1194 * offered during a ringing message. Let's say that party A calls
1195 * parties B, C, and D. B and C do not support CC requests, but D
1196 * does. If we were to receive a ringing notification from B before
1197 * the others, then we would end up sending a ringing message to
1198 * A with no CCNR offer present.
1200 * The approach that we have taken is that if we receive a ringing
1201 * response from a party and no CCNR offer is present, we need to
1202 * wait. Specifically, we need to wait until either a) a called party
1203 * offers CCNR in its ringing response or b) all called parties have
1204 * responded in some way to our call and none offers CCNR.
1206 * The drawback to this is that if one of the parties has a delayed
1207 * response or, god forbid, one just plain doesn't respond to our
1208 * outgoing call, then this will result in a significant delay between
1209 * when the caller places the call and hears ringback.
1211 * Note also that if CC is disabled for this call, then it is perfectly
1212 * fine for ringing frames to get sent through.
1215 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1216 ast_verb(3, "%s is ringing\n", c->name);
1217 /* Setup early media if appropriate */
1218 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1219 ast_channel_early_bridge(in, c);
1220 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1221 ast_indicate(in, AST_CONTROL_RINGING);
1226 case AST_CONTROL_PROGRESS:
1227 ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
1228 /* Setup early media if appropriate */
1229 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1230 ast_channel_early_bridge(in, c);
1231 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1232 if (single || (!single && !pa->sentringing)) {
1233 ast_indicate(in, AST_CONTROL_PROGRESS);
1235 if(!ast_strlen_zero(dtmf_progress)) {
1236 ast_verb(3, "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n", dtmf_progress);
1237 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1240 case AST_CONTROL_VIDUPDATE:
1241 ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
1242 ast_indicate(in, AST_CONTROL_VIDUPDATE);
1244 case AST_CONTROL_SRCUPDATE:
1245 ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
1246 ast_indicate(in, AST_CONTROL_SRCUPDATE);
1248 case AST_CONTROL_CONNECTED_LINE:
1249 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1250 ast_verb(3, "Connected line update to %s prevented.\n", in->name);
1251 } else if (!single) {
1252 struct ast_party_connected_line connected;
1253 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n", c->name, in->name);
1254 ast_party_connected_line_set_init(&connected, &o->connected);
1255 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1256 ast_party_connected_line_set(&o->connected, &connected, NULL);
1257 ast_party_connected_line_free(&connected);
1258 o->pending_connected_update = 1;
1260 if (ast_channel_connected_line_macro(c, in, f, 1, 1)) {
1261 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1265 case AST_CONTROL_AOC:
1267 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1268 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1269 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1270 o->aoc_s_rate_list = decoded;
1272 ast_aoc_destroy_decoded(decoded);
1276 case AST_CONTROL_REDIRECTING:
1277 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1278 ast_verb(3, "Redirecting update to %s prevented.\n", in->name);
1280 ast_verb(3, "%s redirecting info has changed, passing it to %s\n", c->name, in->name);
1281 if (ast_channel_redirecting_macro(c, in, f, 1, 1)) {
1282 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1284 pa->sentringing = 0;
1287 case AST_CONTROL_PROCEEDING:
1288 ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
1289 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1290 ast_channel_early_bridge(in, c);
1291 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1292 ast_indicate(in, AST_CONTROL_PROCEEDING);
1294 case AST_CONTROL_HOLD:
1295 ast_verb(3, "Call on %s placed on hold\n", c->name);
1296 ast_indicate(in, AST_CONTROL_HOLD);
1298 case AST_CONTROL_UNHOLD:
1299 ast_verb(3, "Call on %s left from hold\n", c->name);
1300 ast_indicate(in, AST_CONTROL_UNHOLD);
1302 case AST_CONTROL_OFFHOOK:
1303 case AST_CONTROL_FLASH:
1304 /* Ignore going off hook and flash */
1306 case AST_CONTROL_CC:
1308 ast_handle_cc_control_frame(in, c, f->data.ptr);
1309 cc_frame_received = 1;
1313 if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
1314 ast_verb(3, "%s stopped sounds\n", c->name);
1315 ast_indicate(in, -1);
1316 pa->sentringing = 0;
1320 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1322 } else if (single) {
1323 switch (f->frametype) {
1324 case AST_FRAME_VOICE:
1325 case AST_FRAME_IMAGE:
1326 case AST_FRAME_TEXT:
1327 if (ast_write(in, f)) {
1328 ast_log(LOG_WARNING, "Unable to write frame\n");
1331 case AST_FRAME_HTML:
1332 if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1333 ast_log(LOG_WARNING, "Unable to send URL\n");
1343 struct ast_frame *f = ast_read(in);
1345 if (f && (f->frametype != AST_FRAME_VOICE))
1346 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1347 else if (!f || (f->frametype != AST_FRAME_VOICE))
1348 printf("Hangup received on %s\n", in->name);
1350 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1353 strcpy(pa->status, "CANCEL");
1354 ast_cdr_noanswer(in->cdr);
1356 if (f->data.uint32) {
1357 in->hangupcause = f->data.uint32;
1362 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1367 /* now f is guaranteed non-NULL */
1368 if (f->frametype == AST_FRAME_DTMF) {
1369 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1370 const char *context;
1371 ast_channel_lock(in);
1372 context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1373 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1374 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1376 ast_cdr_noanswer(in->cdr);
1377 *result = f->subclass.integer;
1378 strcpy(pa->status, "CANCEL");
1380 ast_channel_unlock(in);
1382 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1386 ast_channel_unlock(in);
1389 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1390 detect_disconnect(in, f->subclass.integer, featurecode)) {
1391 ast_verb(3, "User requested call disconnect.\n");
1393 strcpy(pa->status, "CANCEL");
1394 ast_cdr_noanswer(in->cdr);
1397 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1403 /* Forward HTML stuff */
1404 if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
1405 if (ast_channel_sendhtml(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1)
1406 ast_log(LOG_WARNING, "Unable to send URL\n");
1408 if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
1409 if (ast_write(outgoing->chan, f))
1410 ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
1412 if (single && (f->frametype == AST_FRAME_CONTROL)) {
1413 if ((f->subclass.integer == AST_CONTROL_HOLD) ||
1414 (f->subclass.integer == AST_CONTROL_UNHOLD) ||
1415 (f->subclass.integer == AST_CONTROL_VIDUPDATE) ||
1416 (f->subclass.integer == AST_CONTROL_SRCUPDATE)) {
1417 ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass.integer, outgoing->chan->name);
1418 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1419 } else if (f->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
1420 if (ast_channel_connected_line_macro(in, outgoing->chan, f, 0, 1)) {
1421 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1423 } else if (f->subclass.integer == AST_CONTROL_REDIRECTING) {
1424 if (ast_channel_redirecting_macro(in, outgoing->chan, f, 0, 1)) {
1425 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1432 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1433 if (!*to || ast_check_hangup(in))
1434 ast_cdr_noanswer(in->cdr);
1438 for (epollo = outgoing; epollo; epollo = epollo->next) {
1440 ast_poll_channel_del(in, epollo->chan);
1445 ast_cc_completed(in, "Recall completed!");
1450 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
1452 struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
1453 struct ast_call_feature feature = { 0, };
1456 ast_str_append(&featurecode, 1, "%c", code);
1458 res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
1460 if (res != AST_FEATURE_RETURN_STOREDIGITS) {
1461 ast_str_reset(featurecode);
1463 if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
1470 static void replace_macro_delimiter(char *s)
1477 /* returns true if there is a valid privacy reply */
1478 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1482 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1484 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1489 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1490 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1496 /* Get the user's intro, store it in priv-callerintros/$CID,
1497 unless it is already there-- this should be done before the
1498 call is actually dialed */
1500 /* all ring indications and moh for the caller has been halted as soon as the
1501 target extension was picked up. We are going to have to kill some
1502 time and make the caller believe the peer hasn't picked up yet */
1504 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1505 char *original_moh = ast_strdupa(chan->musicclass);
1506 ast_indicate(chan, -1);
1507 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1508 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1509 ast_string_field_set(chan, musicclass, original_moh);
1510 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1511 ast_indicate(chan, AST_CONTROL_RINGING);
1515 /* Start autoservice on the other chan ?? */
1516 res2 = ast_autoservice_start(chan);
1517 /* Now Stream the File */
1518 for (loopcount = 0; loopcount < 3; loopcount++) {
1519 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1521 if (!res2) /* on timeout, play the message again */
1522 res2 = ast_play_and_wait(peer, "priv-callpending");
1523 if (!valid_priv_reply(opts, res2))
1525 /* priv-callpending script:
1526 "I have a caller waiting, who introduces themselves as:"
1529 res2 = ast_play_and_wait(peer, pa->privintro);
1530 if (!valid_priv_reply(opts, res2))
1532 /* now get input from the called party, as to their choice */
1534 /* XXX can we have both, or they are mutually exclusive ? */
1535 if (ast_test_flag64(opts, OPT_PRIVACY))
1536 res2 = ast_play_and_wait(peer, "priv-callee-options");
1537 if (ast_test_flag64(opts, OPT_SCREENING))
1538 res2 = ast_play_and_wait(peer, "screen-callee-options");
1540 /*! \page DialPrivacy Dial Privacy scripts
1541 \par priv-callee-options script:
1542 "Dial 1 if you wish this caller to reach you directly in the future,
1543 and immediately connect to their incoming call
1544 Dial 2 if you wish to send this caller to voicemail now and
1546 Dial 3 to send this caller to the torture menus, now and forevermore.
1547 Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1548 Dial 5 to allow this caller to come straight thru to you in the future,
1549 but right now, just this once, send them to voicemail."
1550 \par screen-callee-options script:
1551 "Dial 1 if you wish to immediately connect to the incoming call
1552 Dial 2 if you wish to send this caller to voicemail.
1553 Dial 3 to send this caller to the torture menus.
1554 Dial 4 to send this caller to a simple "go away" menu.
1556 if (valid_priv_reply(opts, res2))
1558 /* invalid option */
1559 res2 = ast_play_and_wait(peer, "vm-sorry");
1562 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1564 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1565 ast_indicate(chan, -1);
1566 pa->sentringing = 0;
1568 ast_autoservice_stop(chan);
1569 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1570 /* map keypresses to various things, the index is res2 - '1' */
1571 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1572 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1574 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1575 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1576 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1582 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1585 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1588 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1591 /* XXX should we set status to DENY ? */
1592 if (ast_test_flag64(opts, OPT_PRIVACY))
1594 /* if not privacy, then 5 is the same as "default" case */
1595 default: /* bad input or -1 if failure to start autoservice */
1596 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
1597 /* well, there seems basically two choices. Just patch the caller thru immediately,
1598 or,... put 'em thru to voicemail. */
1599 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1600 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1601 /* XXX should we set status to DENY ? */
1602 /* XXX what about the privacy flags ? */
1606 if (res2 == '1') { /* the only case where we actually connect */
1607 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1608 just clog things up, and it's not useful information, not being tied to a CID */
1609 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1610 ast_filedelete(pa->privintro, NULL);
1611 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1612 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1614 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1616 return 0; /* the good exit path */
1618 ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
1623 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1624 static int setup_privacy_args(struct privacy_args *pa,
1625 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1630 int silencethreshold;
1632 if (chan->caller.id.number.valid
1633 && !ast_strlen_zero(chan->caller.id.number.str)) {
1634 l = ast_strdupa(chan->caller.id.number.str);
1635 ast_shrink_phone_number(l);
1636 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1637 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1638 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1640 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1641 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1646 tnam = ast_strdupa(chan->name);
1647 /* clean the channel name so slashes don't try to end up in disk file name */
1648 for (tn2 = tnam; *tn2; tn2++) {
1649 if (*tn2 == '/') /* any other chars to be afraid of? */
1652 ast_verb(3, "Privacy-- callerid is empty\n");
1654 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
1656 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1659 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1661 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1662 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1663 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1664 pa->privdb_val = AST_PRIVACY_ALLOW;
1665 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1666 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1669 if (pa->privdb_val == AST_PRIVACY_DENY) {
1670 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1671 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1673 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1674 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1675 return 0; /* Is this right? */
1676 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1677 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1678 return 0; /* is this right??? */
1679 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1680 /* Get the user's intro, store it in priv-callerintros/$CID,
1681 unless it is already there-- this should be done before the
1682 call is actually dialed */
1684 /* make sure the priv-callerintros dir actually exists */
1685 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1686 if ((res = ast_mkdir(pa->privintro, 0755))) {
1687 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1691 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1692 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1693 /* the DELUX version of this code would allow this caller the
1694 option to hear and retape their previously recorded intro.
1697 int duration; /* for feedback from play_and_wait */
1698 /* the file doesn't exist yet. Let the caller submit his
1699 vocal intro for posterity */
1700 /* priv-recordintro script:
1702 "At the tone, please say your name:"
1705 silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1707 res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
1708 /* don't think we'll need a lock removed, we took care of
1709 conflicts by naming the pa.privintro file */
1711 /* Delete the file regardless since they hung up during recording */
1712 ast_filedelete(pa->privintro, NULL);
1713 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1714 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1716 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1719 if (!ast_streamfile(chan, "vm-dialout", chan->language) )
1720 ast_waitstream(chan, "");
1723 return 1; /* success */
1726 static void end_bridge_callback(void *data)
1730 struct ast_channel *chan = data;
1738 ast_channel_lock(chan);
1739 if (chan->cdr->answer.tv_sec) {
1740 snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->answer.tv_sec);
1741 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1744 if (chan->cdr->start.tv_sec) {
1745 snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->start.tv_sec);
1746 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1748 ast_channel_unlock(chan);
1751 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
1752 bconfig->end_bridge_callback_data = originator;
1755 static int dial_handle_playtones(struct ast_channel *chan, const char *data)
1757 struct ast_tone_zone_sound *ts = NULL;
1759 const char *str = data;
1761 if (ast_strlen_zero(str)) {
1762 ast_debug(1,"Nothing to play\n");
1766 ts = ast_get_indication_tone(chan->zone, str);
1768 if (ts && ts->data[0]) {
1769 res = ast_playtones_start(chan, 0, ts->data, 0);
1775 ts = ast_tone_zone_sound_unref(ts);
1779 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
1785 static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
1787 int res = -1; /* default: error */
1788 char *rest, *cur; /* scan the list of destinations */
1789 struct chanlist *outgoing = NULL; /* list of destinations */
1790 struct ast_channel *peer;
1791 int to; /* timeout */
1792 struct cause_args num = { chan, 0, 0, 0 };
1795 char *cid_num = NULL, *cid_name = NULL, *cid_tag = NULL, *cid_pres = NULL;
1797 struct ast_bridge_config config = { { 0, } };
1798 struct timeval calldurationlimit = { 0, };
1799 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
1800 struct privacy_args pa = {
1803 .status = "INVALIDARGS",
1805 int sentringing = 0, moh = 0;
1806 const char *outbound_group = NULL;
1810 int delprivintro = 0;
1811 AST_DECLARE_APP_ARGS(args,
1813 AST_APP_ARG(timeout);
1814 AST_APP_ARG(options);
1817 struct ast_flags64 opts = { 0, };
1818 char *opt_args[OPT_ARG_ARRAY_SIZE];
1819 struct ast_datastore *datastore = NULL;
1820 int fulldial = 0, num_dialed = 0;
1822 char device_name[AST_CHANNEL_NAME];
1824 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
1825 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
1826 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
1827 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
1828 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
1829 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
1831 if (ast_strlen_zero(data)) {
1832 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1833 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1837 parse = ast_strdupa(data);
1839 AST_STANDARD_APP_ARGS(args, parse);
1841 if (!ast_strlen_zero(args.options) &&
1842 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
1843 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1847 if (ast_strlen_zero(args.peers)) {
1848 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1849 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1853 if (ast_cc_call_init(chan, &ignore_cc)) {
1857 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
1858 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
1860 if (delprivintro < 0 || delprivintro > 1) {
1861 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
1866 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
1867 opt_args[OPT_ARG_RINGBACK] = NULL;
1870 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
1871 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
1872 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
1875 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
1876 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
1877 if (!calldurationlimit.tv_sec) {
1878 ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
1879 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1882 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
1885 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
1886 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
1887 dtmfcalled = strsep(&dtmf_progress, ":");
1888 dtmfcalling = strsep(&dtmf_progress, ":");
1891 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
1892 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
1896 if (ast_test_flag64(&opts, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID]))
1897 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &cid_name, &cid_num);
1898 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG) && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG]))
1899 cid_tag = ast_strdupa(opt_args[OPT_ARG_FORCE_CID_TAG]);
1900 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES) && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES]))
1901 cid_pres = ast_strdupa(opt_args[OPT_ARG_FORCE_CID_PRES]);
1902 if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
1903 ast_cdr_reset(chan->cdr, NULL);
1904 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
1905 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
1907 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
1908 res = setup_privacy_args(&pa, &opts, opt_args, chan);
1911 res = -1; /* reset default */
1914 if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
1915 __ast_answer(chan, 0, 0);
1921 /* If a channel group has been specified, get it for use when we create peer channels */
1923 ast_channel_lock(chan);
1924 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
1925 outbound_group = ast_strdupa(outbound_group);
1926 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
1927 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
1928 outbound_group = ast_strdupa(outbound_group);
1930 ast_channel_unlock(chan);
1931 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE |
1932 OPT_CANCEL_TIMEOUT | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
1934 /* loop through the list of dial destinations */
1936 while ((cur = strsep(&rest, "&")) ) {
1937 struct chanlist *tmp;
1938 struct ast_channel *tc; /* channel for this destination */
1939 /* Get a technology/[device:]number pair */
1941 char *interface = ast_strdupa(number);
1942 char *tech = strsep(&number, "/");
1943 /* find if we already dialed this interface */
1944 struct ast_dialed_interface *di;
1945 AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
1948 ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
1951 if (!(tmp = ast_calloc(1, sizeof(*tmp))))
1954 ast_copy_flags64(tmp, &opts,
1955 OPT_CANCEL_ELSEWHERE |
1956 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1957 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1958 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1959 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1960 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1961 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
1962 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
1964 ast_copy_string(numsubst, number, sizeof(numsubst));
1965 /* Request the peer */
1967 ast_channel_lock(chan);
1968 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
1970 * Seed the chanlist's connected line information with previously
1971 * acquired connected line info from the incoming channel. The
1972 * previously acquired connected line info could have been set
1973 * through the CONNECTED_LINE dialplan function.
1975 ast_party_connected_line_copy(&tmp->connected, &chan->connected);
1976 ast_channel_unlock(chan);
1979 dialed_interfaces = datastore->data;
1981 if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
1982 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
1987 datastore->inheritance = DATASTORE_INHERIT_FOREVER;
1989 if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
1990 ast_datastore_free(datastore);
1995 datastore->data = dialed_interfaces;
1996 AST_LIST_HEAD_INIT(dialed_interfaces);
1998 ast_channel_lock(chan);
1999 ast_channel_datastore_add(chan, datastore);
2000 ast_channel_unlock(chan);
2003 AST_LIST_LOCK(dialed_interfaces);
2004 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
2005 if (!strcasecmp(di->interface, interface)) {
2006 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
2011 AST_LIST_UNLOCK(dialed_interfaces);
2019 /* It is always ok to dial a Local interface. We only keep track of
2020 * which "real" interfaces have been dialed. The Local channel will
2021 * inherit this list so that if it ends up dialing a real interface,
2022 * it won't call one that has already been called. */
2023 if (strcasecmp(tech, "Local")) {
2024 if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
2025 AST_LIST_UNLOCK(dialed_interfaces);
2029 strcpy(di->interface, interface);
2031 AST_LIST_LOCK(dialed_interfaces);
2032 AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
2033 AST_LIST_UNLOCK(dialed_interfaces);
2036 tc = ast_request(tech, chan->nativeformats, chan, numsubst, &cause);
2038 /* If we can't, just go on to the next call */
2039 ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
2040 tech, cause, ast_cause2str(cause));
2041 handle_cause(cause, &num);
2042 if (!rest) /* we are on the last destination */
2043 chan->hangupcause = cause;
2045 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2046 if (!ast_cc_callback(chan, tech, numsubst, ast_cc_busy_interface)) {
2047 ast_cc_extension_monitor_add_dialstring(chan, interface, "");
2052 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2054 ast_cc_extension_monitor_add_dialstring(chan, interface, device_name);
2056 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
2058 ast_channel_lock(tc);
2059 while (ast_channel_trylock(chan)) {
2060 CHANNEL_DEADLOCK_AVOIDANCE(tc);
2062 /* Setup outgoing SDP to match incoming one */
2063 if (!outgoing && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2064 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2067 /* Inherit specially named variables from parent channel */
2068 ast_channel_inherit_variables(chan, tc);
2069 ast_channel_datastore_inherit(chan, tc);
2071 tc->appl = "AppDial";
2072 tc->data = "(Outgoing Line)";
2073 memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
2075 /* If the new channel has no callerid, try to guess what it should be */
2076 if (!tc->caller.id.number.valid) {
2077 if (chan->connected.id.number.valid) {
2078 struct ast_party_caller caller;
2080 ast_party_caller_set_init(&caller, &tc->caller);
2081 caller.id = chan->connected.id;
2082 caller.ani = chan->connected.ani;
2083 ast_channel_set_caller_event(tc, &caller, NULL);
2084 } else if (!ast_strlen_zero(chan->dialed.number.str)) {
2085 ast_set_callerid(tc, chan->dialed.number.str, NULL, NULL);
2086 } else if (!ast_strlen_zero(S_OR(chan->macroexten, chan->exten))) {
2087 ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), NULL, NULL);
2089 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2092 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2093 struct ast_party_connected_line connected;
2096 ast_party_connected_line_set_init(&connected, &tmp->chan->connected);
2098 pres = ast_parse_caller_presentation(cid_pres);
2100 pres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2103 pres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2106 connected.id.number.valid = 1;
2107 connected.id.number.str = cid_num;
2108 connected.id.number.presentation = pres;
2111 connected.id.name.valid = 1;
2112 connected.id.name.str = cid_name;
2113 connected.id.name.presentation = pres;
2115 connected.id.tag = cid_tag;
2116 ast_channel_set_connected_line(tmp->chan, &connected, NULL);
2118 ast_connected_line_copy_from_caller(&tc->connected, &chan->caller);
2121 ast_party_redirecting_copy(&tc->redirecting, &chan->redirecting);
2123 tc->dialed.transit_network_select = chan->dialed.transit_network_select;
2125 if (!ast_strlen_zero(chan->accountcode)) {
2126 ast_string_field_set(tc, peeraccount, chan->accountcode);
2128 if (ast_strlen_zero(tc->musicclass))
2129 ast_string_field_set(tc, musicclass, chan->musicclass);
2131 /* Pass ADSI CPE and transfer capability */
2132 tc->adsicpe = chan->adsicpe;
2133 tc->transfercapability = chan->transfercapability;
2135 /* If we have an outbound group, set this peer channel to it */
2137 ast_app_group_set_channel(tc, outbound_group);
2138 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2139 if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
2140 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
2142 /* Check if we're forced by configuration */
2143 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2144 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
2147 /* Inherit context and extension */
2148 ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
2149 if (!ast_strlen_zero(chan->macroexten))
2150 ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
2152 ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
2154 ast_channel_unlock(tc);
2155 res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
2157 /* Save the info in cdr's that we called them */
2159 ast_cdr_setdestchan(chan->cdr, tc->name);
2161 /* check the results of ast_call */
2163 /* Again, keep going even if there's an error */
2164 ast_debug(1, "ast call on peer returned %d\n", res);
2165 ast_verb(3, "Couldn't call %s\n", numsubst);
2166 if (tc->hangupcause) {
2167 chan->hangupcause = tc->hangupcause;
2169 ast_channel_unlock(chan);
2170 ast_cc_call_failed(chan, tc, interface);
2176 const char *tmpexten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
2177 senddialevent(chan, tc, numsubst);
2178 ast_verb(3, "Called %s\n", numsubst);
2179 ast_channel_unlock(chan);
2180 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2181 char cidname[AST_MAX_EXTENSION];
2182 ast_set_callerid(tc, tmpexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
2185 /* Put them in the list of outgoing thingies... We're ready now.
2186 XXX If we're forcibly removed, these outgoing calls won't get
2188 ast_set_flag64(tmp, DIAL_STILLGOING);
2190 tmp->next = outgoing;
2192 /* If this line is up, don't try anybody else */
2193 if (outgoing->chan->_state == AST_STATE_UP)
2197 if (ast_strlen_zero(args.timeout)) {
2200 to = atoi(args.timeout);
2204 ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2210 strcpy(pa.status, "CHANUNAVAIL");
2211 if (fulldial == num_dialed) {
2216 /* Our status will at least be NOANSWER */
2217 strcpy(pa.status, "NOANSWER");
2218 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2220 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2221 char *original_moh = ast_strdupa(chan->musicclass);
2222 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
2223 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2224 ast_string_field_set(chan, musicclass, original_moh);
2226 ast_moh_start(chan, NULL, NULL);
2228 ast_indicate(chan, AST_CONTROL_PROGRESS);
2229 } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
2230 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
2231 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
2232 ast_indicate(chan, AST_CONTROL_RINGING);
2235 ast_indicate(chan, AST_CONTROL_PROGRESS);
2238 ast_indicate(chan, AST_CONTROL_RINGING);
2244 peer = wait_for_answer(chan, outgoing, &to, peerflags, opt_args, &pa, &num, &result, dtmf_progress, ignore_cc);
2246 /* The ast_channel_datastore_remove() function could fail here if the
2247 * datastore was moved to another channel during a masquerade. If this is
2248 * the case, don't free the datastore here because later, when the channel
2249 * to which the datastore was moved hangs up, it will attempt to free this
2250 * datastore again, causing a crash
2252 if (!ast_channel_datastore_remove(chan, datastore))
2253 ast_datastore_free(datastore);
2257 } else if (to) { /* Musta gotten hung up */
2259 } else { /* Nobody answered, next please? */
2263 /* SIP, in particular, sends back this error code to indicate an
2264 * overlap dialled number needs more digits. */
2265 if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
2266 res = AST_PBX_INCOMPLETE;
2269 /* almost done, although the 'else' block is 400 lines */
2273 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER))
2276 strcpy(pa.status, "ANSWER");
2277 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2278 /* Ah ha! Someone answered within the desired timeframe. Of course after this
2279 we will always return with -1 so that it is hung up properly after the
2281 hanguptree(outgoing, peer, 1);
2283 /* If appropriate, log that we have a destination channel and set the answer time */
2285 ast_cdr_setdestchan(chan->cdr, peer->name);
2286 ast_cdr_setanswer(chan->cdr, peer->cdr->answer);
2289 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
2291 ast_channel_lock(peer);
2292 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
2295 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
2296 ast_channel_unlock(peer);
2298 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
2299 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
2300 ast_channel_sendurl( peer, args.url );
2302 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
2303 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
2308 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
2312 struct ast_channel *chans[2];
2313 struct ast_channel *active_chan;
2318 /* we need to stream the announcment while monitoring the caller for a hangup */
2320 /* stream the file */
2321 res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
2324 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
2327 ast_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
2328 while (peer->stream) {
2331 ms = ast_sched_wait(peer->sched);
2333 if (ms < 0 && !peer->timingfunc) {
2334 ast_stopstream(peer);
2340 active_chan = ast_waitfor_n(chans, 2, &ms);
2342 struct ast_frame *fr = ast_read(active_chan);
2348 switch(fr->frametype) {
2349 case AST_FRAME_DTMF_END:
2350 digit = fr->subclass.integer;
2351 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
2352 ast_stopstream(peer);
2353 res = ast_senddigit(chan, digit, 0);
2356 case AST_FRAME_CONTROL:
2357 switch (fr->subclass.integer) {
2358 case AST_CONTROL_HANGUP:
2368 /* Ignore all others */
2373 ast_sched_runq(peer->sched);
2375 ast_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
2378 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
2379 /* chan and peer are going into the PBX, they both
2380 * should probably get CDR records. */
2381 ast_clear_flag(chan->cdr, AST_CDR_FLAG_DIALED);
2382 ast_clear_flag(peer->cdr, AST_CDR_FLAG_DIALED);
2384 replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
2385 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
2386 /* peer goes to the same context and extension as chan, so just copy info from chan*/
2387 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2388 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2389 peer->priority = chan->priority + 2;
2390 ast_pbx_start(peer);
2391 hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
2398 if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
2399 struct ast_app *theapp;
2400 const char *macro_result;
2402 res = ast_autoservice_start(chan);
2404 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2408 theapp = pbx_findapp("Macro");
2410 if (theapp && !res) { /* XXX why check res here ? */
2411 /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
2412 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2413 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2415 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
2416 res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
2417 ast_debug(1, "Macro exited with status %d\n", res);
2420 ast_log(LOG_ERROR, "Could not find application Macro\n");
2424 if (ast_autoservice_stop(chan) < 0) {
2428 ast_channel_lock(peer);
2430 if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
2431 char *macro_transfer_dest;
2433 if (!strcasecmp(macro_result, "BUSY")) {
2434 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2435 ast_set_flag64(peerflags, OPT_GO_ON);
2437 } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
2438 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2439 ast_set_flag64(peerflags, OPT_GO_ON);
2441 } else if (!strcasecmp(macro_result, "CONTINUE")) {
2442 /* hangup peer and keep chan alive assuming the macro has changed
2443 the context / exten / priority or perhaps
2444 the next priority in the current exten is desired.
2446 ast_set_flag64(peerflags, OPT_GO_ON);
2448 } else if (!strcasecmp(macro_result, "ABORT")) {
2449 /* Hangup both ends unless the caller has the g flag */
2451 } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
2453 /* perform a transfer to a new extension */
2454 if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
2455 replace_macro_delimiter(macro_transfer_dest);
2456 if (!ast_parseable_goto(chan, macro_transfer_dest))
2457 ast_set_flag64(peerflags, OPT_GO_ON);
2462 ast_channel_unlock(peer);
2465 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
2466 struct ast_app *theapp;
2467 const char *gosub_result;
2468 char *gosub_args, *gosub_argstart;
2471 res9 = ast_autoservice_start(chan);
2473 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2477 theapp = pbx_findapp("Gosub");
2479 if (theapp && !res9) {
2480 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
2482 /* Set where we came from */
2483 ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
2484 ast_copy_string(peer->exten, "s", sizeof(peer->exten));
2487 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
2488 if (gosub_argstart) {
2489 *gosub_argstart = 0;
2490 if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
2491 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2494 *gosub_argstart = ',';
2496 if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
2497 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2503 res9 = pbx_exec(peer, theapp, gosub_args);
2505 struct ast_pbx_args args;
2506 /* A struct initializer fails to compile for this case ... */
2507 memset(&args, 0, sizeof(args));
2508 args.no_hangup_chan = 1;
2509 ast_pbx_run_args(peer, &args);
2511 ast_free(gosub_args);
2512 ast_debug(1, "Gosub exited with status %d\n", res9);
2514 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
2518 ast_log(LOG_ERROR, "Could not find application Gosub\n");
2522 if (ast_autoservice_stop(chan) < 0) {
2523 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2527 ast_channel_lock(peer);
2529 if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
2530 char *gosub_transfer_dest;
2531 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
2533 /* Inherit return value from the peer, so it can be used in the master */
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