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app_dial.c
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1/*
2 * Asterisk -- An open source telephony toolkit.
3 *
4 * Copyright (C) 1999 - 2012, Digium, Inc.
5 *
6 * Mark Spencer <markster@digium.com>
7 *
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.
13 *
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.
17 */
18
19/*! \file
20 *
21 * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
22 *
23 * \author Mark Spencer <markster@digium.com>
24 *
25 * \ingroup applications
26 */
27
28/*** MODULEINFO
29 <support_level>core</support_level>
30 ***/
31
32
33#include "asterisk.h"
34
35#include <sys/time.h>
36#include <signal.h>
37#include <sys/stat.h>
38#include <netinet/in.h>
39
40#include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
41#include "asterisk/lock.h"
42#include "asterisk/file.h"
43#include "asterisk/channel.h"
44#include "asterisk/pbx.h"
45#include "asterisk/module.h"
46#include "asterisk/translate.h"
47#include "asterisk/say.h"
48#include "asterisk/config.h"
49#include "asterisk/features.h"
50#include "asterisk/musiconhold.h"
51#include "asterisk/callerid.h"
52#include "asterisk/utils.h"
53#include "asterisk/app.h"
54#include "asterisk/causes.h"
55#include "asterisk/rtp_engine.h"
56#include "asterisk/manager.h"
57#include "asterisk/privacy.h"
58#include "asterisk/stringfields.h"
59#include "asterisk/dsp.h"
60#include "asterisk/aoc.h"
61#include "asterisk/ccss.h"
62#include "asterisk/indications.h"
63#include "asterisk/framehook.h"
64#include "asterisk/dial.h"
65#include "asterisk/stasis_channels.h"
66#include "asterisk/bridge_after.h"
67#include "asterisk/features_config.h"
68#include "asterisk/max_forwards.h"
69#include "asterisk/stream.h"
70
71/*** DOCUMENTATION
72 <application name="Dial" language="en_US">
73 <since>
74 <version>0.1.0</version>
75 </since>
76 <synopsis>
77 Attempt to connect to another device or endpoint and bridge the call.
78 </synopsis>
79 <syntax>
80 <parameter name="Technology/Resource" required="false" argsep="&amp;">
81 <argument name="Technology/Resource" required="true">
82 <para>Specification of the device(s) to dial. These must be in the format of
83 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
84 represents a particular channel driver, and <replaceable>Resource</replaceable>
85 represents a resource available to that particular channel driver.</para>
86 </argument>
87 <argument name="Technology2/Resource2" required="false" multiple="true">
88 <para>Optional extra devices to dial in parallel</para>
89 <para>If you need more than one enter them as
90 Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
91 </argument>
92 <xi:include xpointer="xpointer(/docs/info[@name='Dial_Resource'])" />
93 </parameter>
94 <parameter name="timeout" required="false" argsep="^">
95 <para>Specifies the number of seconds we attempt to dial the specified devices.</para>
96 <para>If not specified, this defaults to 136 years.</para>
97 <para>If a second argument is specified, this controls the number of seconds we attempt to dial the specified devices
98 without receiving early media or ringing. If neither progress, ringing, nor voice frames have been received when this
99 timeout expires, the call will be treated as a CHANUNAVAIL. This can be used to skip destinations that may not be responsive.</para>
100 </parameter>
101 <parameter name="options" required="false">
102 <optionlist>
103 <option name="A" argsep=":">
104 <argument name="x">
105 <para>The file to play to the called party</para>
106 </argument>
107 <argument name="y">
108 <para>The file to play to the calling party</para>
109 </argument>
110 <para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
111 is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
112 to be played to the caller. The files may be different and will be played to each party
113 simultaneously.</para>
114 </option>
115 <option name="a">
116 <para>Immediately answer the calling channel when the called channel answers in
117 all cases. Normally, the calling channel is answered when the called channel
118 answers, but when options such as <literal>A()</literal> and
119 <literal>M()</literal> are used, the calling channel is
120 not answered until all actions on the called channel (such as playing an
121 announcement) are completed. This option can be used to answer the calling
122 channel before doing anything on the called channel. You will rarely need to use
123 this option, the default behavior is adequate in most cases.</para>
124 </option>
125 <option name="b" argsep="^">
126 <para>Before initiating an outgoing call, <literal>Gosub</literal> to the specified
127 location using the newly created channel. The <literal>Gosub</literal> will be
128 executed for each destination channel.</para>
129 <argument name="context" required="false" />
130 <argument name="exten" required="false" />
131 <argument name="priority" required="true" hasparams="optional" argsep="^">
132 <argument name="arg1" multiple="true" required="true" />
133 <argument name="argN" />
134 </argument>
135 </option>
136 <option name="B" argsep="^">
137 <para>Before initiating the outgoing call(s), <literal>Gosub</literal> to the
138 specified location using the current channel.</para>
139 <argument name="context" required="false" />
140 <argument name="exten" required="false" />
141 <argument name="priority" required="true" hasparams="optional" argsep="^">
142 <argument name="arg1" multiple="true" required="true" />
143 <argument name="argN" />
144 </argument>
145 </option>
146 <option name="C">
147 <para>Reset the call detail record (CDR) for this call.</para>
148 </option>
149 <option name="c">
150 <para>If the Dial() application cancels this call, always set
151 <variable>HANGUPCAUSE</variable> to 'answered elsewhere'</para>
152 </option>
153 <option name="d">
154 <para>Allow the calling user to dial a 1 digit extension while waiting for
155 a call to be answered. Exit to that extension if it exists in the
156 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
157 if it exists.</para>
158 <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
159 connected. If you wish to use this option with these phones, you
160 can use the <literal>Answer</literal> application before dialing.</para>
161 </option>
162 <option name="D" argsep=":">
163 <argument name="called" />
164 <argument name="calling" />
165 <argument name="progress" />
166 <argument name="mfprogress" />
167 <argument name="mfwink" />
168 <argument name="sfprogress" />
169 <argument name="sfwink" />
170 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
171 party has answered, but before the call gets bridged. The
172 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
173 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
174 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
175 to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
176 <para>See <literal>SendDTMF</literal> for valid digits.</para>
177 <para>If <replaceable>mfprogress</replaceable> is specified, its MF is sent
178 to the called party immediately after receiving a <literal>PROGRESS</literal> message.
179 If <replaceable>mfwink</replaceable> is specified, its MF is sent
180 to the called party immediately after receiving a <literal>WINK</literal> message.</para>
181 <para>See <literal>SendMF</literal> for valid digits.</para>
182 <para>If <replaceable>sfprogress</replaceable> is specified, its SF is sent
183 to the called party immediately after receiving a <literal>PROGRESS</literal> message.
184 If <replaceable>sfwink</replaceable> is specified, its SF is sent
185 to the called party immediately after receiving a <literal>WINK</literal> message.</para>
186 <para>See <literal>SendSF</literal> for valid digits.</para>
187 </option>
188 <option name="E">
189 <para>Enable echoing of sent MF or SF digits back to caller (e.g. "hearpulsing").
190 Used in conjunction with the D option.</para>
191 </option>
192 <option name="e">
193 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
194 </option>
195 <option name="f">
196 <argument name="x" required="false" />
197 <para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
198 deflection to the dialplan extension of this <literal>Dial()</literal> using a dialplan <literal>hint</literal>.
199 For example, some PSTNs do not allow CallerID to be set to anything
200 other than the numbers assigned to you.
201 If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
202 </option>
203 <option name="F" argsep="^">
204 <argument name="context" required="false" />
205 <argument name="exten" required="false" />
206 <argument name="priority" required="true" />
207 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
208 to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
209 <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
210 prefixed with one or two underbars ('_').</para>
211 </option>
212 <option name="F">
213 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
214 and <emphasis>start</emphasis> execution at that location.</para>
215 <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
216 prefixed with one or two underbars ('_').</para>
217 <para>NOTE: Using this option from a GoSub() might not make sense as there would be no return points.</para>
218 </option>
219 <option name="g">
220 <para>Proceed with dialplan execution at the next priority in the current extension if the
221 destination channel hangs up.</para>
222 </option>
223 <option name="G" argsep="^">
224 <argument name="context" required="false" />
225 <argument name="exten" required="false" />
226 <argument name="priority" required="true" />
227 <para>If the call is answered, transfer the calling party to
228 the specified <replaceable>priority</replaceable> and the called party to the specified
229 <replaceable>priority</replaceable> plus one.</para>
230 <para>NOTE: You cannot use any additional action post answer options in conjunction with this option.</para>
231 </option>
232 <option name="h">
233 <para>Allow the called party to hang up by sending the DTMF sequence
234 defined for disconnect in <filename>features.conf</filename>.</para>
235 </option>
236 <option name="H">
237 <para>Allow the calling party to hang up by sending the DTMF sequence
238 defined for disconnect in <filename>features.conf</filename>.</para>
239 <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
240 connected. If you wish to allow DTMF disconnect before the dialed
241 party answers with these phones, you can use the <literal>Answer</literal>
242 application before dialing.</para>
243 </option>
244 <option name="i">
245 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
246 </option>
247 <option name="I">
248 <para>Asterisk will ignore any connected line update requests or any redirecting party
249 update requests it may receive on this dial attempt.</para>
250 </option>
251 <option name="j">
252 <para>Use the initial stream topology of the caller for outgoing channels, even if the caller topology has changed.</para>
253 <para>NOTE: For this option to work, it has to be present in all invocations of Dial that the caller channel goes through.</para>
254 </option>
255 <option name="k">
256 <para>Allow the called party to enable parking of the call by sending
257 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
258 </option>
259 <option name="K">
260 <para>Allow the calling party to enable parking of the call by sending
261 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
262 </option>
263 <option name="L" argsep=":">
264 <argument name="x" required="true">
265 <para>Maximum call time, in milliseconds</para>
266 </argument>
267 <argument name="y">
268 <para>Warning time, in milliseconds</para>
269 </argument>
270 <argument name="z">
271 <para>Repeat time, in milliseconds</para>
272 </argument>
273 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
274 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
275 <para>This option is affected by the following variables:</para>
276 <variablelist>
277 <variable name="LIMIT_PLAYAUDIO_CALLER">
278 <value name="yes" default="true" />
279 <value name="no" />
280 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
281 </variable>
282 <variable name="LIMIT_PLAYAUDIO_CALLEE">
283 <value name="yes" />
284 <value name="no" default="true"/>
285 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
286 </variable>
287 <variable name="LIMIT_TIMEOUT_FILE">
288 <value name="filename"/>
289 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
290 If not set, the time remaining will be announced.</para>
291 </variable>
292 <variable name="LIMIT_CONNECT_FILE">
293 <value name="filename"/>
294 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
295 If not set, the time remaining will be announced.</para>
296 </variable>
297 <variable name="LIMIT_WARNING_FILE">
298 <value name="filename"/>
299 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
300 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
301 </variable>
302 </variablelist>
303 </option>
304 <option name="m">
305 <argument name="class" required="false"/>
306 <para>Provide hold music to the calling party until a requested
307 channel answers. A specific music on hold <replaceable>class</replaceable>
308 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
309 </option>
310 <option name="n">
311 <argument name="delete">
312 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
313 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
314 yet answered.</para>
315 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
316 always be deleted.</para>
317 </argument>
318 <para>This option is a modifier for the call screening/privacy mode. (See the
319 <literal>p</literal> and <literal>P</literal> options.) It specifies
320 that no introductions are to be saved in the <directory>priv-callerintros</directory>
321 directory.</para>
322 </option>
323 <option name="N">
324 <para>This option is a modifier for the call screening/privacy mode. It specifies
325 that if CallerID is present, do not screen the call.</para>
326 </option>
327 <option name="o">
328 <argument name="x" required="false" />
329 <para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
330 <emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
331 This was the behavior of Asterisk 1.0 and earlier.
332 If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
333 Note that <literal>o(${CALLERID(all)})</literal> is similar to option <literal>o</literal> without the parameter.</para>
334 </option>
335 <option name="O">
336 <argument name="mode">
337 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
338 the originator hanging up will cause the phone to ring back immediately.</para>
339 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
340 flashes the trunk, it will ring their phone back.</para>
341 </argument>
342 <para>Enables <emphasis>operator services</emphasis> mode. This option only
343 works when bridging a DAHDI channel to another DAHDI channel
344 only. If specified on non-DAHDI interfaces, it will be ignored.
345 When the destination answers (presumably an operator services
346 station), the originator no longer has control of their line.
347 They may hang up, but the switch will not release their line
348 until the destination party (the operator) hangs up.</para>
349 </option>
350 <option name="p">
351 <para>This option enables screening mode. This is basically Privacy mode
352 without memory.</para>
353 </option>
354 <option name="P">
355 <argument name="x" />
356 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
357 it is provided. The current extension is used if a database family/key is not specified.</para>
358 </option>
359 <option name="Q">
360 <argument name="cause" required="true"/>
361 <para>Specify the Q.850/Q.931 <replaceable>cause</replaceable> to send on
362 unanswered channels when another channel answers the call.
363 As with <literal>Hangup()</literal>, <replaceable>cause</replaceable>
364 can be a numeric cause code or a name such as
365 <literal>NO_ANSWER</literal>,
366 <literal>USER_BUSY</literal>,
367 <literal>CALL_REJECTED</literal> or
368 <literal>ANSWERED_ELSEWHERE</literal> (the default if Q isn't specified).
369 You can also specify <literal>0</literal> or <literal>NONE</literal>
370 to send no cause. See the <filename>causes.h</filename> file for the
371 full list of valid causes and names.
372 </para>
373 </option>
374 <option name="r">
375 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
376 party until the called channel has answered.</para>
377 <argument name="tone" required="false">
378 <para>Indicate progress to calling party. Send audio 'tone' from the <filename>indications.conf</filename> tonezone currently in use.</para>
379 </argument>
380 </option>
381 <option name="R">
382 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
383 Allow interruption of the ringback if early media is received on the channel.</para>
384 </option>
385 <option name="S">
386 <argument name="x" required="true" />
387 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
388 answered the call.</para>
389 </option>
390 <option name="s">
391 <argument name="x" required="true" />
392 <para>Force the outgoing CallerID tag parameter to be set to the string <replaceable>x</replaceable>.</para>
393 <para>Works with the <literal>f</literal> option.</para>
394 </option>
395 <option name="t">
396 <para>Allow the called party to transfer the calling party by sending the
397 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
398 transfers initiated by other methods.</para>
399 </option>
400 <option name="T">
401 <para>Allow the calling party to transfer the called party by sending the
402 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
403 transfers initiated by other methods.</para>
404 </option>
405 <option name="U" argsep="^">
406 <argument name="x" required="true">
407 <para>Name of the subroutine context to execute via <literal>Gosub</literal>.
408 The subroutine execution starts in the named context at the s exten and priority 1.</para>
409 </argument>
410 <argument name="arg" multiple="true" required="false">
411 <para>Arguments for the <literal>Gosub</literal> routine</para>
412 </argument>
413 <para>Execute via <literal>Gosub</literal> the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
414 to the calling channel. Arguments can be specified to the <literal>Gosub</literal>
415 using <literal>^</literal> as a delimiter. The <literal>Gosub</literal> routine can set the variable
416 <variable>GOSUB_RESULT</variable> to specify the following actions after the <literal>Gosub</literal> returns.</para>
417 <variablelist>
418 <variable name="GOSUB_RESULT">
419 <value name="ABORT">
420 Hangup both legs of the call.
421 </value>
422 <value name="CONGESTION">
423 Behave as if line congestion was encountered.
424 </value>
425 <value name="BUSY">
426 Behave as if a busy signal was encountered.
427 </value>
428 <value name="CONTINUE">
429 Hangup the called party and allow the calling party
430 to continue dialplan execution at the next priority.
431 </value>
432 <value name="GOTO:[[&lt;context&gt;^]&lt;exten&gt;^]&lt;priority&gt;">
433 Transfer the call to the specified destination.
434 </value>
435 </variable>
436 </variablelist>
437 <para>NOTE: You cannot use any additional action post answer options in conjunction
438 with this option. Also, pbx services are run on the <emphasis>called</emphasis> channel,
439 so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
440 </option>
441 <option name="u">
442 <argument name = "x" required="true">
443 <para>Force the outgoing callerid presentation indicator parameter to be set
444 to one of the values passed in <replaceable>x</replaceable>:
445 <literal>allowed_not_screened</literal>
446 <literal>allowed_passed_screen</literal>
447 <literal>allowed_failed_screen</literal>
448 <literal>allowed</literal>
449 <literal>prohib_not_screened</literal>
450 <literal>prohib_passed_screen</literal>
451 <literal>prohib_failed_screen</literal>
452 <literal>prohib</literal>
453 <literal>unavailable</literal></para>
454 </argument>
455 <para>Works with the <literal>f</literal> option.</para>
456 </option>
457 <option name="w">
458 <para>Allow the called party to enable recording of the call by sending
459 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
460 </option>
461 <option name="W">
462 <para>Allow the calling party to enable recording of the call by sending
463 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
464 </option>
465 <option name="x">
466 <para>Allow the called party to enable recording of the call by sending
467 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
468 </option>
469 <option name="X">
470 <para>Allow the calling party to enable recording of the call by sending
471 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
472 </option>
473 <option name="z">
474 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
475 </option>
476 </optionlist>
477 </parameter>
478 <parameter name="URL">
479 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
480 </parameter>
481 </syntax>
482 <description>
483 <para>This application will place calls to one or more specified channels. As soon
484 as one of the requested channels answers, the originating channel will be
485 answered, if it has not already been answered. These two channels will then
486 be active in a bridged call. All other channels that were requested will then
487 be hung up.</para>
488 <para>Unless there is a timeout specified, the Dial application will wait
489 indefinitely until one of the called channels answers, the user hangs up, or
490 if all of the called channels are busy or unavailable. Dialplan execution will
491 continue if no requested channels can be called, or if the timeout expires.
492 This application will report normal termination if the originating channel
493 hangs up, or if the call is bridged and either of the parties in the bridge
494 ends the call.</para>
495 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
496 application will be put into that group (as in <literal>Set(GROUP()=...</literal>).
497 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
498 application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
499 however, the variable will be unset after use.</para>
500 <example title="Dial with 30 second timeout">
501 same => n,Dial(PJSIP/alice,30)
502 </example>
503 <example title="Parallel dial with 45 second timeout">
504 same => n,Dial(PJSIP/alice&amp;PJIP/bob,45)
505 </example>
506 <example title="Dial with 'g' continuation option">
507 same => n,Dial(PJSIP/alice,,g)
508 same => n,Log(NOTICE, Alice call result: ${DIALSTATUS})
509 </example>
510 <example title="Dial with transfer/recording features for calling party">
511 same => n,Dial(PJSIP/alice,,TX)
512 </example>
513 <example title="Dial with call length limit">
514 same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
515 </example>
516 <example title="Dial alice and bob and send NO_ANSWER to bob instead of ANSWERED_ELSEWHERE when alice answers">
517 same => n,Dial(PJSIP/alice&amp;PJSIP/bob,,Q(NO_ANSWER))
518 </example>
519 <example title="Dial with pre-dial subroutines">
520 [default]
521 exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
522 same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
523 same => n,Return()
524 exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
525 same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
526 same => n,Return()
527 exten => _X.,1,NoOp()
528 same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
529 same => n,Hangup()
530 </example>
531 <example title="Dial with post-answer subroutine executed on outbound channel">
532 [my_gosub_routine]
533 exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
534 same => n,Playback(hello)
535 same => n,Return()
536 [default]
537 exten => _X.,1,NoOp()
538 same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
539 same => n,Hangup()
540 </example>
541 <example title="Dial into ConfBridge using 'G' option">
542 same => n,Dial(PJSIP/alice,,G(jump_to_here))
543 same => n(jump_to_here),Goto(confbridge)
544 same => n,Goto(confbridge)
545 same => n(confbridge),ConfBridge(${EXTEN})
546 </example>
547 <para>This application sets the following channel variables:</para>
548 <variablelist>
549 <variable name="DIALEDTIME">
550 <para>This is the time from dialing a channel until when it is disconnected.</para>
551 </variable>
552 <variable name="DIALEDTIME_MS">
553 <para>This is the milliseconds version of the DIALEDTIME variable.</para>
554 </variable>
555 <variable name="ANSWEREDTIME">
556 <para>This is the amount of time for actual call.</para>
557 </variable>
558 <variable name="ANSWEREDTIME_MS">
559 <para>This is the milliseconds version of the ANSWEREDTIME variable.</para>
560 </variable>
561 <variable name="RINGTIME">
562 <para>This is the time from creating the channel to the first RINGING event received. Empty if there was no ring.</para>
563 </variable>
564 <variable name="RINGTIME_MS">
565 <para>This is the milliseconds version of the RINGTIME variable.</para>
566 </variable>
567 <variable name="PROGRESSTIME">
568 <para>This is the time from creating the channel to the first PROGRESS event received. Empty if there was no such event.</para>
569 </variable>
570 <variable name="PROGRESSTIME_MS">
571 <para>This is the milliseconds version of the PROGRESSTIME variable.</para>
572 </variable>
573 <variable name="DIALEDPEERNAME">
574 <para>The name of the outbound channel that answered the call.</para>
575 </variable>
576 <variable name="DIALEDPEERNUMBER">
577 <para>The number that was dialed for the answered outbound channel.</para>
578 </variable>
579 <variable name="FORWARDERNAME">
580 <para>If a call forward occurred, the name of the forwarded channel.</para>
581 </variable>
582 <variable name="DIALSTATUS">
583 <para>This is the status of the call</para>
584 <value name="CHANUNAVAIL">
585 Either the dialed peer exists but is not currently reachable, e.g.
586 endpoint is not registered, or an attempt was made to call a
587 nonexistent location, e.g. nonexistent DNS hostname.
588 </value>
589 <value name="CONGESTION">
590 Channel or switching congestion occurred when routing the call.
591 This can occur if there is a slow or no response from the remote end.
592 </value>
593 <value name="NOANSWER">
594 Called party did not answer.
595 </value>
596 <value name="BUSY">
597 The called party was busy or indicated a busy status.
598 Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
599 modes are active. In this case, you can use DEVICE_STATUS to check if the
600 endpoint is actually in use, if needed.
601 </value>
602 <value name="ANSWER">
603 The call was answered.
604 Any other result implicitly indicates the call was not answered.
605 </value>
606 <value name="CANCEL">
607 Dial was cancelled before call was answered or reached some other terminating event.
608 </value>
609 <value name="DONTCALL">
610 For the Privacy and Screening Modes.
611 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
612 </value>
613 <value name="TORTURE">
614 For the Privacy and Screening Modes.
615 Will be set if the called party chooses to send the calling party to the 'torture' script.
616 </value>
617 <value name="INVALIDARGS">
618 Dial failed due to invalid syntax.
619 </value>
620 </variable>
621 </variablelist>
622 </description>
623 <see-also>
624 <ref type="application">RetryDial</ref>
625 <ref type="application">SendDTMF</ref>
626 <ref type="application">Gosub</ref>
627 </see-also>
628 </application>
629 <application name="RetryDial" language="en_US">
630 <since>
631 <version>1.2.0</version>
632 </since>
633 <synopsis>
634 Place a call, retrying on failure allowing an optional exit extension.
635 </synopsis>
636 <syntax>
637 <parameter name="announce" required="true">
638 <para>Filename of sound that will be played when no channel can be reached</para>
639 </parameter>
640 <parameter name="sleep" required="true">
641 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
642 </parameter>
643 <parameter name="retries" required="true">
644 <para>Number of retries</para>
645 <para>When this is reached flow will continue at the next priority in the dialplan</para>
646 </parameter>
647 <parameter name="dialargs" required="true">
648 <para>Same format as arguments provided to the Dial application</para>
649 </parameter>
650 </syntax>
651 <description>
652 <para>This application will attempt to place a call using the normal Dial application.
653 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
654 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
655 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
656 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
657 While waiting to retry a call, a 1 digit extension may be dialed. If that
658 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
659 one, The call will jump to that extension immediately.
660 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
661 to the Dial application.</para>
662 </description>
663 <see-also>
664 <ref type="application">Dial</ref>
665 </see-also>
666 </application>
667 ***/
668
669static const char app[] = "Dial";
670static const char rapp[] = "RetryDial";
671
672enum {
673 OPT_ANNOUNCE = (1 << 0),
674 OPT_RESETCDR = (1 << 1),
675 OPT_DTMF_EXIT = (1 << 2),
676 OPT_SENDDTMF = (1 << 3),
677 OPT_FORCECLID = (1 << 4),
678 OPT_GO_ON = (1 << 5),
679 OPT_CALLEE_HANGUP = (1 << 6),
680 OPT_CALLER_HANGUP = (1 << 7),
681 OPT_ORIGINAL_CLID = (1 << 8),
682 OPT_DURATION_LIMIT = (1 << 9),
683 OPT_MUSICBACK = (1 << 10),
684 OPT_SCREEN_NOINTRO = (1 << 12),
685 OPT_SCREEN_NOCALLERID = (1 << 13),
686 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
687 OPT_SCREENING = (1 << 15),
688 OPT_PRIVACY = (1 << 16),
689 OPT_RINGBACK = (1 << 17),
690 OPT_DURATION_STOP = (1 << 18),
691 OPT_CALLEE_TRANSFER = (1 << 19),
692 OPT_CALLER_TRANSFER = (1 << 20),
693 OPT_CALLEE_MONITOR = (1 << 21),
694 OPT_CALLER_MONITOR = (1 << 22),
695 OPT_GOTO = (1 << 23),
696 OPT_OPERMODE = (1 << 24),
697 OPT_CALLEE_PARK = (1 << 25),
698 OPT_CALLER_PARK = (1 << 26),
699 OPT_IGNORE_FORWARDING = (1 << 27),
700 OPT_CALLEE_GOSUB = (1 << 28),
701 OPT_CALLEE_MIXMONITOR = (1 << 29),
702 OPT_CALLER_MIXMONITOR = (1 << 30),
703};
704
705/* flags are now 64 bits, so keep it up! */
706#define DIAL_STILLGOING (1LLU << 31)
707#define DIAL_NOFORWARDHTML (1LLU << 32)
708#define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */
709#define OPT_CANCEL_ELSEWHERE (1LLU << 34)
710#define OPT_PEER_H (1LLU << 35)
711#define OPT_CALLEE_GO_ON (1LLU << 36)
712#define OPT_CANCEL_TIMEOUT (1LLU << 37)
713#define OPT_FORCE_CID_TAG (1LLU << 38)
714#define OPT_FORCE_CID_PRES (1LLU << 39)
715#define OPT_CALLER_ANSWER (1LLU << 40)
716#define OPT_PREDIAL_CALLEE (1LLU << 41)
717#define OPT_PREDIAL_CALLER (1LLU << 42)
718#define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
719#define OPT_HANGUPCAUSE (1LLU << 44)
720#define OPT_HEARPULSING (1LLU << 45)
721#define OPT_TOPOLOGY_PRESERVE (1LLU << 46)
722
723enum {
724 OPT_ARG_ANNOUNCE = 0,
725 OPT_ARG_SENDDTMF,
726 OPT_ARG_GOTO,
727 OPT_ARG_DURATION_LIMIT,
728 OPT_ARG_MUSICBACK,
729 OPT_ARG_RINGBACK,
730 OPT_ARG_CALLEE_GOSUB,
731 OPT_ARG_CALLEE_GO_ON,
732 OPT_ARG_PRIVACY,
733 OPT_ARG_DURATION_STOP,
734 OPT_ARG_OPERMODE,
735 OPT_ARG_SCREEN_NOINTRO,
736 OPT_ARG_ORIGINAL_CLID,
737 OPT_ARG_FORCECLID,
738 OPT_ARG_FORCE_CID_TAG,
739 OPT_ARG_FORCE_CID_PRES,
740 OPT_ARG_PREDIAL_CALLEE,
741 OPT_ARG_PREDIAL_CALLER,
742 OPT_ARG_HANGUPCAUSE,
743 /* note: this entry _MUST_ be the last one in the enum */
744 OPT_ARG_ARRAY_SIZE
745};
746
747AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
748 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
749 AST_APP_OPTION('a', OPT_CALLER_ANSWER),
750 AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
751 AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
752 AST_APP_OPTION('C', OPT_RESETCDR),
753 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
754 AST_APP_OPTION('d', OPT_DTMF_EXIT),
755 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
756 AST_APP_OPTION('E', OPT_HEARPULSING),
757 AST_APP_OPTION('e', OPT_PEER_H),
758 AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
759 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
760 AST_APP_OPTION('g', OPT_GO_ON),
761 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
762 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
763 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
764 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
765 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
766 AST_APP_OPTION('j', OPT_TOPOLOGY_PRESERVE),
767 AST_APP_OPTION('k', OPT_CALLEE_PARK),
768 AST_APP_OPTION('K', OPT_CALLER_PARK),
769 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
770 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
771 AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
772 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
773 AST_APP_OPTION_ARG('o', OPT_ORIGINAL_CLID, OPT_ARG_ORIGINAL_CLID),
774 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
775 AST_APP_OPTION('p', OPT_SCREENING),
776 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
777 AST_APP_OPTION_ARG('Q', OPT_HANGUPCAUSE, OPT_ARG_HANGUPCAUSE),
778 AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
779 AST_APP_OPTION('R', OPT_RING_WITH_EARLY_MEDIA),
780 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
781 AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
782 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
783 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
784 AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
785 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
786 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
787 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
788 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
789 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
790 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
791END_OPTIONS );
792
793#define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
794 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
795 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
796 OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_GOSUB) && \
797 !ast_channel_audiohooks(chan) && !ast_channel_audiohooks(peer) && \
798 ast_framehook_list_is_empty(ast_channel_framehooks(chan)) && ast_framehook_list_is_empty(ast_channel_framehooks(peer)))
799
800/*
801 * The list of active channels
802 */
803struct chanlist {
804 AST_LIST_ENTRY(chanlist) node;
805 struct ast_channel *chan;
806 /*! Channel interface dialing string (is tech/number). (Stored in stuff[]) */
807 const char *interface;
808 /*! Channel technology name. (Stored in stuff[]) */
809 const char *tech;
810 /*! Channel device addressing. (Stored in stuff[]) */
811 const char *number;
812 /*! Original channel name. Must be freed. Could be NULL if allocation failed. */
813 char *orig_chan_name;
814 uint64_t flags;
815 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
816 struct ast_party_connected_line connected;
817 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
818 unsigned int pending_connected_update:1;
819 struct ast_aoc_decoded *aoc_s_rate_list;
820 /*! The interface, tech, and number strings are stuffed here. */
821 char stuff[0];
822};
823
824AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
825
826static void topology_ds_destroy(void *data) {
827 struct ast_stream_topology *top = data;
828 ast_stream_topology_free(top);
829}
830
831static const struct ast_datastore_info topology_ds_info = {
832 .type = "app_dial_topology_preserve",
833 .destroy = topology_ds_destroy,
834};
835
836static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
837
838static void chanlist_free(struct chanlist *outgoing)
839{
840 ast_party_connected_line_free(&outgoing->connected);
841 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
842 ast_free(outgoing->orig_chan_name);
843 ast_free(outgoing);
844}
845
846static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
847{
848 /* Hang up a tree of stuff */
849 struct chanlist *outgoing;
850
851 while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
852 /* Hangup any existing lines we have open */
853 if (outgoing->chan && (outgoing->chan != exception)) {
854 if (hangupcause >= 0) {
855 /* This is for the channel drivers */
856 ast_channel_hangupcause_set(outgoing->chan, hangupcause);
857 }
858 ast_hangup(outgoing->chan);
859 }
860 chanlist_free(outgoing);
861 }
862}
863
864#define AST_MAX_WATCHERS 256
865
866/*
867 * argument to handle_cause() and other functions.
868 */
869struct cause_args {
870 struct ast_channel *chan;
871 int busy;
872 int congestion;
873 int nochan;
874};
875
876static void handle_cause(int cause, struct cause_args *num)
877{
878 switch(cause) {
879 case AST_CAUSE_BUSY:
880 num->busy++;
881 break;
882 case AST_CAUSE_CONGESTION:
883 num->congestion++;
884 break;
885 case AST_CAUSE_NO_ROUTE_DESTINATION:
886 case AST_CAUSE_UNREGISTERED:
887 num->nochan++;
888 break;
889 case AST_CAUSE_NO_ANSWER:
890 case AST_CAUSE_NORMAL_CLEARING:
891 break;
892 default:
893 num->nochan++;
894 break;
895 }
896}
897
898static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
899{
900 char rexten[2] = { exten, '\0' };
901
902 if (context) {
903 if (!ast_goto_if_exists(chan, context, rexten, pri))
904 return 1;
905 } else {
906 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
907 return 1;
908 }
909 return 0;
910}
911
912/* do not call with chan lock held */
913static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
914{
915 const char *context;
916 const char *exten;
917
918 ast_channel_lock(chan);
919 context = ast_strdupa(ast_channel_context(chan));
920 exten = ast_strdupa(ast_channel_exten(chan));
921 ast_channel_unlock(chan);
922
923 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
924}
925
926/*!
927 * helper function for wait_for_answer()
928 *
929 * \param o Outgoing call channel list.
930 * \param num Incoming call channel cause accumulation
931 * \param peerflags Dial option flags
932 * \param single TRUE if there is only one outgoing call.
933 * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
934 * \param to Remaining call timeout time.
935 * \param forced_clid OPT_FORCECLID caller id to send
936 * \param stored_clid Caller id representing the called party if needed
937 *
938 * XXX this code is highly suspicious, as it essentially overwrites
939 * the outgoing channel without properly deleting it.
940 *
941 * \todo eventually this function should be integrated into and replaced by ast_call_forward()
942 */
943static void do_forward(struct chanlist *o, struct cause_args *num,
944 struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
945 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
946{
947 char tmpchan[256];
948 char forwarder[AST_CHANNEL_NAME];
949 struct ast_channel *original = o->chan;
950 struct ast_channel *c = o->chan; /* the winner */
951 struct ast_channel *in = num->chan; /* the input channel */
952 char *stuff;
953 char *tech;
954 int cause;
955 struct ast_party_caller caller;
956
957 ast_copy_string(forwarder, ast_channel_name(c), sizeof(forwarder));
958 ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
959 if ((stuff = strchr(tmpchan, '/'))) {
960 *stuff++ = '\0';
961 tech = tmpchan;
962 } else {
963 const char *forward_context;
964 ast_channel_lock(c);
965 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
966 if (ast_strlen_zero(forward_context)) {
967 forward_context = NULL;
968 }
969 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
970 ast_channel_unlock(c);
971 stuff = tmpchan;
972 tech = "Local";
973 }
974 if (!strcasecmp(tech, "Local")) {
975 /*
976 * Drop the connected line update block for local channels since
977 * this is going to run dialplan and the user can change his
978 * mind about what connected line information he wants to send.
979 */
980 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
981 }
982
983 /* Before processing channel, go ahead and check for forwarding */
984 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
985 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
986 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
987 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
988 ast_channel_publish_dial_forward(in, original, NULL, NULL, "CANCEL",
989 ast_channel_call_forward(original));
990 c = o->chan = NULL;
991 cause = AST_CAUSE_BUSY;
992 } else {
993 struct ast_stream_topology *topology;
994
995 ast_channel_lock(in);
996 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(in));
997 ast_channel_unlock(in);
998
999 /* Setup parameters */
1000 c = o->chan = ast_request_with_stream_topology(tech, topology, NULL, in, stuff, &cause);
1001
1002 ast_stream_topology_free(topology);
1003
1004 if (c) {
1005 if (single && !caller_entertained) {
1006 ast_channel_make_compatible(in, o->chan);
1007 }
1008 ast_channel_lock_both(in, o->chan);
1009 ast_channel_inherit_variables(in, o->chan);
1010 ast_channel_datastore_inherit(in, o->chan);
1011 pbx_builtin_setvar_helper(o->chan, "FORWARDERNAME", forwarder);
1012 ast_max_forwards_decrement(o->chan);
1013 ast_channel_unlock(in);
1014 ast_channel_unlock(o->chan);
1015 /* When a call is forwarded, we don't want to track new interfaces
1016 * dialed for CC purposes. Setting the done flag will ensure that
1017 * any Dial operations that happen later won't record CC interfaces.
1018 */
1019 ast_ignore_cc(o->chan);
1020 ast_verb(3, "Not accepting call completion offers from call-forward recipient %s\n",
1021 ast_channel_name(o->chan));
1022 } else
1023 ast_log(LOG_NOTICE,
1024 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
1025 tech, stuff, cause);
1026 }
1027 if (!c) {
1028 ast_channel_publish_dial(in, original, stuff, "BUSY");
1029 ast_clear_flag64(o, DIAL_STILLGOING);
1030 handle_cause(cause, num);
1031 ast_hangup(original);
1032 } else {
1033 ast_channel_lock_both(c, original);
1034 ast_party_redirecting_copy(ast_channel_redirecting(c),
1035 ast_channel_redirecting(original));
1036 ast_channel_unlock(c);
1037 ast_channel_unlock(original);
1038
1039 ast_channel_lock_both(c, in);
1040
1041 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
1042 ast_rtp_instance_early_bridge_make_compatible(c, in);
1043 }
1044
1045 if (!ast_channel_redirecting(c)->from.number.valid
1046 || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
1047 /*
1048 * The call was not previously redirected so it is
1049 * now redirected from this number.
1050 */
1051 ast_party_number_free(&ast_channel_redirecting(c)->from.number);
1052 ast_party_number_init(&ast_channel_redirecting(c)->from.number);
1053 ast_channel_redirecting(c)->from.number.valid = 1;
1054 ast_channel_redirecting(c)->from.number.str =
1055 ast_strdup(ast_channel_exten(in));
1056 }
1057
1058 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
1059
1060 /* Determine CallerID to store in outgoing channel. */
1061 ast_party_caller_set_init(&caller, ast_channel_caller(c));
1062 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1063 caller.id = *stored_clid;
1064 ast_channel_set_caller_event(c, &caller, NULL);
1065 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1066 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
1067 ast_channel_caller(c)->id.number.str, NULL))) {
1068 /*
1069 * The new channel has no preset CallerID number by the channel
1070 * driver. Use the dialplan extension and hint name.
1071 */
1072 caller.id = *stored_clid;
1073 ast_channel_set_caller_event(c, &caller, NULL);
1074 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1075 } else {
1076 ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
1077 }
1078
1079 /* Determine CallerID for outgoing channel to send. */
1080 if (ast_test_flag64(o, OPT_FORCECLID)) {
1081 struct ast_party_connected_line connected;
1082
1083 ast_party_connected_line_init(&connected);
1084 connected.id = *forced_clid;
1085 ast_party_connected_line_copy(ast_channel_connected(c), &connected);
1086 } else {
1087 ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
1088 }
1089
1090 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
1091
1092 ast_channel_appl_set(c, "AppDial");
1093 ast_channel_data_set(c, "(Outgoing Line)");
1094 ast_channel_publish_snapshot(c);
1095
1096 ast_channel_unlock(in);
1097 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1098 struct ast_party_redirecting redirecting;
1099
1100 /*
1101 * Redirecting updates to the caller make sense only on single
1102 * calls.
1103 *
1104 * Need to re-evalute if unlocking is still required here as macro is gone
1105 */
1106 ast_party_redirecting_init(&redirecting);
1107 ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
1108 ast_channel_unlock(c);
1109 if (ast_channel_redirecting_sub(c, in, &redirecting, 0)) {
1110 ast_channel_update_redirecting(in, &redirecting, NULL);
1111 }
1112 ast_party_redirecting_free(&redirecting);
1113 } else {
1114 ast_channel_unlock(c);
1115 }
1116
1117 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1118 *to = -1;
1119 }
1120
1121 if (ast_call(c, stuff, 0)) {
1122 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1123 tech, stuff);
1124 ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1125 ast_clear_flag64(o, DIAL_STILLGOING);
1126 ast_hangup(original);
1127 ast_hangup(c);
1128 c = o->chan = NULL;
1129 num->nochan++;
1130 } else {
1131 ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1132 ast_channel_call_forward(original));
1133
1134 ast_channel_publish_dial(in, c, stuff, NULL);
1135
1136 /* Hangup the original channel now, in case we needed it */
1137 ast_hangup(original);
1138 }
1139 if (single && !caller_entertained) {
1140 ast_indicate(in, -1);
1141 }
1142 }
1143}
1144
1145/* argument used for some functions. */
1146struct privacy_args {
1147 int sentringing;
1148 int privdb_val;
1149 char privcid[256];
1150 char privintro[1024];
1151 char status[256];
1152 int canceled;
1153};
1154
1155static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
1156{
1157 struct chanlist *outgoing;
1158 AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1159 if (!outgoing->chan || outgoing->chan == exception) {
1160 continue;
1161 }
1162 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1163 }
1164}
1165
1166/*!
1167 * \internal
1168 * \brief Update connected line on chan from peer.
1169 * \since 13.6.0
1170 *
1171 * \param chan Channel to get connected line updated.
1172 * \param peer Channel providing connected line information.
1173 * \param is_caller Non-zero if chan is the calling channel.
1174 */
1175static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
1176{
1177 struct ast_party_connected_line connected_caller;
1178
1179 ast_party_connected_line_init(&connected_caller);
1180
1181 ast_channel_lock(peer);
1182 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1183 ast_channel_unlock(peer);
1184 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1185 if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)) {
1186 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1187 }
1188 ast_party_connected_line_free(&connected_caller);
1189}
1190
1191/*!
1192 * \internal
1193 * \pre chan is locked
1194 */
1195static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
1196{
1197 char buf[32];
1198 char full_var_name[128];
1199
1200 snprintf(buf, sizeof(buf), "%" PRId64, duration / 1000);
1201 pbx_builtin_setvar_helper(chan, var_base, buf);
1202
1203 snprintf(full_var_name, sizeof(full_var_name), "%s_MS", var_base);
1204 snprintf(buf, sizeof(buf), "%" PRId64, duration);
1205 pbx_builtin_setvar_helper(chan, full_var_name, buf);
1206}
1207
1208static struct ast_channel *wait_for_answer(struct ast_channel *in,
1209 struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags,
1210 char *opt_args[],
1211 struct privacy_args *pa,
1212 const struct cause_args *num_in, int *result, char *dtmf_progress,
1213 char *mf_progress, char *mf_wink,
1214 char *sf_progress, char *sf_wink,
1215 const int hearpulsing,
1216 const int ignore_cc,
1217 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
1218 struct ast_bridge_config *config)
1219{
1220 struct cause_args num = *num_in;
1221 int prestart = num.busy + num.congestion + num.nochan;
1222 int orig_answer_to = *to_answer;
1223 int orig_progress_to = *to_progress;
1224 struct ast_channel *peer = NULL;
1225 struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1226 /* single is set if only one destination is enabled */
1227 int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1228 int caller_entertained = outgoing
1229 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1230 struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1231 int cc_recall_core_id;
1232 int is_cc_recall;
1233 int cc_frame_received = 0;
1234 int num_ringing = 0;
1235 int sent_ring = 0;
1236 int sent_progress = 0, sent_wink = 0;
1237 struct timeval start = ast_tvnow();
1238 SCOPE_ENTER(3, "%s\n", ast_channel_name(in));
1239
1240 if (single) {
1241 /* Turn off hold music, etc */
1242 if (!caller_entertained) {
1243 ast_deactivate_generator(in);
1244 /* If we are calling a single channel, and not providing ringback or music, */
1245 /* then, make them compatible for in-band tone purpose */
1246 if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1247 /* If these channels can not be made compatible,
1248 * there is no point in continuing. The bridge
1249 * will just fail if it gets that far.
1250 */
1251 *to_answer = -1;
1252 strcpy(pa->status, "CONGESTION");
1253 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1254 SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
1255 ast_channel_name(in), ast_channel_name(outgoing->chan));
1256 }
1257 }
1258
1259 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1260 && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1261 update_connected_line_from_peer(in, outgoing->chan, 1);
1262 }
1263 }
1264
1265 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1266
1267 while ((*to_answer = ast_remaining_ms(start, orig_answer_to)) && (*to_progress = ast_remaining_ms(start, orig_progress_to)) && !peer) {
1268 struct chanlist *o;
1269 int pos = 0; /* how many channels do we handle */
1270 int numlines = prestart;
1271 struct ast_channel *winner;
1272 struct ast_channel *watchers[AST_MAX_WATCHERS];
1273
1274 watchers[pos++] = in;
1275 AST_LIST_TRAVERSE(out_chans, o, node) {
1276 /* Keep track of important channels */
1277 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1278 watchers[pos++] = o->chan;
1279 numlines++;
1280 }
1281 if (pos == 1) { /* only the input channel is available */
1282 if (numlines == (num.busy + num.congestion + num.nochan)) {
1283 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1284 if (num.busy)
1285 strcpy(pa->status, "BUSY");
1286 else if (num.congestion)
1287 strcpy(pa->status, "CONGESTION");
1288 else if (num.nochan)
1289 strcpy(pa->status, "CHANUNAVAIL");
1290 } else {
1291 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1292 }
1293 *to_answer = 0;
1294 if (is_cc_recall) {
1295 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1296 }
1297 SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
1298 }
1299
1300 /* If progress timeout is active, use that if it's the shorter of the 2 timeouts. */
1301 winner = ast_waitfor_n(watchers, pos, *to_progress > 0 && (*to_answer < 0 || *to_progress < *to_answer) ? to_progress : to_answer);
1302
1303 AST_LIST_TRAVERSE(out_chans, o, node) {
1304 int res = 0;
1305 struct ast_frame *f;
1306 struct ast_channel *c = o->chan;
1307
1308 if (c == NULL)
1309 continue;
1310 if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1311 if (!peer) {
1312 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1313 if (o->orig_chan_name
1314 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1315 /*
1316 * The channel name changed so we must generate COLP update.
1317 * Likely because a call pickup channel masqueraded in.
1318 */
1319 update_connected_line_from_peer(in, c, 1);
1320 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1321 if (o->pending_connected_update) {
1322 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1323 ast_channel_update_connected_line(in, &o->connected, NULL);
1324 }
1325 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1326 update_connected_line_from_peer(in, c, 1);
1327 }
1328 }
1329 if (o->aoc_s_rate_list) {
1330 size_t encoded_size;
1331 struct ast_aoc_encoded *encoded;
1332 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1333 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1334 ast_aoc_destroy_encoded(encoded);
1335 }
1336 }
1337 peer = c;
1338 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1339 ast_copy_flags64(peerflags, o,
1340 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1341 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1342 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1343 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1344 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1345 DIAL_NOFORWARDHTML);
1346 ast_channel_dialcontext_set(c, "");
1347 ast_channel_exten_set(c, "");
1348 }
1349 continue;
1350 }
1351 if (c != winner)
1352 continue;
1353 /* here, o->chan == c == winner */
1354 if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1355 pa->sentringing = 0;
1356 if (!ignore_cc && (f = ast_read(c))) {
1357 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1358 /* This channel is forwarding the call, and is capable of CC, so
1359 * be sure to add the new device interface to the list
1360 */
1361 ast_handle_cc_control_frame(in, c, f->data.ptr);
1362 }
1363 ast_frfree(f);
1364 }
1365
1366 if (o->pending_connected_update) {
1367 /*
1368 * Re-seed the chanlist's connected line information with
1369 * previously acquired connected line info from the incoming
1370 * channel. The previously acquired connected line info could
1371 * have been set through the CONNECTED_LINE dialplan function.
1372 */
1373 o->pending_connected_update = 0;
1374 ast_channel_lock(in);
1375 ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1376 ast_channel_unlock(in);
1377 }
1378
1379 do_forward(o, &num, peerflags, single, caller_entertained, &orig_answer_to,
1380 forced_clid, stored_clid);
1381
1382 if (o->chan) {
1383 ast_free(o->orig_chan_name);
1384 o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1385 if (single
1386 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1387 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1388 update_connected_line_from_peer(in, o->chan, 1);
1389 }
1390 }
1391 continue;
1392 }
1393 f = ast_read(winner);
1394 if (!f) {
1395 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1396 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1397 ast_hangup(c);
1398 c = o->chan = NULL;
1399 ast_clear_flag64(o, DIAL_STILLGOING);
1400 handle_cause(ast_channel_hangupcause(in), &num);
1401 continue;
1402 }
1403 switch (f->frametype) {
1404 case AST_FRAME_CONTROL:
1405 switch (f->subclass.integer) {
1406 case AST_CONTROL_ANSWER:
1407 /* This is our guy if someone answered. */
1408 if (!peer) {
1409 ast_trace(-1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1410 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1411 if (o->orig_chan_name
1412 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1413 /*
1414 * The channel name changed so we must generate COLP update.
1415 * Likely because a call pickup channel masqueraded in.
1416 */
1417 update_connected_line_from_peer(in, c, 1);
1418 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1419 if (o->pending_connected_update) {
1420 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1421 ast_channel_update_connected_line(in, &o->connected, NULL);
1422 }
1423 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1424 update_connected_line_from_peer(in, c, 1);
1425 }
1426 }
1427 if (o->aoc_s_rate_list) {
1428 size_t encoded_size;
1429 struct ast_aoc_encoded *encoded;
1430 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1431 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1432 ast_aoc_destroy_encoded(encoded);
1433 }
1434 }
1435 peer = c;
1436 /* Answer can optionally include a topology */
1437 if (f->subclass.topology) {
1438 /*
1439 * We need to bump the refcount on the topology to prevent it
1440 * from being cleaned up when the frame is cleaned up.
1441 */
1442 config->answer_topology = ao2_bump(f->subclass.topology);
1443 ast_trace(-1, "%s Found topology in frame: %p %p %s\n",
1444 ast_channel_name(peer), f, config->answer_topology,
1445 ast_str_tmp(256, ast_stream_topology_to_str(config->answer_topology, &STR_TMP)));
1446 }
1447
1448 /* Inform everyone else that they've been canceled.
1449 * The dial end event for the peer will be sent out after
1450 * other Dial options have been handled.
1451 */
1452 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1453 ast_copy_flags64(peerflags, o,
1454 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1455 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1456 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1457 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1458 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1459 DIAL_NOFORWARDHTML);
1460 ast_channel_dialcontext_set(c, "");
1461 ast_channel_exten_set(c, "");
1462 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1463 /* Setup early bridge if appropriate */
1464 ast_channel_early_bridge(in, peer);
1465 }
1466 }
1467 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1468 ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1469 ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1470 break;
1471 case AST_CONTROL_BUSY:
1472 ast_verb(3, "%s is busy\n", ast_channel_name(c));
1473 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1474 ast_channel_publish_dial(in, c, NULL, "BUSY");
1475 ast_hangup(c);
1476 c = o->chan = NULL;
1477 ast_clear_flag64(o, DIAL_STILLGOING);
1478 handle_cause(AST_CAUSE_BUSY, &num);
1479 break;
1480 case AST_CONTROL_CONGESTION:
1481 ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1482 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1483 ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1484 ast_hangup(c);
1485 c = o->chan = NULL;
1486 ast_clear_flag64(o, DIAL_STILLGOING);
1487 handle_cause(AST_CAUSE_CONGESTION, &num);
1488 break;
1489 case AST_CONTROL_RINGING:
1490 /* This is a tricky area to get right when using a native
1491 * CC agent. The reason is that we do the best we can to send only a
1492 * single ringing notification to the caller.
1493 *
1494 * Call completion complicates the logic used here. CCNR is typically
1495 * offered during a ringing message. Let's say that party A calls
1496 * parties B, C, and D. B and C do not support CC requests, but D
1497 * does. If we were to receive a ringing notification from B before
1498 * the others, then we would end up sending a ringing message to
1499 * A with no CCNR offer present.
1500 *
1501 * The approach that we have taken is that if we receive a ringing
1502 * response from a party and no CCNR offer is present, we need to
1503 * wait. Specifically, we need to wait until either a) a called party
1504 * offers CCNR in its ringing response or b) all called parties have
1505 * responded in some way to our call and none offers CCNR.
1506 *
1507 * The drawback to this is that if one of the parties has a delayed
1508 * response or, god forbid, one just plain doesn't respond to our
1509 * outgoing call, then this will result in a significant delay between
1510 * when the caller places the call and hears ringback.
1511 *
1512 * Note also that if CC is disabled for this call, then it is perfectly
1513 * fine for ringing frames to get sent through.
1514 */
1515 ++num_ringing;
1516 *to_progress = -1;
1517 orig_progress_to = -1;
1518 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1519 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1520 /* Setup early media if appropriate */
1521 if (single && !caller_entertained
1522 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1523 ast_channel_early_bridge(in, c);
1524 }
1525 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1526 ast_indicate(in, AST_CONTROL_RINGING);
1527 pa->sentringing++;
1528 }
1529 if (!sent_ring) {
1530 struct timeval now, then;
1531 int64_t diff;
1532
1533 now = ast_tvnow();
1534
1535 ast_channel_lock(in);
1536 ast_channel_stage_snapshot(in);
1537
1538 then = ast_channel_creationtime(c);
1539 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1540 set_duration_var(in, "RINGTIME", diff);
1541
1542 ast_channel_stage_snapshot_done(in);
1543 ast_channel_unlock(in);
1544 sent_ring = 1;
1545 }
1546 }
1547 ast_channel_publish_dial(in, c, NULL, "RINGING");
1548 break;
1549 case AST_CONTROL_PROGRESS:
1550 ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1551 /* Setup early media if appropriate */
1552 if (single && !caller_entertained
1553 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1554 ast_channel_early_bridge(in, c);
1555 }
1556 if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1557 if (single || (!single && !pa->sentringing)) {
1558 ast_indicate(in, AST_CONTROL_PROGRESS);
1559 }
1560 }
1561 *to_progress = -1;
1562 orig_progress_to = -1;
1563 if (!sent_progress) {
1564 struct timeval now, then;
1565 int64_t diff;
1566
1567 now = ast_tvnow();
1568
1569 ast_channel_lock(in);
1570 ast_channel_stage_snapshot(in);
1571
1572 then = ast_channel_creationtime(c);
1573 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1574 set_duration_var(in, "PROGRESSTIME", diff);
1575
1576 ast_channel_stage_snapshot_done(in);
1577 ast_channel_unlock(in);
1578 sent_progress = 1;
1579
1580 if (!ast_strlen_zero(mf_progress)) {
1581 ast_verb(3,
1582 "Sending MF '%s' to %s as result of "
1583 "receiving a PROGRESS message.\n",
1584 mf_progress, hearpulsing ? "parties" : "called party");
1585 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1586 (hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
1587 }
1588 if (!ast_strlen_zero(sf_progress)) {
1589 ast_verb(3,
1590 "Sending SF '%s' to %s as result of "
1591 "receiving a PROGRESS message.\n",
1592 sf_progress, (hearpulsing ? "parties" : "called party"));
1593 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1594 (hearpulsing ? in : NULL), sf_progress, 0, 0);
1595 }
1596 if (!ast_strlen_zero(dtmf_progress)) {
1597 ast_verb(3,
1598 "Sending DTMF '%s' to the called party as result of "
1599 "receiving a PROGRESS message.\n",
1600 dtmf_progress);
1601 res |= ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1602 }
1603 if (res) {
1604 ast_log(LOG_WARNING, "Called channel %s hung up post-progress before all digits could be sent\n", ast_channel_name(c));
1605 goto wait_over;
1606 }
1607 }
1608 ast_channel_publish_dial(in, c, NULL, "PROGRESS");
1609 break;
1610 case AST_CONTROL_WINK:
1611 ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1612 if (!sent_wink) {
1613 sent_wink = 1;
1614 if (!ast_strlen_zero(mf_wink)) {
1615 ast_verb(3,
1616 "Sending MF '%s' to %s as result of "
1617 "receiving a WINK message.\n",
1618 mf_wink, (hearpulsing ? "parties" : "called party"));
1619 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1620 (hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
1621 }
1622 if (!ast_strlen_zero(sf_wink)) {
1623 ast_verb(3,
1624 "Sending SF '%s' to %s as result of "
1625 "receiving a WINK message.\n",
1626 sf_wink, (hearpulsing ? "parties" : "called party"));
1627 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1628 (hearpulsing ? in : NULL), sf_wink, 0, 0);
1629 }
1630 if (res) {
1631 ast_log(LOG_WARNING, "Called channel %s hung up post-wink before all digits could be sent\n", ast_channel_name(c));
1632 goto wait_over;
1633 }
1634 }
1635 ast_indicate(in, AST_CONTROL_WINK);
1636 break;
1637 case AST_CONTROL_VIDUPDATE:
1638 case AST_CONTROL_SRCUPDATE:
1639 case AST_CONTROL_SRCCHANGE:
1640 if (!single || caller_entertained) {
1641 break;
1642 }
1643 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1644 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1645 ast_indicate(in, f->subclass.integer);
1646 break;
1647 case AST_CONTROL_CONNECTED_LINE:
1648 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1649 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1650 break;
1651 }
1652 if (!single) {
1653 struct ast_party_connected_line connected;
1654
1655 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1656 ast_channel_name(c), ast_channel_name(in));
1657 ast_party_connected_line_set_init(&connected, &o->connected);
1658 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1659 ast_party_connected_line_set(&o->connected, &connected, NULL);
1660 ast_party_connected_line_free(&connected);
1661 o->pending_connected_update = 1;
1662 break;
1663 }
1664 if (ast_channel_connected_line_sub(c, in, f, 1)) {
1665 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1666 }
1667 break;
1668 case AST_CONTROL_AOC:
1669 {
1670 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1671 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1672 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1673 o->aoc_s_rate_list = decoded;
1674 } else {
1675 ast_aoc_destroy_decoded(decoded);
1676 }
1677 }
1678 break;
1679 case AST_CONTROL_REDIRECTING:
1680 if (!single) {
1681 /*
1682 * Redirecting updates to the caller make sense only on single
1683 * calls.
1684 */
1685 break;
1686 }
1687 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1688 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1689 break;
1690 }
1691 ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1692 ast_channel_name(c), ast_channel_name(in));
1693 if (ast_channel_redirecting_sub(c, in, f, 1)) {
1694 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1695 }
1696 pa->sentringing = 0;
1697 break;
1698 case AST_CONTROL_PROCEEDING:
1699 ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1700 if (single && !caller_entertained
1701 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1702 ast_channel_early_bridge(in, c);
1703 }
1704 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1705 ast_indicate(in, AST_CONTROL_PROCEEDING);
1706 ast_channel_publish_dial(in, c, NULL, "PROCEEDING");
1707 break;
1708 case AST_CONTROL_HOLD:
1709 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1710 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1711 ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1712 break;
1713 case AST_CONTROL_UNHOLD:
1714 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1715 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1716 ast_indicate(in, AST_CONTROL_UNHOLD);
1717 break;
1718 case AST_CONTROL_OFFHOOK:
1719 case AST_CONTROL_FLASH:
1720 /* Ignore going off hook and flash */
1721 break;
1722 case AST_CONTROL_CC:
1723 if (!ignore_cc) {
1724 ast_handle_cc_control_frame(in, c, f->data.ptr);
1725 cc_frame_received = 1;
1726 }
1727 break;
1728 case AST_CONTROL_PVT_CAUSE_CODE:
1729 ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1730 break;
1731 case AST_CONTROL_PLAYBACK_BEGIN:
1732 if (!f->data.ptr) {
1733 ast_log(LOG_WARNING, "Got playback begin directive without filename on %s\n", ast_channel_name(c));
1734 } else {
1735 const char *filename = f->data.ptr;
1736 ast_verb(3, "Playing audio file %s on %s\n", filename, ast_channel_name(in));
1737 ast_streamfile(in, filename, ast_channel_language(in));
1738 }
1739 break;
1740 case -1:
1741 if (single && !caller_entertained) {
1742 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1743 ast_indicate(in, -1);
1744 pa->sentringing = 0;
1745 }
1746 break;
1747 default:
1748 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1749 break;
1750 }
1751 break;
1752 case AST_FRAME_VIDEO:
1753 case AST_FRAME_VOICE:
1754 case AST_FRAME_IMAGE:
1755 case AST_FRAME_DTMF_BEGIN:
1756 case AST_FRAME_DTMF_END:
1757 if (caller_entertained) {
1758 break;
1759 }
1760 *to_progress = -1;
1761 orig_progress_to = -1;
1762 /* Fall through */
1763 case AST_FRAME_TEXT:
1764 if (single && ast_write(in, f)) {
1765 ast_log(LOG_WARNING, "Unable to write frametype: %u\n",
1766 f->frametype);
1767 }
1768 break;
1769 case AST_FRAME_HTML:
1770 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1771 && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1772 ast_log(LOG_WARNING, "Unable to send URL\n");
1773 }
1774 break;
1775 default:
1776 break;
1777 }
1778 ast_frfree(f);
1779 } /* end for */
1780 if (winner == in) {
1781 struct ast_frame *f = ast_read(in);
1782#if 0
1783 if (f && (f->frametype != AST_FRAME_VOICE))
1784 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1785 else if (!f || (f->frametype != AST_FRAME_VOICE))
1786 printf("Hangup received on %s\n", in->name);
1787#endif
1788 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1789 /* Got hung up */
1790 *to_answer = -1;
1791 strcpy(pa->status, "CANCEL");
1792 pa->canceled = 1;
1793 publish_dial_end_event(in, out_chans, NULL, pa->status);
1794 if (f) {
1795 if (f->data.uint32) {
1796 ast_channel_hangupcause_set(in, f->data.uint32);
1797 }
1798 ast_frfree(f);
1799 }
1800 if (is_cc_recall) {
1801 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1802 }
1803 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller hung up\n", ast_channel_name(in));
1804 }
1805
1806 /* now f is guaranteed non-NULL */
1807 if (f->frametype == AST_FRAME_DTMF) {
1808 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1809 const char *context;
1810 ast_channel_lock(in);
1811 if ((context = pbx_builtin_getvar_helper(in, "EXITCONTEXT"))) {
1812 context = ast_strdupa(context);
1813 }
1814 ast_channel_unlock(in);
1815 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1816 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1817 *to_answer = 0;
1818 *result = f->subclass.integer;
1819 strcpy(pa->status, "CANCEL");
1820 pa->canceled = 1;
1821 publish_dial_end_event(in, out_chans, NULL, pa->status);
1822 ast_frfree(f);
1823 if (is_cc_recall) {
1824 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1825 }
1826 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller pressed %c to end call\n",
1827 ast_channel_name(in), f->subclass.integer);
1828 }
1829 }
1830
1831 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1832 detect_disconnect(in, f->subclass.integer, &featurecode)) {
1833 ast_verb(3, "User requested call disconnect.\n");
1834 *to_answer = 0;
1835 strcpy(pa->status, "CANCEL");
1836 pa->canceled = 1;
1837 publish_dial_end_event(in, out_chans, NULL, pa->status);
1838 ast_frfree(f);
1839 if (is_cc_recall) {
1840 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1841 }
1842 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller requested disconnect\n",
1843 ast_channel_name(in));
1844 }
1845 }
1846
1847 /* Send the frame from the in channel to all outgoing channels. */
1848 AST_LIST_TRAVERSE(out_chans, o, node) {
1849 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1850 /* This outgoing channel has died so don't send the frame to it. */
1851 continue;
1852 }
1853 switch (f->frametype) {
1854 case AST_FRAME_HTML:
1855 /* Forward HTML stuff */
1856 if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1857 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1858 ast_log(LOG_WARNING, "Unable to send URL\n");
1859 }
1860 break;
1861 case AST_FRAME_VIDEO:
1862 case AST_FRAME_VOICE:
1863 case AST_FRAME_IMAGE:
1864 if (!single || caller_entertained) {
1865 /*
1866 * We are calling multiple parties or caller is being
1867 * entertained and has thus not been made compatible.
1868 * No need to check any other called parties.
1869 */
1870 goto skip_frame;
1871 }
1872 /* Fall through */
1873 case AST_FRAME_TEXT:
1874 case AST_FRAME_DTMF_BEGIN:
1875 case AST_FRAME_DTMF_END:
1876 if (ast_write(o->chan, f)) {
1877 ast_log(LOG_WARNING, "Unable to forward frametype: %u\n",
1878 f->frametype);
1879 }
1880 break;
1881 case AST_FRAME_CONTROL:
1882 switch (f->subclass.integer) {
1883 case AST_CONTROL_HOLD:
1884 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1885 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1886 break;
1887 case AST_CONTROL_UNHOLD:
1888 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1889 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1890 break;
1891 case AST_CONTROL_FLASH:
1892 ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
1893 ast_indicate(o->chan, AST_CONTROL_FLASH);
1894 break;
1895 case AST_CONTROL_VIDUPDATE:
1896 case AST_CONTROL_SRCUPDATE:
1897 case AST_CONTROL_SRCCHANGE:
1898 if (!single || caller_entertained) {
1899 /*
1900 * We are calling multiple parties or caller is being
1901 * entertained and has thus not been made compatible.
1902 * No need to check any other called parties.
1903 */
1904 goto skip_frame;
1905 }
1906 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1907 ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1908 ast_indicate(o->chan, f->subclass.integer);
1909 break;
1910 case AST_CONTROL_CONNECTED_LINE:
1911 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1912 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
1913 break;
1914 }
1915 if (ast_channel_connected_line_sub(in, o->chan, f, 1)) {
1916 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1917 }
1918 break;
1919 case AST_CONTROL_REDIRECTING:
1920 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1921 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
1922 break;
1923 }
1924 if (ast_channel_redirecting_sub(in, o->chan, f, 1)) {
1925 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1926 }
1927 break;
1928 default:
1929 /* We are not going to do anything with this frame. */
1930 goto skip_frame;
1931 }
1932 break;
1933 default:
1934 /* We are not going to do anything with this frame. */
1935 goto skip_frame;
1936 }
1937 }
1938skip_frame:;
1939 ast_frfree(f);
1940 }
1941 }
1942
1943wait_over:
1944 if (!*to_answer || ast_check_hangup(in)) {
1945 ast_verb(3, "Nobody picked up in %d ms\n", orig_answer_to);
1946 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1947 } else if (!*to_progress) {
1948 ast_verb(3, "No early media received in %d ms\n", orig_progress_to);
1949 publish_dial_end_event(in, out_chans, NULL, "CHANUNAVAIL");
1950 strcpy(pa->status, "CHANUNAVAIL");
1951 *to_answer = 0; /* Reset to prevent hangup */
1952 }
1953
1954 if (is_cc_recall) {
1955 ast_cc_completed(in, "Recall completed!");
1956 }
1957 SCOPE_EXIT_RTN_VALUE(peer, "%s: %s%s\n", ast_channel_name(in),
1958 peer ? "Answered by " : "No answer", peer ? ast_channel_name(peer) : "");
1959}
1960
1961static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1962{
1963 char disconnect_code[AST_FEATURE_MAX_LEN];
1964 int res;
1965
1966 ast_str_append(featurecode, 1, "%c", code);
1967
1968 res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1969 if (res) {
1970 ast_str_reset(*featurecode);
1971 return 0;
1972 }
1973
1974 if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1975 /* Could be a partial match, anyway */
1976 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1977 ast_str_reset(*featurecode);
1978 }
1979 return 0;
1980 }
1981
1982 if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
1983 ast_str_reset(*featurecode);
1984 return 0;
1985 }
1986
1987 return 1;
1988}
1989
1990/* returns true if there is a valid privacy reply */
1991static int valid_priv_reply(struct ast_flags64 *opts, int res)
1992{
1993 if (res < '1')
1994 return 0;
1995 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1996 return 1;
1997 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1998 return 1;
1999 return 0;
2000}
2001
2002static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
2003 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
2004{
2005
2006 int res2;
2007 int loopcount = 0;
2008
2009 /* Get the user's intro, store it in priv-callerintros/$CID,
2010 unless it is already there-- this should be done before the
2011 call is actually dialed */
2012
2013 /* all ring indications and moh for the caller has been halted as soon as the
2014 target extension was picked up. We are going to have to kill some
2015 time and make the caller believe the peer hasn't picked up yet */
2016
2017 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2018 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2019 ast_indicate(chan, -1);
2020 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2021 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2022 ast_channel_musicclass_set(chan, original_moh);
2023 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2024 ast_indicate(chan, AST_CONTROL_RINGING);
2025 pa->sentringing++;
2026 }
2027
2028 /* Start autoservice on the other chan ?? */
2029 res2 = ast_autoservice_start(chan);
2030 /* Now Stream the File */
2031 for (loopcount = 0; loopcount < 3; loopcount++) {
2032 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
2033 break;
2034 if (!res2) /* on timeout, play the message again */
2035 res2 = ast_play_and_wait(peer, "priv-callpending");
2036 if (!valid_priv_reply(opts, res2))
2037 res2 = 0;
2038 /* priv-callpending script:
2039 "I have a caller waiting, who introduces themselves as:"
2040 */
2041 if (!res2)
2042 res2 = ast_play_and_wait(peer, pa->privintro);
2043 if (!valid_priv_reply(opts, res2))
2044 res2 = 0;
2045 /* now get input from the called party, as to their choice */
2046 if (!res2) {
2047 /* XXX can we have both, or they are mutually exclusive ? */
2048 if (ast_test_flag64(opts, OPT_PRIVACY))
2049 res2 = ast_play_and_wait(peer, "priv-callee-options");
2050 if (ast_test_flag64(opts, OPT_SCREENING))
2051 res2 = ast_play_and_wait(peer, "screen-callee-options");
2052 }
2053
2054 /*! \page DialPrivacy Dial Privacy scripts
2055 * \par priv-callee-options script:
2056 * \li Dial 1 if you wish this caller to reach you directly in the future,
2057 * and immediately connect to their incoming call.
2058 * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
2059 * \li Dial 3 to send this caller to the torture menus, now and forevermore.
2060 * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
2061 * \li Dial 5 to allow this caller to come straight thru to you in the future,
2062 * but right now, just this once, send them to voicemail.
2063 *
2064 * \par screen-callee-options script:
2065 * \li Dial 1 if you wish to immediately connect to the incoming call
2066 * \li Dial 2 if you wish to send this caller to voicemail.
2067 * \li Dial 3 to send this caller to the torture menus.
2068 * \li Dial 4 to send this caller to a simple "go away" menu.
2069 */
2070 if (valid_priv_reply(opts, res2))
2071 break;
2072 /* invalid option */
2073 res2 = ast_play_and_wait(peer, "vm-sorry");
2074 }
2075
2076 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
2077 ast_moh_stop(chan);
2078 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2079 ast_indicate(chan, -1);
2080 pa->sentringing = 0;
2081 }
2082 ast_autoservice_stop(chan);
2083 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
2084 /* map keypresses to various things, the index is res2 - '1' */
2085 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
2086 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
2087 int i = res2 - '1';
2088 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
2089 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
2090 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
2091 }
2092 switch (res2) {
2093 case '1':
2094 break;
2095 case '2':
2096 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2097 break;
2098 case '3':
2099 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2100 break;
2101 case '4':
2102 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2103 break;
2104 case '5':
2105 if (ast_test_flag64(opts, OPT_PRIVACY)) {
2106 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2107 break;
2108 }
2109 /* if not privacy, then 5 is the same as "default" case */
2110 default: /* bad input or -1 if failure to start autoservice */
2111 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
2112 /* well, there seems basically two choices. Just patch the caller thru immediately,
2113 or,... put 'em thru to voicemail. */
2114 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
2115 ast_verb(3, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
2116 /* XXX should we set status to DENY ? */
2117 /* XXX what about the privacy flags ? */
2118 break;
2119 }
2120
2121 if (res2 == '1') { /* the only case where we actually connect */
2122 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
2123 just clog things up, and it's not useful information, not being tied to a CID */
2124 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
2125 ast_filedelete(pa->privintro, NULL);
2126 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2127 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2128 else
2129 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2130 }
2131 return 0; /* the good exit path */
2132 } else {
2133 return -1;
2134 }
2135}
2136
2137/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
2138static int setup_privacy_args(struct privacy_args *pa,
2139 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
2140{
2141 char callerid[60];
2142 int res;
2143 char *l;
2144
2145 if (ast_channel_caller(chan)->id.number.valid
2146 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2147 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2148 ast_shrink_phone_number(l);
2149 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
2150 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
2151 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
2152 } else {
2153 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
2154 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2155 }
2156 } else {
2157 char *tnam, *tn2;
2158
2159 tnam = ast_strdupa(ast_channel_name(chan));
2160 /* clean the channel name so slashes don't try to end up in disk file name */
2161 for (tn2 = tnam; *tn2; tn2++) {
2162 if (*tn2 == '/') /* any other chars to be afraid of? */
2163 *tn2 = '=';
2164 }
2165 ast_verb(3, "Privacy-- callerid is empty\n");
2166
2167 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
2168 l = callerid;
2169 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2170 }
2171
2172 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
2173
2174 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
2175 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
2176 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
2177 pa->privdb_val = AST_PRIVACY_ALLOW;
2178 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
2179 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
2180 }
2181
2182 if (pa->privdb_val == AST_PRIVACY_DENY) {
2183 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
2184 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2185 return 0;
2186 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
2187 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2188 return 0; /* Is this right? */
2189 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
2190 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2191 return 0; /* is this right??? */
2192 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
2193 /* Get the user's intro, store it in priv-callerintros/$CID,
2194 unless it is already there-- this should be done before the
2195 call is actually dialed */
2196
2197 /* make sure the priv-callerintros dir actually exists */
2198 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
2199 if ((res = ast_mkdir(pa->privintro, 0755))) {
2200 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
2201 return -1;
2202 }
2203
2204 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
2205 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
2206 /* the DELUX version of this code would allow this caller the
2207 option to hear and retape their previously recorded intro.
2208 */
2209 } else {
2210 int duration; /* for feedback from play_and_wait */
2211 /* the file doesn't exist yet. Let the caller submit his
2212 vocal intro for posterity */
2213 /* priv-recordintro script:
2214 "At the tone, please say your name:"
2215 */
2216 int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
2217 ast_answer(chan);
2218 res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, NULL, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
2219 /* don't think we'll need a lock removed, we took care of
2220 conflicts by naming the pa.privintro file */
2221 if (res == -1) {
2222 /* Delete the file regardless since they hung up during recording */
2223 ast_filedelete(pa->privintro, NULL);
2224 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2225 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2226 else
2227 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2228 return -1;
2229 }
2230 if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
2231 ast_waitstream(chan, "");
2232 }
2233 }
2234 return 1; /* success */
2235}
2236
2237static void end_bridge_callback(void *data)
2238{
2239 struct ast_channel *chan = data;
2240
2241 ast_channel_lock(chan);
2242 ast_channel_stage_snapshot(chan);
2243 set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
2244 set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
2245 ast_channel_stage_snapshot_done(chan);
2246 ast_channel_unlock(chan);
2247}
2248
2249static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2250 bconfig->end_bridge_callback_data = originator;
2251}
2252
2253static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2254{
2255 struct ast_tone_zone_sound *ts = NULL;
2256 int res;
2257 const char *str = data;
2258
2259 if (ast_strlen_zero(str)) {
2260 ast_debug(1,"Nothing to play\n");
2261 return -1;
2262 }
2263
2264 ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2265
2266 if (ts && ts->data[0]) {
2267 res = ast_playtones_start(chan, 0, ts->data, 0);
2268 } else {
2269 res = -1;
2270 }
2271
2272 if (ts) {
2273 ts = ast_tone_zone_sound_unref(ts);
2274 }
2275
2276 if (res) {
2277 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2278 }
2279
2280 return res;
2281}
2282
2283/*!
2284 * \internal
2285 * \brief Setup the after bridge goto location on the peer.
2286 * \since 12.0.0
2287 *
2288 * \param chan Calling channel for bridge.
2289 * \param peer Peer channel for bridge.
2290 * \param opts Dialing option flags.
2291 * \param opt_args Dialing option argument strings.
2292 */
2293static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
2294{
2295 const char *context;
2296 const char *extension;
2297 int priority;
2298
2299 if (ast_test_flag64(opts, OPT_PEER_H)) {
2300 ast_channel_lock(chan);
2301 context = ast_strdupa(ast_channel_context(chan));
2302 ast_channel_unlock(chan);
2303 ast_bridge_set_after_h(peer, context);
2304 } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2305 ast_channel_lock(chan);
2306 context = ast_strdupa(ast_channel_context(chan));
2307 extension = ast_strdupa(ast_channel_exten(chan));
2308 priority = ast_channel_priority(chan);
2309 ast_channel_unlock(chan);
2310 ast_bridge_set_after_go_on(peer, context, extension, priority,
2311 opt_args[OPT_ARG_CALLEE_GO_ON]);
2312 }
2313}
2314
2315static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2316{
2317 int res = -1; /* default: error */
2318 char *rest, *cur; /* scan the list of destinations */
2319 struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2320 struct chanlist *outgoing;
2321 struct chanlist *tmp;
2322 struct ast_channel *peer = NULL;
2323 int to_answer, to_progress; /* timeouts */
2324 struct cause_args num = { chan, 0, 0, 0 };
2325 int cause, hanguptreecause = -1;
2326
2327 struct ast_bridge_config config = { { 0, } };
2328 struct timeval calldurationlimit = { 0, };
2329 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
2330 char *mf_progress = NULL, *mf_wink = NULL;
2331 char *sf_progress = NULL, *sf_wink = NULL;
2332 struct privacy_args pa = {
2333 .sentringing = 0,
2334 .privdb_val = 0,
2335 .status = "INVALIDARGS",
2336 .canceled = 0,
2337 };
2338 int sentringing = 0, moh = 0;
2339 const char *outbound_group = NULL;
2340 int result = 0;
2341 char *parse;
2342 int opermode = 0;
2343 int delprivintro = 0;
2344 AST_DECLARE_APP_ARGS(args,
2345 AST_APP_ARG(peers);
2346 AST_APP_ARG(timeout);
2347 AST_APP_ARG(options);
2348 AST_APP_ARG(url);
2349 );
2350 struct ast_flags64 opts = { 0, };
2351 char *opt_args[OPT_ARG_ARRAY_SIZE];
2352 int fulldial = 0, num_dialed = 0;
2353 int ignore_cc = 0;
2354 char device_name[AST_CHANNEL_NAME];
2355 char forced_clid_name[AST_MAX_EXTENSION];
2356 char stored_clid_name[AST_MAX_EXTENSION];
2357 int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2358 /*!
2359 * \brief Forced CallerID party information to send.
2360 * \note This will not have any malloced strings so do not free it.
2361 */
2362 struct ast_party_id forced_clid;
2363 /*!
2364 * \brief Stored CallerID information if needed.
2365 *
2366 * \note If OPT_ORIGINAL_CLID set then this is the o option
2367 * CallerID. Otherwise it is the dialplan extension and hint
2368 * name.
2369 *
2370 * \note This will not have any malloced strings so do not free it.
2371 */
2372 struct ast_party_id stored_clid;
2373 /*!
2374 * \brief CallerID party information to store.
2375 * \note This will not have any malloced strings so do not free it.
2376 */
2377 struct ast_party_caller caller;
2378 int max_forwards;
2379 struct ast_datastore *topology_ds = NULL;
2380 SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
2381
2382 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2383 ast_channel_lock(chan);
2384 ast_channel_stage_snapshot(chan);
2385 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2386 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2387 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2388 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2389 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME_MS", "");
2390 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2391 pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
2392 pbx_builtin_setvar_helper(chan, "RINGTIME", "");
2393 pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
2394 pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
2395 pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
2396 ast_channel_stage_snapshot_done(chan);
2397 max_forwards = ast_max_forwards_get(chan);
2398 ast_channel_unlock(chan);
2399
2400 if (max_forwards <= 0) {
2401 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2402 ast_channel_name(chan));
2403 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2404 SCOPE_EXIT_RTN_VALUE(-1, "%s: Max forwards exceeded\n", ast_channel_name(chan));
2405 }
2406
2407 if (ast_check_hangup_locked(chan)) {
2408 /*
2409 * Caller hung up before we could dial. If dial is executed
2410 * within an AGI then the AGI has likely eaten all queued
2411 * frames before executing the dial in DeadAGI mode. With
2412 * the caller hung up and no pending frames from the caller's
2413 * read queue, dial would not know that the call has hung up
2414 * until a called channel answers. It is rather annoying to
2415 * whoever just answered the non-existent call.
2416 *
2417 * Dial should not continue execution in DeadAGI mode, hangup
2418 * handlers, or the h exten.
2419 */
2420 ast_verb(3, "Caller hung up before dial.\n");
2421 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2422 SCOPE_EXIT_RTN_VALUE(-1, "%s: Caller hung up before dial\n", ast_channel_name(chan));
2423 }
2424
2425 parse = ast_strdupa(data ?: "");
2426
2427 AST_STANDARD_APP_ARGS(args, parse);
2428
2429 if (!ast_strlen_zero(args.options) &&
2430 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2431 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2432 goto done;
2433 }
2434
2435 if (ast_cc_call_init(chan, &ignore_cc)) {
2436 goto done;
2437 }
2438
2439 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2440 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2441
2442 if (delprivintro < 0 || delprivintro > 1) {
2443 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2444 delprivintro = 0;
2445 }
2446 }
2447
2448 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2449 opt_args[OPT_ARG_RINGBACK] = NULL;
2450 }
2451
2452 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2453 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2454 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2455 }
2456
2457 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2458 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2459 if (!calldurationlimit.tv_sec) {
2460 ast_log(LOG_WARNING, "Dial does not accept S(%s)\n", opt_args[OPT_ARG_DURATION_STOP]);
2461 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2462 goto done;
2463 }
2464 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2465 }
2466
2467 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2468 sf_wink = opt_args[OPT_ARG_SENDDTMF];
2469 dtmfcalled = strsep(&sf_wink, ":");
2470 dtmfcalling = strsep(&sf_wink, ":");
2471 dtmf_progress = strsep(&sf_wink, ":");
2472 mf_progress = strsep(&sf_wink, ":");
2473 mf_wink = strsep(&sf_wink, ":");
2474 sf_progress = strsep(&sf_wink, ":");
2475 }
2476
2477 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2478 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2479 goto done;
2480 }
2481
2482 /* Setup the forced CallerID information to send if used. */
2483 ast_party_id_init(&forced_clid);
2484 force_forwards_only = 0;
2485 if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2486 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2487 ast_channel_lock(chan);
2488 forced_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2489 ast_channel_unlock(chan);
2490 forced_clid_name[0] = '\0';
2491 forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2492 sizeof(forced_clid_name), chan);
2493 force_forwards_only = 1;
2494 } else {
2495 /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2496 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2497 &forced_clid.number.str);
2498 }
2499 if (!ast_strlen_zero(forced_clid.name.str)) {
2500 forced_clid.name.valid = 1;
2501 }
2502 if (!ast_strlen_zero(forced_clid.number.str)) {
2503 forced_clid.number.valid = 1;
2504 }
2505 }
2506 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2507 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2508 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2509 }
2510 forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2511 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2512 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2513 int pres;
2514
2515 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2516 if (0 <= pres) {
2517 forced_clid.number.presentation = pres;
2518 }
2519 }
2520
2521 /* Setup the stored CallerID information if needed. */
2522 ast_party_id_init(&stored_clid);
2523 if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2524 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2525 ast_channel_lock(chan);
2526 ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2527 if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2528 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2529 }
2530 if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2531 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2532 }
2533 if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2534 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2535 }
2536 if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2537 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2538 }
2539 ast_channel_unlock(chan);
2540 } else {
2541 /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2542 ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2543 &stored_clid.number.str);
2544 if (!ast_strlen_zero(stored_clid.name.str)) {
2545 stored_clid.name.valid = 1;
2546 }
2547 if (!ast_strlen_zero(stored_clid.number.str)) {
2548 stored_clid.number.valid = 1;
2549 }
2550 }
2551 } else {
2552 /*
2553 * In case the new channel has no preset CallerID number by the
2554 * channel driver, setup the dialplan extension and hint name.
2555 */
2556 stored_clid_name[0] = '\0';
2557 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2558 sizeof(stored_clid_name), chan);
2559 if (ast_strlen_zero(stored_clid.name.str)) {
2560 stored_clid.name.str = NULL;
2561 } else {
2562 stored_clid.name.valid = 1;
2563 }
2564 ast_channel_lock(chan);
2565 stored_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2566 stored_clid.number.valid = 1;
2567 ast_channel_unlock(chan);
2568 }
2569
2570 if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2571 ast_cdr_reset(ast_channel_name(chan), 0);
2572 }
2573 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2574 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2575
2576 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2577 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2578 if (res <= 0)
2579 goto out;
2580 res = -1; /* reset default */
2581 }
2582
2583 if (continue_exec)
2584 *continue_exec = 0;
2585
2586 /* If a channel group has been specified, get it for use when we create peer channels */
2587
2588 ast_channel_lock(chan);
2589 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2590 outbound_group = ast_strdupa(outbound_group);
2591 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2592 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2593 outbound_group = ast_strdupa(outbound_group);
2594 }
2595 ast_channel_unlock(chan);
2596
2597 /* Set per dial instance flags. These flags are also passed back to RetryDial. */
2598 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2599 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2600 | OPT_ANNOUNCE | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2601
2602 /* PREDIAL: Run gosub on the caller's channel */
2603 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2604 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2605 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2606 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2607 }
2608
2609 /* loop through the list of dial destinations */
2610 rest = args.peers;
2611 while ((cur = strsep(&rest, "&"))) {
2612 struct ast_channel *tc; /* channel for this destination */
2613 char *number;
2614 char *tech;
2615 int i;
2616 size_t tech_len;
2617 size_t number_len;
2618 struct ast_stream_topology *topology;
2619 struct ast_stream *stream;
2620
2621 cur = ast_strip(cur);
2622 if (ast_strlen_zero(cur)) {
2623 /* No tech/resource in this position. */
2624 continue;
2625 }
2626
2627 /* Get a technology/resource pair */
2628 number = cur;
2629 tech = strsep(&number, "/");
2630
2631 num_dialed++;
2632 if (ast_strlen_zero(number)) {
2633 ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2634 goto out;
2635 }
2636
2637 tech_len = strlen(tech) + 1;
2638 number_len = strlen(number) + 1;
2639 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2640 if (!tmp) {
2641 goto out;
2642 }
2643
2644 /* Save tech, number, and interface. */
2645 cur = tmp->stuff;
2646 strcpy(cur, tech);
2647 tmp->tech = cur;
2648 cur += tech_len;
2649 strcpy(cur, tech);
2650 cur[tech_len - 1] = '/';
2651 tmp->interface = cur;
2652 cur += tech_len;
2653 strcpy(cur, number);
2654 tmp->number = cur;
2655
2656 if (opts.flags) {
2657 /* Set per outgoing call leg options. */
2658 ast_copy_flags64(tmp, &opts,
2659 OPT_CANCEL_ELSEWHERE |
2660 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2661 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2662 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2663 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2664 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2665 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2666 OPT_RING_WITH_EARLY_MEDIA);
2667 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2668 }
2669
2670 /* Request the peer */
2671
2672 ast_channel_lock(chan);
2673 /*
2674 * Seed the chanlist's connected line information with previously
2675 * acquired connected line info from the incoming channel. The
2676 * previously acquired connected line info could have been set
2677 * through the CONNECTED_LINE dialplan function.
2678 */
2679 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2680
2681 if (ast_test_flag64(&opts, OPT_TOPOLOGY_PRESERVE)) {
2682 topology_ds = ast_channel_datastore_find(chan, &topology_ds_info, NULL);
2683
2684 if (!topology_ds && (topology_ds = ast_datastore_alloc(&topology_ds_info, NULL))) {
2685 topology_ds->data = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2686 ast_channel_datastore_add(chan, topology_ds);
2687 }
2688 }
2689
2690 if (topology_ds) {
2691 ao2_ref(topology_ds->data, +1);
2692 topology = topology_ds->data;
2693 } else {
2694 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2695 }
2696
2697 ast_channel_unlock(chan);
2698
2699 for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
2700 stream = ast_stream_topology_get_stream(topology, i);
2701 /* For both recvonly and sendonly the stream state reflects our state, that is we
2702 * are receiving only and we are sending only. Since we are requesting a
2703 * channel for the peer, we need to swap this to reflect what we will be doing.
2704 * That is, if we are receiving from Alice then we want to be sending to Bob,
2705 * so swap recvonly to sendonly and vice versa.
2706 */
2707 if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
2708 ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
2709 } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
2710 ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
2711 }
2712 }
2713
2714 tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
2715
2716 ast_stream_topology_free(topology);
2717
2718 if (!tc) {
2719 /* If we can't, just go on to the next call */
2720 /* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
2721 ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
2722 tmp->tech, cause, ast_cause2str(cause));
2723 handle_cause(cause, &num);
2724 if (!rest) {
2725 /* we are on the last destination */
2726 ast_channel_hangupcause_set(chan, cause);
2727 }
2728 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2729 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2730 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2731 }
2732 }
2733 chanlist_free(tmp);
2734 continue;
2735 }
2736
2737 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2738 if (!ignore_cc) {
2739 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2740 }
2741
2742 ast_channel_lock_both(tc, chan);
2743 ast_channel_stage_snapshot(tc);
2744
2745 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2746
2747 /* Setup outgoing SDP to match incoming one */
2748 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2749 /* We are on the only destination. */
2750 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2751 }
2752
2753 /* Inherit specially named variables from parent channel */
2754 ast_channel_inherit_variables(chan, tc);
2755 ast_channel_datastore_inherit(chan, tc);
2756 ast_max_forwards_decrement(tc);
2757
2758 ast_channel_appl_set(tc, "AppDial");
2759 ast_channel_data_set(tc, "(Outgoing Line)");
2760
2761 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2762
2763 /* Determine CallerID to store in outgoing channel. */
2764 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2765 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2766 caller.id = stored_clid;
2767 ast_channel_set_caller_event(tc, &caller, NULL);
2768 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2769 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2770 ast_channel_caller(tc)->id.number.str, NULL))) {
2771 /*
2772 * The new channel has no preset CallerID number by the channel
2773 * driver. Use the dialplan extension and hint name.
2774 */
2775 caller.id = stored_clid;
2776 if (!caller.id.name.valid
2777 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2778 ast_channel_connected(chan)->id.name.str, NULL))) {
2779 /*
2780 * No hint name available. We have a connected name supplied by
2781 * the dialplan we can use instead.
2782 */
2783 caller.id.name.valid = 1;
2784 caller.id.name = ast_channel_connected(chan)->id.name;
2785 }
2786 ast_channel_set_caller_event(tc, &caller, NULL);
2787 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2788 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2789 NULL))) {
2790 /* The new channel has no preset CallerID name by the channel driver. */
2791 if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2792 ast_channel_connected(chan)->id.name.str, NULL))) {
2793 /*
2794 * We have a connected name supplied by the dialplan we can
2795 * use instead.
2796 */
2797 caller.id.name.valid = 1;
2798 caller.id.name = ast_channel_connected(chan)->id.name;
2799 ast_channel_set_caller_event(tc, &caller, NULL);
2800 }
2801 }
2802
2803 /* Determine CallerID for outgoing channel to send. */
2804 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2805 struct ast_party_connected_line connected;
2806
2807 ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2808 connected.id = forced_clid;
2809 ast_channel_set_connected_line(tc, &connected, NULL);
2810 } else {
2811 ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2812 }
2813
2814 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2815
2816 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2817
2818 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2819 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2820 ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
2821 }
2822
2823 /* Pass ADSI CPE and transfer capability */
2824 ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
2825 ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
2826
2827 /* If we have an outbound group, set this peer channel to it */
2828 if (outbound_group)
2829 ast_app_group_set_channel(tc, outbound_group);
2830 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2831 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
2832 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2833
2834 /* Check if we're forced by configuration */
2835 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2836 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2837
2838
2839 /* Inherit context and extension */
2840 ast_channel_dialcontext_set(tc, ast_channel_context(chan));
2841 ast_channel_exten_set(tc, ast_channel_exten(chan));
2842
2843 ast_channel_stage_snapshot_done(tc);
2844
2845 /* Save the original channel name to detect call pickup masquerading in. */
2846 tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));
2847
2848 ast_channel_unlock(tc);
2849 ast_channel_unlock(chan);
2850
2851 /* Put channel in the list of outgoing thingies. */
2852 tmp->chan = tc;
2853 AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
2854 }
2855
2856 /* As long as we attempted to dial valid peers, don't throw a warning. */
2857 /* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
2858 if (!num_dialed) {
2859 ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
2860 if (continue_exec) {
2861 /* There is no point in having RetryDial try again */
2862 *continue_exec = 1;
2863 }
2864 strcpy(pa.status, "CHANUNAVAIL");
2865 res = 0;
2866 goto out;
2867 }
2868
2869 /*
2870 * PREDIAL: Run gosub on all of the callee channels
2871 *
2872 * We run the callee predial before ast_call() in case the user
2873 * wishes to do something on the newly created channels before
2874 * the channel does anything important.
2875 *
2876 * Inside the target gosub we will be able to do something with
2877 * the newly created channel name ie: now the calling channel
2878 * can know what channel will be used to call the destination
2879 * ex: now we will know that SIP/abc-123 is calling SIP/def-124
2880 */
2881 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
2882 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
2883 && !AST_LIST_EMPTY(&out_chans)) {
2884 const char *predial_callee;
2885
2886 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
2887 predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
2888 if (predial_callee) {
2889 ast_autoservice_start(chan);
2890 AST_LIST_TRAVERSE(&out_chans, tmp, node) {
2891 ast_pre_call(tmp->chan, predial_callee);
2892 }
2893 ast_autoservice_stop(chan);
2894 ast_free((char *) predial_callee);
2895 }
2896 }
2897
2898 /* Start all outgoing calls */
2899 AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
2900 res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
2901 ast_channel_lock(chan);
2902
2903 /* check the results of ast_call */
2904 if (res) {
2905 /* Again, keep going even if there's an error */
2906 ast_debug(1, "ast call on peer returned %d\n", res);
2907 ast_verb(3, "Couldn't call %s\n", tmp->interface);
2908 if (ast_channel_hangupcause(tmp->chan)) {
2909 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
2910 }
2911 ast_channel_unlock(chan);
2912 ast_cc_call_failed(chan, tmp->chan, tmp->interface);
2913 ast_hangup(tmp->chan);
2914 tmp->chan = NULL;
2915 AST_LIST_REMOVE_CURRENT(node);
2916 chanlist_free(tmp);
2917 continue;
2918 }
2919
2920 ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
2921 ast_channel_unlock(chan);
2922
2923 ast_verb(3, "Called %s\n", tmp->interface);
2924 ast_set_flag64(tmp, DIAL_STILLGOING);
2925
2926 /* If this line is up, don't try anybody else */
2927 if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
2928 break;
2929 }
2930 }
2931 AST_LIST_TRAVERSE_SAFE_END;
2932
2933 if (ast_strlen_zero(args.timeout)) {
2934 to_answer = -1;
2935 to_progress = -1;
2936 } else {
2937 char *anstimeout = strsep(&args.timeout, "^");
2938 if (!ast_strlen_zero(anstimeout)) {
2939 to_answer = atoi(anstimeout);
2940 if (to_answer > 0) {
2941 to_answer *= 1000;
2942 } else {
2943 ast_log(LOG_WARNING, "Invalid answer timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2944 to_answer = -1;
2945 }
2946 } else {
2947 to_answer = -1;
2948 }
2949 if (!ast_strlen_zero(args.timeout)) {
2950 to_progress = atoi(args.timeout);
2951 if (to_progress > 0) {
2952 to_progress *= 1000;
2953 } else {
2954 ast_log(LOG_WARNING, "Invalid progress timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2955 to_progress = -1;
2956 }
2957 } else {
2958 to_progress = -1;
2959 }
2960 }
2961
2962 outgoing = AST_LIST_FIRST(&out_chans);
2963 if (!outgoing) {
2964 strcpy(pa.status, "CHANUNAVAIL");
2965 if (fulldial == num_dialed) {
2966 res = -1;
2967 goto out;
2968 }
2969 } else {
2970 /* Our status will at least be NOANSWER */
2971 strcpy(pa.status, "NOANSWER");
2972 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2973 moh = 1;
2974 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2975 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2976 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2977 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2978 ast_channel_musicclass_set(chan, original_moh);
2979 } else {
2980 ast_moh_start(chan, NULL, NULL);
2981 }
2982 ast_indicate(chan, AST_CONTROL_PROGRESS);
2983 } else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
2984 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
2985 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
2986 ast_indicate(chan, AST_CONTROL_RINGING);
2987 sentringing++;
2988 } else {
2989 ast_indicate(chan, AST_CONTROL_PROGRESS);
2990 }
2991 } else {
2992 ast_indicate(chan, AST_CONTROL_RINGING);
2993 sentringing++;
2994 }
2995 }
2996 }
2997
2998 peer = wait_for_answer(chan, &out_chans, &to_answer, &to_progress, peerflags, opt_args, &pa, &num, &result,
2999 dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
3000 (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
3001 ignore_cc, &forced_clid, &stored_clid, &config);
3002
3003 if (!peer) {
3004 if (result) {
3005 res = result;
3006 } else if (to_answer) { /* Musta gotten hung up */
3007 res = -1;
3008 } else { /* Nobody answered, next please? */
3009 res = 0;
3010 }
3011 } else {
3012 const char *number;
3013 const char *name;
3014 int dial_end_raised = 0;
3015 int cause = -1;
3016
3017 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
3018 ast_answer(chan);
3019 }
3020
3021 /* Ah ha! Someone answered within the desired timeframe. Of course after this
3022 we will always return with -1 so that it is hung up properly after the
3023 conversation. */
3024
3025 if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
3026 && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
3027 cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
3028 if (cause <= 0) {
3029 if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
3030 cause = 0;
3031 } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
3032 || cause < 0) {
3033 ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
3034 opt_args[OPT_ARG_HANGUPCAUSE]);
3035 cause = -1;
3036 }
3037 }
3038 }
3039 hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
3040
3041 /* If appropriate, log that we have a destination channel and set the answer time */
3042
3043 ast_channel_lock(peer);
3044 name = ast_strdupa(ast_channel_name(peer));
3045
3046 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
3047 if (ast_strlen_zero(number)) {
3048 number = NULL;
3049 } else {
3050 number = ast_strdupa(number);
3051 }
3052 ast_channel_unlock(peer);
3053
3054 ast_channel_lock(chan);
3055 ast_channel_stage_snapshot(chan);
3056
3057 strcpy(pa.status, "ANSWER");
3058 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3059
3060 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
3061 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
3062
3063 ast_channel_stage_snapshot_done(chan);
3064 ast_channel_unlock(chan);
3065
3066 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
3067 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
3068 ast_channel_sendurl( peer, args.url );
3069 }
3070 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
3071 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
3072 ast_channel_publish_dial(chan, peer, NULL, pa.status);
3073 /* hang up on the callee -- he didn't want to talk anyway! */
3074 ast_autoservice_chan_hangup_peer(chan, peer);
3075 res = 0;
3076 goto out;
3077 }
3078 }
3079 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
3080 res = 0;
3081 } else {
3082 int digit = 0;
3083 struct ast_channel *chans[2];
3084 struct ast_channel *active_chan;
3085 char *calledfile = NULL, *callerfile = NULL;
3086 int calledstream = 0, callerstream = 0;
3087
3088 chans[0] = chan;
3089 chans[1] = peer;
3090
3091 /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
3092 callerfile = opt_args[OPT_ARG_ANNOUNCE];
3093 calledfile = strsep(&callerfile, ":");
3094
3095 /* stream the file(s) */
3096 if (!ast_strlen_zero(calledfile)) {
3097 res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
3098 if (res) {
3099 res = 0;
3100 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
3101 } else {
3102 calledstream = 1;
3103 }
3104 }
3105 if (!ast_strlen_zero(callerfile)) {
3106 res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
3107 if (res) {
3108 res = 0;
3109 ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
3110 } else {
3111 callerstream = 1;
3112 }
3113 }
3114
3115 /* can't use ast_waitstream, because we're streaming two files at once, and can't block
3116 We'll need to handle both channels at once. */
3117
3118 ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
3119 while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
3120 int mspeer, mschan;
3121
3122 mspeer = ast_sched_wait(ast_channel_sched(peer));
3123 mschan = ast_sched_wait(ast_channel_sched(chan));
3124
3125 if (calledstream) {
3126 if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
3127 ast_stopstream(peer);
3128 calledstream = 0;
3129 }
3130 }
3131 if (callerstream) {
3132 if (mschan < 0 && !ast_channel_timingfunc(chan)) {
3133 ast_stopstream(chan);
3134 callerstream = 0;
3135 }
3136 }
3137
3138 if (!calledstream && !callerstream) {
3139 break;
3140 }
3141
3142 if (mspeer < 0)
3143 mspeer = 1000;
3144
3145 if (mschan < 0)
3146 mschan = 1000;
3147
3148 /* wait for the lowest maximum of the two */
3149 active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
3150 if (active_chan) {
3151 struct ast_channel *other_chan;
3152 struct ast_frame *fr = ast_read(active_chan);
3153
3154 if (!fr) {
3155 ast_autoservice_chan_hangup_peer(chan, peer);
3156 res = -1;
3157 goto done;
3158 }
3159 switch (fr->frametype) {
3160 case AST_FRAME_DTMF_END:
3161 digit = fr->subclass.integer;
3162 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
3163 ast_stopstream(peer);
3164 res = ast_senddigit(chan, digit, 0);
3165 }
3166 break;
3167 case AST_FRAME_CONTROL:
3168 switch (fr->subclass.integer) {
3169 case AST_CONTROL_HANGUP:
3170 ast_frfree(fr);
3171 ast_autoservice_chan_hangup_peer(chan, peer);
3172 res = -1;
3173 goto done;
3174 case AST_CONTROL_CONNECTED_LINE:
3175 /* Pass COLP update to the other channel. */
3176 if (active_chan == chan) {
3177 other_chan = peer;
3178 } else {
3179 other_chan = chan;
3180 }
3181 if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)) {
3182 ast_indicate_data(other_chan, fr->subclass.integer,
3183 fr->data.ptr, fr->datalen);
3184 }
3185 break;
3186 default:
3187 break;
3188 }
3189 break;
3190 default:
3191 /* Ignore all others */
3192 break;
3193 }
3194 ast_frfree(fr);
3195 }
3196 ast_sched_runq(ast_channel_sched(peer));
3197 ast_sched_runq(ast_channel_sched(chan));
3198 }
3199 ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
3200 }
3201
3202 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
3203 /* chan and peer are going into the PBX; as such neither are considered
3204 * outgoing channels any longer */
3205 ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);
3206
3207 ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
3208 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
3209 /* peer goes to the same context and extension as chan, so just copy info from chan*/
3210 ast_channel_lock(peer);
3211 ast_channel_stage_snapshot(peer);
3212 ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
3213 ast_channel_context_set(peer, ast_channel_context(chan));
3214 ast_channel_exten_set(peer, ast_channel_exten(chan));
3215 ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
3216 ast_channel_stage_snapshot_done(peer);
3217 ast_channel_unlock(peer);
3218 if (ast_pbx_start(peer)) {
3219 ast_autoservice_chan_hangup_peer(chan, peer);
3220 }
3221 if (continue_exec)
3222 *continue_exec = 1;
3223 res = 0;
3224 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3225 goto done;
3226 }
3227
3228 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
3229 const char *gosub_result_peer;
3230 char *gosub_argstart;
3231 char *gosub_args = NULL;
3232 int gosub_res = -1;
3233
3234 ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
3235 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
3236 if (gosub_argstart) {
3237 const char *what_is_s = "s";
3238 *gosub_argstart = 0;
3239 if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
3240 ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
3241 what_is_s = "~~s~~";
3242 }
3243 if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
3244 gosub_args = NULL;
3245 }
3246 *gosub_argstart = ',';
3247 } else {
3248 const char *what_is_s = "s";
3249 if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
3250 ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
3251 what_is_s = "~~s~~";
3252 }
3253 if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
3254 gosub_args = NULL;
3255 }
3256 }
3257 if (gosub_args) {
3258 gosub_res = ast_app_exec_sub(chan, peer, gosub_args, 0);
3259 ast_free(gosub_args);
3260 } else {
3261 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
3262 }
3263
3264 ast_channel_lock_both(chan, peer);
3265
3266 if (!gosub_res && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
3267 char *gosub_transfer_dest;
3268 char *gosub_result = ast_strdupa(gosub_result_peer);
3269 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
3270
3271 /* Inherit return value from the peer, so it can be used in the master */
3272 if (gosub_retval) {
3273 pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
3274 }
3275
3276 ast_channel_unlock(peer);
3277 ast_channel_unlock(chan);
3278
3279 if (!strcasecmp(gosub_result, "BUSY")) {
3280 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3281 ast_set_flag64(peerflags, OPT_GO_ON);
3282 gosub_res = -1;
3283 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
3284 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3285 ast_set_flag64(peerflags, OPT_GO_ON);
3286 gosub_res = -1;
3287 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
3288 /* Hangup peer and continue with the next extension priority. */
3289 ast_set_flag64(peerflags, OPT_GO_ON);
3290 gosub_res = -1;
3291 } else if (!strcasecmp(gosub_result, "ABORT")) {
3292 /* Hangup both ends unless the caller has the g flag */
3293 gosub_res = -1;
3294 } else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
3295 gosub_transfer_dest = gosub_result + 5;
3296 gosub_res = -1;
3297 /* perform a transfer to a new extension */
3298 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
3299 ast_replace_subargument_delimiter(gosub_transfer_dest);
3300 }
3301 if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
3302 ast_set_flag64(peerflags, OPT_GO_ON);
3303 }
3304 }
3305 if (gosub_res) {
3306 res = gosub_res;
3307 if (!dial_end_raised) {
3308 ast_channel_publish_dial(chan, peer, NULL, gosub_result);
3309 dial_end_raised = 1;
3310 }
3311 }
3312 } else {
3313 ast_channel_unlock(peer);
3314 ast_channel_unlock(chan);
3315 }
3316 }
3317
3318 if (!res) {
3319
3320 /* None of the Dial options changed our status; inform
3321 * everyone that this channel answered
3322 */
3323 if (!dial_end_raised) {
3324 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3325 dial_end_raised = 1;
3326 }
3327
3328 if (!ast_tvzero(calldurationlimit)) {
3329 struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
3330 ast_channel_lock(peer);
3331 ast_channel_whentohangup_set(peer, &whentohangup);
3332 ast_channel_unlock(peer);
3333 }
3334 if (!ast_strlen_zero(dtmfcalled)) {
3335 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
3336 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
3337 }
3338 if (!ast_strlen_zero(dtmfcalling)) {
3339 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
3340 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
3341 }
3342 }
3343
3344 if (res) { /* some error */
3345 if (!ast_check_hangup(chan) && ast_check_hangup(peer)) {
3346 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
3347 }
3348 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3349 if (ast_bridge_setup_after_goto(peer)
3350 || ast_pbx_start(peer)) {
3351 ast_autoservice_chan_hangup_peer(chan, peer);
3352 }
3353 res = -1;
3354 } else {
3355 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
3356 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
3357 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
3358 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
3359 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
3360 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
3361 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
3362 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
3363 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
3364 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
3365 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
3366 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
3367 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
3368 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
3369 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
3370 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
3371 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
3372 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
3373 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
3374 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
3375
3376 config.end_bridge_callback = end_bridge_callback;
3377 config.end_bridge_callback_data = chan;
3378 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
3379
3380 if (moh) {
3381 moh = 0;
3382 ast_moh_stop(chan);
3383 } else if (sentringing) {
3384 sentringing = 0;
3385 ast_indicate(chan, -1);
3386 }
3387 /* Be sure no generators are left on it and reset the visible indication */
3388 ast_deactivate_generator(chan);
3389 ast_channel_visible_indication_set(chan, 0);
3390 /* Make sure channels are compatible */
3391 res = ast_channel_make_compatible(chan, peer);
3392 if (res < 0) {
3393 ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", ast_channel_name(chan), ast_channel_name(peer));
3394 ast_autoservice_chan_hangup_peer(chan, peer);
3395 res = -1;
3396 goto done;
3397 }
3398 if (opermode) {
3399 struct oprmode oprmode;
3400
3401 oprmode.peer = peer;
3402 oprmode.mode = opermode;
3403
3404 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
3405 }
3406 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3407
3408 res = ast_bridge_call(chan, peer, &config);
3409 }
3410 }
3411out:
3412 if (moh) {
3413 moh = 0;
3414 ast_moh_stop(chan);
3415 } else if (sentringing) {
3416 sentringing = 0;
3417 ast_indicate(chan, -1);
3418 }
3419
3420 if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3421 ast_filedelete(pa.privintro, NULL);
3422 if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3423 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
3424 } else {
3425 ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
3426 }
3427 }
3428
3429 ast_channel_early_bridge(chan, NULL);
3430 /* forward 'answered elsewhere' if we received it */
3431 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
3432 hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
3433 } else if (pa.canceled) { /* Caller canceled */
3434 if (ast_channel_hangupcause(chan))
3435 hanguptreecause = ast_channel_hangupcause(chan);
3436 else
3437 hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
3438 }
3439 hanguptree(&out_chans, NULL, hanguptreecause);
3440 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3441 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
3442
3443 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
3444 if (!ast_tvzero(calldurationlimit))
3445 memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
3446 res = 0;
3447 }
3448
3449done:
3450 if (config.answer_topology) {
3451 ast_trace(2, "%s Cleaning up topology: %p %s\n",
3452 peer ? ast_channel_name(peer) : "<no channel>", &config.answer_topology,
3453 ast_str_tmp(256, ast_stream_topology_to_str(config.answer_topology, &STR_TMP)));
3454
3455 /*
3456 * At this point, the channel driver that answered should have bumped the
3457 * topology refcount for itself. Here we're cleaning up the reference we added
3458 * in wait_for_answer().
3459 */
3460 ast_stream_topology_free(config.answer_topology);
3461 }
3462 if (config.warning_sound) {
3463 ast_free((char *)config.warning_sound);
3464 }
3465 if (config.end_sound) {
3466 ast_free((char *)config.end_sound);
3467 }
3468 if (config.start_sound) {
3469 ast_free((char *)config.start_sound);
3470 }
3471 ast_ignore_cc(chan);
3472 SCOPE_EXIT_RTN_VALUE(res, "%s: Done\n", ast_channel_name(chan));
3473}
3474
3475static int dial_exec(struct ast_channel *chan, const char *data)
3476{
3477 struct ast_flags64 peerflags;
3478
3479 memset(&peerflags, 0, sizeof(peerflags));
3480
3481 return dial_exec_full(chan, data, &peerflags, NULL);
3482}
3483
3484static int retrydial_exec(struct ast_channel *chan, const char *data)
3485{
3486 char *parse;
3487 const char *context = NULL;
3488 int sleepms = 0, loops = 0, res = -1;
3489 struct ast_flags64 peerflags = { 0, };
3490 AST_DECLARE_APP_ARGS(args,
3491 AST_APP_ARG(announce);
3492 AST_APP_ARG(sleep);
3493 AST_APP_ARG(retries);
3494 AST_APP_ARG(dialdata);
3495 );
3496
3497 if (ast_strlen_zero(data)) {
3498 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
3499 return -1;
3500 }
3501
3502 parse = ast_strdupa(data);
3503 AST_STANDARD_APP_ARGS(args, parse);
3504
3505 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
3506 sleepms *= 1000;
3507
3508 if (!ast_strlen_zero(args.retries)) {
3509 loops = atoi(args.retries);
3510 }
3511
3512 if (!args.dialdata) {
3513 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
3514 goto done;
3515 }
3516
3517 if (sleepms < 1000)
3518 sleepms = 10000;
3519
3520 if (!loops)
3521 loops = -1; /* run forever */
3522
3523 ast_channel_lock(chan);
3524 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
3525 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
3526 ast_channel_unlock(chan);
3527
3528 res = 0;
3529 while (loops) {
3530 int continue_exec;
3531
3532 ast_channel_data_set(chan, "Retrying");
3533 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3534 ast_moh_stop(chan);
3535
3536 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
3537 if (continue_exec)
3538 break;
3539
3540 if (res == 0) {
3541 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
3542 if (!ast_strlen_zero(args.announce)) {
3543 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3544 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3545 ast_waitstream(chan, AST_DIGIT_ANY);
3546 } else
3547 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3548 }
3549 if (!res && sleepms) {
3550 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3551 ast_moh_start(chan, NULL, NULL);
3552 res = ast_waitfordigit(chan, sleepms);
3553 }
3554 } else {
3555 if (!ast_strlen_zero(args.announce)) {
3556 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3557 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3558 res = ast_waitstream(chan, "");
3559 } else
3560 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3561 }
3562 if (sleepms) {
3563 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3564 ast_moh_start(chan, NULL, NULL);
3565 if (!res)
3566 res = ast_waitfordigit(chan, sleepms);
3567 }
3568 }
3569 }
3570
3571 if (res < 0 || res == AST_PBX_INCOMPLETE) {
3572 break;
3573 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
3574 if (onedigit_goto(chan, context, (char) res, 1)) {
3575 res = 0;
3576 break;
3577 }
3578 }
3579 loops--;
3580 }
3581 if (loops == 0)
3582 res = 0;
3583 else if (res == 1)
3584 res = 0;
3585
3586 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3587 ast_moh_stop(chan);
3588 done:
3589 return res;
3590}
3591
3592static int unload_module(void)
3593{
3594 int res;
3595
3596 res = ast_unregister_application(app);
3597 res |= ast_unregister_application(rapp);
3598
3599 return res;
3600}
3601
3602static int load_module(void)
3603{
3604 int res;
3605
3606 res = ast_register_application_xml(app, dial_exec);
3607 res |= ast_register_application_xml(rapp, retrydial_exec);
3608
3609 return res;
3610}
3611
3612AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application",
3613 .support_level = AST_MODULE_SUPPORT_CORE,
3614 .load = load_module,
3615 .unload = unload_module,
3616 .requires = "ccss",
3617);
aoc.h
Generic Advice of Charge encode and decode routines.
ast_aoc_destroy_encoded
void * ast_aoc_destroy_encoded(struct ast_aoc_encoded *encoded)
free an ast_aoc_encoded object
Definition: aoc.c:322
ast_aoc_get_msg_type
enum ast_aoc_type ast_aoc_get_msg_type(struct ast_aoc_decoded *decoded)
get the message type, AOC-D, AOC-E, or AOC Request
Definition: aoc.c:901
ast_aoc_decode
struct ast_aoc_decoded * ast_aoc_decode(struct ast_aoc_encoded *encoded, size_t size, struct ast_channel *chan)
decodes an encoded aoc payload.
Definition: aoc.c:458
ast_aoc_destroy_decoded
void * ast_aoc_destroy_decoded(struct ast_aoc_decoded *decoded)
free an ast_aoc_decoded object
Definition: aoc.c:316
ast_aoc_encode
struct ast_aoc_encoded * ast_aoc_encode(struct ast_aoc_decoded *decoded, size_t *out_size, struct ast_channel *chan)
encodes a decoded aoc structure so it can be passed on the wire
Definition: aoc.c:659
AST_AOC_S
@ AST_AOC_S
Definition: aoc.h:64
digit
char digit
Definition: app_alarmreceiver.c:131
topology_ds_destroy
static void topology_ds_destroy(void *data)
Definition: app_dial.c:826
DIAL_STILLGOING
#define DIAL_STILLGOING
Definition: app_dial.c:706
dial_exec_full
static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
Definition: app_dial.c:2315
OPT_PREDIAL_CALLER
#define OPT_PREDIAL_CALLER
Definition: app_dial.c:717
OPT_RESETCDR
@ OPT_RESETCDR
Definition: app_dial.c:674
OPT_SCREEN_NOINTRO
@ OPT_SCREEN_NOINTRO
Definition: app_dial.c:684
OPT_DTMF_EXIT
@ OPT_DTMF_EXIT
Definition: app_dial.c:675
OPT_ANNOUNCE
@ OPT_ANNOUNCE
Definition: app_dial.c:673
OPT_CALLEE_PARK
@ OPT_CALLEE_PARK
Definition: app_dial.c:697
OPT_DURATION_LIMIT
@ OPT_DURATION_LIMIT
Definition: app_dial.c:682
OPT_SCREEN_NOCALLERID
@ OPT_SCREEN_NOCALLERID
Definition: app_dial.c:685
OPT_IGNORE_FORWARDING
@ OPT_IGNORE_FORWARDING
Definition: app_dial.c:699
OPT_OPERMODE
@ OPT_OPERMODE
Definition: app_dial.c:696
OPT_DURATION_STOP
@ OPT_DURATION_STOP
Definition: app_dial.c:690
OPT_GO_ON
@ OPT_GO_ON
Definition: app_dial.c:678
OPT_RINGBACK
@ OPT_RINGBACK
Definition: app_dial.c:689
OPT_GOTO
@ OPT_GOTO
Definition: app_dial.c:695
OPT_IGNORE_CONNECTEDLINE
@ OPT_IGNORE_CONNECTEDLINE
Definition: app_dial.c:686
OPT_CALLEE_TRANSFER
@ OPT_CALLEE_TRANSFER
Definition: app_dial.c:691
OPT_SENDDTMF
@ OPT_SENDDTMF
Definition: app_dial.c:676
OPT_CALLER_MIXMONITOR
@ OPT_CALLER_MIXMONITOR
Definition: app_dial.c:702
OPT_CALLER_PARK
@ OPT_CALLER_PARK
Definition: app_dial.c:698
OPT_CALLER_MONITOR
@ OPT_CALLER_MONITOR
Definition: app_dial.c:694
OPT_CALLEE_MONITOR
@ OPT_CALLEE_MONITOR
Definition: app_dial.c:693
OPT_CALLEE_GOSUB
@ OPT_CALLEE_GOSUB
Definition: app_dial.c:700
OPT_CALLER_HANGUP
@ OPT_CALLER_HANGUP
Definition: app_dial.c:680
OPT_FORCECLID
@ OPT_FORCECLID
Definition: app_dial.c:677
OPT_CALLEE_HANGUP
@ OPT_CALLEE_HANGUP
Definition: app_dial.c:679
OPT_SCREENING
@ OPT_SCREENING
Definition: app_dial.c:687
OPT_MUSICBACK
@ OPT_MUSICBACK
Definition: app_dial.c:683
OPT_CALLER_TRANSFER
@ OPT_CALLER_TRANSFER
Definition: app_dial.c:692
OPT_CALLEE_MIXMONITOR
@ OPT_CALLEE_MIXMONITOR
Definition: app_dial.c:701
OPT_ORIGINAL_CLID
@ OPT_ORIGINAL_CLID
Definition: app_dial.c:681
OPT_PRIVACY
@ OPT_PRIVACY
Definition: app_dial.c:688
OPT_CANCEL_ELSEWHERE
#define OPT_CANCEL_ELSEWHERE
Definition: app_dial.c:709
get_cid_name
static const char * get_cid_name(char *name, int namelen, struct ast_channel *chan)
Definition: app_dial.c:913
app
static const char app[]
Definition: app_dial.c:669
dial_exec_options
static const struct ast_app_option dial_exec_options[128]
Definition: app_dial.c:791
OPT_PEER_H
#define OPT_PEER_H
Definition: app_dial.c:710
do_forward
static void do_forward(struct chanlist *o, struct cause_args *num, struct ast_flags64 *peerflags, int single, int caller_entertained, int *to, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
Definition: app_dial.c:943
OPT_PREDIAL_CALLEE
#define OPT_PREDIAL_CALLEE
Definition: app_dial.c:716
DIAL_CALLERID_ABSENT
#define DIAL_CALLERID_ABSENT
Definition: app_dial.c:708
OPT_FORCE_CID_PRES
#define OPT_FORCE_CID_PRES
Definition: app_dial.c:714
setup_peer_after_bridge_goto
static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
Definition: app_dial.c:2293
CAN_EARLY_BRIDGE
#define CAN_EARLY_BRIDGE(flags, chan, peer)
Definition: app_dial.c:793
OPT_TOPOLOGY_PRESERVE
#define OPT_TOPOLOGY_PRESERVE
Definition: app_dial.c:721
OPT_RING_WITH_EARLY_MEDIA
#define OPT_RING_WITH_EARLY_MEDIA
Definition: app_dial.c:718
OPT_FORCE_CID_TAG
#define OPT_FORCE_CID_TAG
Definition: app_dial.c:713
OPT_HEARPULSING
#define OPT_HEARPULSING
Definition: app_dial.c:720
dial_exec
static int dial_exec(struct ast_channel *chan, const char *data)
Definition: app_dial.c:3475
DIAL_NOFORWARDHTML
#define DIAL_NOFORWARDHTML
Definition: app_dial.c:707
AST_MAX_WATCHERS
#define AST_MAX_WATCHERS
Definition: app_dial.c:864
OPT_CANCEL_TIMEOUT
#define OPT_CANCEL_TIMEOUT
Definition: app_dial.c:712
chanlist_free
static void chanlist_free(struct chanlist *outgoing)
Definition: app_dial.c:838
publish_dial_end_event
static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
Definition: app_dial.c:1155
onedigit_goto
static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
Definition: app_dial.c:898
rapp
static const char rapp[]
Definition: app_dial.c:670
handle_cause
static void handle_cause(int cause, struct cause_args *num)
Definition: app_dial.c:876
setup_privacy_args
static int setup_privacy_args(struct privacy_args *pa, struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
returns 1 if successful, 0 or <0 if the caller should 'goto out'
Definition: app_dial.c:2138
set_duration_var
static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
Definition: app_dial.c:1195
update_connected_line_from_peer
static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
Definition: app_dial.c:1175
detect_disconnect
static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
Definition: app_dial.c:1961
OPT_HANGUPCAUSE
#define OPT_HANGUPCAUSE
Definition: app_dial.c:719
hanguptree
static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
Definition: app_dial.c:846
OPT_ARG_CALLEE_GO_ON
@ OPT_ARG_CALLEE_GO_ON
Definition: app_dial.c:731
OPT_ARG_SENDDTMF
@ OPT_ARG_SENDDTMF
Definition: app_dial.c:725
OPT_ARG_DURATION_STOP
@ OPT_ARG_DURATION_STOP
Definition: app_dial.c:733
OPT_ARG_PREDIAL_CALLEE
@ OPT_ARG_PREDIAL_CALLEE
Definition: app_dial.c:740
OPT_ARG_RINGBACK
@ OPT_ARG_RINGBACK
Definition: app_dial.c:729
OPT_ARG_MUSICBACK
@ OPT_ARG_MUSICBACK
Definition: app_dial.c:728
OPT_ARG_CALLEE_GOSUB
@ OPT_ARG_CALLEE_GOSUB
Definition: app_dial.c:730
OPT_ARG_HANGUPCAUSE
@ OPT_ARG_HANGUPCAUSE
Definition: app_dial.c:742
OPT_ARG_FORCE_CID_PRES
@ OPT_ARG_FORCE_CID_PRES
Definition: app_dial.c:739
OPT_ARG_ANNOUNCE
@ OPT_ARG_ANNOUNCE
Definition: app_dial.c:724
OPT_ARG_GOTO
@ OPT_ARG_GOTO
Definition: app_dial.c:726
OPT_ARG_DURATION_LIMIT
@ OPT_ARG_DURATION_LIMIT
Definition: app_dial.c:727
OPT_ARG_ORIGINAL_CLID
@ OPT_ARG_ORIGINAL_CLID
Definition: app_dial.c:736
OPT_ARG_OPERMODE
@ OPT_ARG_OPERMODE
Definition: app_dial.c:734
OPT_ARG_FORCECLID
@ OPT_ARG_FORCECLID
Definition: app_dial.c:737
OPT_ARG_PREDIAL_CALLER
@ OPT_ARG_PREDIAL_CALLER
Definition: app_dial.c:741
OPT_ARG_ARRAY_SIZE
@ OPT_ARG_ARRAY_SIZE
Definition: app_dial.c:744
OPT_ARG_PRIVACY
@ OPT_ARG_PRIVACY
Definition: app_dial.c:732
OPT_ARG_SCREEN_NOINTRO
@ OPT_ARG_SCREEN_NOINTRO
Definition: app_dial.c:735
OPT_ARG_FORCE_CID_TAG
@ OPT_ARG_FORCE_CID_TAG
Definition: app_dial.c:738
topology_ds_info
static const struct ast_datastore_info topology_ds_info
Definition: app_dial.c:831
load_module
static int load_module(void)
Definition: app_dial.c:3602
retrydial_exec
static int retrydial_exec(struct ast_channel *chan, const char *data)
Definition: app_dial.c:3484
dial_handle_playtones
static int dial_handle_playtones(struct ast_channel *chan, const char *data)
Definition: app_dial.c:2253
end_bridge_callback
static void end_bridge_callback(void *data)
Definition: app_dial.c:2237
unload_module
static int unload_module(void)
Definition: app_dial.c:3592
OPT_CALLER_ANSWER
#define OPT_CALLER_ANSWER
Definition: app_dial.c:715
wait_for_answer
static struct ast_channel * wait_for_answer(struct ast_channel *in, struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags, char *opt_args[], struct privacy_args *pa, const struct cause_args *num_in, int *result, char *dtmf_progress, char *mf_progress, char *mf_wink, char *sf_progress, char *sf_wink, const int hearpulsing, const int ignore_cc, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid, struct ast_bridge_config *config)
Definition: app_dial.c:1208
end_bridge_callback_data_fixup
static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator)
Definition: app_dial.c:2249
valid_priv_reply
static int valid_priv_reply(struct ast_flags64 *opts, int res)
Definition: app_dial.c:1991
OPT_CALLEE_GO_ON
#define OPT_CALLEE_GO_ON
Definition: app_dial.c:711
status
jack_status_t status
Definition: app_jack.c:149
str
const char * str
Definition: app_jack.c:150
silencethreshold
static int silencethreshold
Definition: app_voicemail.c:1109
strsep
char * strsep(char **str, const char *delims)
asterisk.h
Asterisk main include file. File version handling, generic pbx functions.
ast_free
#define ast_free(a)
Definition: astmm.h:180
ast_strdup
#define ast_strdup(str)
A wrapper for strdup()
Definition: astmm.h:241
ast_strdupa
#define ast_strdupa(s)
duplicate a string in memory from the stack
Definition: astmm.h:298
ast_asprintf
#define ast_asprintf(ret, fmt,...)
A wrapper for asprintf()
Definition: astmm.h:267
ast_calloc
#define ast_calloc(num, len)
A wrapper for calloc()
Definition: astmm.h:202
ast_log
#define ast_log
Definition: astobj2.c:42
ao2_ref
#define ao2_ref(o, delta)
Reference/unreference an object and return the old refcount.
Definition: astobj2.h:459
ao2_bump
#define ao2_bump(obj)
Bump refcount on an AO2 object by one, returning the object.
Definition: astobj2.h:480
bridge_after.h
After Bridge Execution API.
ast_bridge_setup_after_goto
int ast_bridge_setup_after_goto(struct ast_channel *chan)
Setup any after bridge goto location to begin execution.
Definition: bridge_after.c:435
ast_bridge_set_after_go_on
void ast_bridge_set_after_go_on(struct ast_channel *chan, const char *context, const char *exten, int priority, const char *parseable_goto)
Set channel to go on in the dialplan after the bridge.
Definition: bridge_after.c:622
ast_bridge_set_after_h
void ast_bridge_set_after_h(struct ast_channel *chan, const char *context)
Set channel to run the h exten after the bridge.
Definition: bridge_after.c:617
callerid.h
CallerID (and other GR30) management and generation Includes code and algorithms from the Zapata libr...
AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN
#define AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN
Definition: callerid.h:440
ast_parse_caller_presentation
int ast_parse_caller_presentation(const char *data)
Convert caller ID text code to value (used in config file parsing)
Definition: callerid.c:1343
ast_callerid_parse
int ast_callerid_parse(char *instr, char **name, char **location)
Destructively parse inbuf into name and location (or number)
Definition: callerid.c:1162
AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER
@ AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER
Definition: callerid.h:554
ast_shrink_phone_number
void ast_shrink_phone_number(char *n)
Shrink a phone number in place to just digits (more accurately it just removes ()'s,...
Definition: callerid.c:1101
causes.h
Internal Asterisk hangup causes.
AST_CAUSE_CONGESTION
#define AST_CAUSE_CONGESTION
Definition: causes.h:153
AST_CAUSE_ANSWERED_ELSEWHERE
#define AST_CAUSE_ANSWERED_ELSEWHERE
Definition: causes.h:114
AST_CAUSE_NO_ROUTE_DESTINATION
#define AST_CAUSE_NO_ROUTE_DESTINATION
Definition: causes.h:100
AST_CAUSE_UNREGISTERED
#define AST_CAUSE_UNREGISTERED
Definition: causes.h:154
AST_CAUSE_BUSY
#define AST_CAUSE_BUSY
Definition: causes.h:149
AST_CAUSE_NO_ANSWER
#define AST_CAUSE_NO_ANSWER
Definition: causes.h:109
AST_CAUSE_NORMAL_CLEARING
#define AST_CAUSE_NORMAL_CLEARING
Definition: causes.h:106
ccss.h
Call Completion Supplementary Services API.
ast_cc_call_failed
void ast_cc_call_failed(struct ast_channel *incoming, struct ast_channel *outgoing, const char *const dialstring)
Make CCBS available in the case that ast_call fails.
Definition: ccss.c:4170
ast_ignore_cc
void ast_ignore_cc(struct ast_channel *chan)
Mark the channel to ignore further CC activity.
Definition: ccss.c:3691
ast_cc_is_recall
int ast_cc_is_recall(struct ast_channel *chan, int *core_id, const char *const monitor_type)
Decide if a call to a particular channel is a CC recall.
Definition: ccss.c:3411
ast_handle_cc_control_frame
void ast_handle_cc_control_frame(struct ast_channel *inbound, struct ast_channel *outbound, void *frame_data)
Properly react to a CC control frame.
Definition: ccss.c:2299
ast_cc_completed
int ast_cc_completed(struct ast_channel *chan, const char *const debug,...)
Indicate recall has been acknowledged.
Definition: ccss.c:3813
ast_cc_busy_interface
void ast_cc_busy_interface(struct ast_channel *inbound, struct ast_cc_config_params *cc_params, const char *monitor_type, const char *const device_name, const char *const dialstring, void *private_data)
Callback made from ast_cc_callback for certain channel types.
Definition: ccss.c:4203
ast_cc_extension_monitor_add_dialstring
void ast_cc_extension_monitor_add_dialstring(struct ast_channel *incoming, const char *const dialstring, const char *const device_name)
Add a child dialstring to an extension monitor.
Definition: ccss.c:1989
ast_cc_call_init
int ast_cc_call_init(struct ast_channel *chan, int *ignore_cc)
Start the CC process on a call.
Definition: ccss.c:2392
ast_cc_failed
int ast_cc_failed(int core_id, const char *const debug,...)
Indicate failure has occurred.
Definition: ccss.c:3850
ast_cc_callback
int ast_cc_callback(struct ast_channel *inbound, const char *const tech, const char *const dest, ast_cc_callback_fn callback)
Run a callback for potential matching destinations.
Definition: ccss.c:4215
ast_cdr_reset
int ast_cdr_reset(const char *channel_name, int keep_variables)
Reset the detail record.
Definition: cdr.c:3746
priority
static int priority
Definition: cdr_beanstalkd.c:76
result
static PGresult * result
Definition: cel_pgsql.c:84
config
static const char config[]
Definition: chan_ooh323.c:111
channel.h
General Asterisk PBX channel definitions.
ast_channel_exten_set
void ast_channel_exten_set(struct ast_channel *chan, const char *value)
Definition: channel_internal_api.c:240
ast_waitfordigit
int ast_waitfordigit(struct ast_channel *c, int ms)
Waits for a digit.
Definition: channel.c:3145
ast_str2cause
int ast_str2cause(const char *name) attribute_pure
Convert the string form of a cause code to a number.
Definition: channel.c:625
ast_channel_name
const char * ast_channel_name(const struct ast_channel *chan)
ast_autoservice_stop
int ast_autoservice_stop(struct ast_channel *chan)
Stop servicing a channel for us...
Definition: autoservice.c:266
ast_channel_appl_set
void ast_channel_appl_set(struct ast_channel *chan, const char *value)
Definition: channel_internal_api.c:192
ast_channel_visible_indication_set
void ast_channel_visible_indication_set(struct ast_channel *chan, int value)
Definition: channel_internal_api.c:344
ast_channel_get_device_name
int ast_channel_get_device_name(struct ast_channel *chan, char *device_name, size_t name_buffer_length)
Get a device name given its channel structure.
Definition: channel.c:10510
ast_party_redirecting_init
void ast_party_redirecting_init(struct ast_party_redirecting *init)
Initialize the given redirecting structure.
Definition: channel.c:2090
ast_call
int ast_call(struct ast_channel *chan, const char *addr, int timeout)
Make a call.
Definition: channel.c:6429
ast_channel_clear_flag
void ast_channel_clear_flag(struct ast_channel *chan, unsigned int flag)
Clear a flag on a channel.
Definition: channel.c:11048
ast_channel_datastore_add
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
Add a datastore to a channel.
Definition: channel.c:2355
ast_party_id_init
void ast_party_id_init(struct ast_party_id *init)
Initialize the given party id structure.
Definition: channel.c:1725
ast_channel_connected_line_sub
int ast_channel_connected_line_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *connected_info, int frame)
Run a connected line interception subroutine and update a channel's connected line information.
Definition: channel.c:10352
ast_party_number_init
void ast_party_number_init(struct ast_party_number *init)
Initialize the given number structure.
Definition: channel.c:1612
ast_hangup
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition: channel.c:2511
AST_CHANNEL_REQUESTOR_BRIDGE_PEER
@ AST_CHANNEL_REQUESTOR_BRIDGE_PEER
Definition: channel.h:1525
ast_channel_set_connected_line
void ast_channel_set_connected_line(struct ast_channel *chan, const struct ast_party_connected_line *connected, const struct ast_set_party_connected_line *update)
Set the connected line information in the Asterisk channel.
Definition: channel.c:8322
ast_channel_musicclass
const char * ast_channel_musicclass(const struct ast_channel *chan)
ast_channel_sendhtml
int ast_channel_sendhtml(struct ast_channel *channel, int subclass, const char *data, int datalen)
Sends HTML on given channel Send HTML or URL on link.
Definition: channel.c:6596
ast_party_connected_line_free
void ast_party_connected_line_free(struct ast_party_connected_line *doomed)
Destroy the connected line information contents.
Definition: channel.c:2040
ast_channel_get_up_time_ms
int64_t ast_channel_get_up_time_ms(struct ast_channel *chan)
Obtain how long it has been since the channel was answered in ms.
Definition: channel.c:2806
ast_channel_set_caller_event
void ast_channel_set_caller_event(struct ast_channel *chan, const struct ast_party_caller *caller, const struct ast_set_party_caller *update)
Set the caller id information in the Asterisk channel and generate an AMI event if the caller id name...
Definition: channel.c:7354
ast_waitfor_n
struct ast_channel * ast_waitfor_n(struct ast_channel **chan, int n, int *ms)
Waits for input on a group of channels Wait for input on an array of channels for a given # of millis...
Definition: channel.c:3127
ast_senddigit
int ast_senddigit(struct ast_channel *chan, char digit, unsigned int duration)
Send a DTMF digit to a channel.
Definition: channel.c:4942
ast_channel_lock
#define ast_channel_lock(chan)
Definition: channel.h:2972
ast_channel_make_compatible
int ast_channel_make_compatible(struct ast_channel *chan, struct ast_channel *peer)
Make the frame formats of two channels compatible.
Definition: channel.c:6688
ast_channel_data_set
void ast_channel_data_set(struct ast_channel *chan, const char *value)
Definition: channel_internal_api.c:209
ast_channel_redirecting
struct ast_party_redirecting * ast_channel_redirecting(struct ast_channel *chan)
Definition: channel_internal_api.c:766
ast_channel_transfercapability
unsigned short ast_channel_transfercapability(const struct ast_channel *chan)
Definition: channel_internal_api.c:364
ast_party_connected_line_copy
void ast_party_connected_line_copy(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src)
Copy the source connected line information to the destination connected line.
Definition: channel.c:1999
ast_channel_flags
struct ast_flags * ast_channel_flags(struct ast_channel *chan)
Definition: channel_internal_api.c:1118
ast_party_connected_line_set
void ast_party_connected_line_set(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src, const struct ast_set_party_connected_line *update)
Set the connected line information based on another connected line source.
Definition: channel.c:2022
ast_channel_priority
int ast_channel_priority(const struct ast_channel *chan)
Definition: channel_internal_api.c:307
ast_channel_lock_both
#define ast_channel_lock_both(chan1, chan2)
Lock two channels.
Definition: channel.h:2979
ast_channel_connected
struct ast_party_connected_line * ast_channel_connected(struct ast_channel *chan)
Definition: channel_internal_api.c:750
AST_FLAG_OUTGOING
@ AST_FLAG_OUTGOING
Definition: channel.h:1019
AST_FLAG_END_DTMF_ONLY
@ AST_FLAG_END_DTMF_ONLY
Definition: channel.h:1027
AST_FLAG_MOH
@ AST_FLAG_MOH
Definition: channel.h:1011
ast_channel_datastore_inherit
int ast_channel_datastore_inherit(struct ast_channel *from, struct ast_channel *to)
Inherit datastores from a parent to a child.
Definition: channel.c:2338
ast_channel_req_accountcodes
void ast_channel_req_accountcodes(struct ast_channel *chan, const struct ast_channel *requestor, enum ast_channel_requestor_relationship relationship)
Setup new channel accountcodes from the requestor channel after ast_request().
Definition: channel.c:6402
ast_channel_context
const char * ast_channel_context(const struct ast_channel *chan)
Definition: channel_internal_api.c:215
ast_deactivate_generator
void ast_deactivate_generator(struct ast_channel *chan)
Definition: channel.c:2863
ast_check_hangup_locked
int ast_check_hangup_locked(struct ast_channel *chan)
Definition: channel.c:459
ast_write
int ast_write(struct ast_channel *chan, struct ast_frame *frame)
Write a frame to a channel This function writes the given frame to the indicated channel.
Definition: channel.c:5112
ast_autoservice_start
int ast_autoservice_start(struct ast_channel *chan)
Automatically service a channel for us...
Definition: autoservice.c:200
ast_read
struct ast_frame * ast_read(struct ast_channel *chan)
Reads a frame.
Definition: channel.c:4223
ast_channel_update_connected_line
void ast_channel_update_connected_line(struct ast_channel *chan, const struct ast_party_connected_line *connected, const struct ast_set_party_connected_line *update)
Indicate that the connected line information has changed.
Definition: channel.c:9107
ast_channel_adsicpe
ast_channel_adsicpe
Definition: channel.h:888
ast_party_caller_set_init
void ast_party_caller_set_init(struct ast_party_caller *init, const struct ast_party_caller *guide)
Initialize the given caller structure using the given guide for a set update operation.
Definition: channel.c:1967
ast_party_id_set_init
void ast_party_id_set_init(struct ast_party_id *init, const struct ast_party_id *guide)
Initialize the given party id structure using the given guide for a set update operation.
Definition: channel.c:1748
ast_channel_creationtime
struct timeval ast_channel_creationtime(struct ast_channel *chan)
Definition: channel_internal_api.c:834
ast_channel_redirecting_sub
int ast_channel_redirecting_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *redirecting_info, int is_frame)
Run a redirecting interception subroutine and update a channel's redirecting information.
Definition: channel.c:10397
ast_channel_inherit_variables
void ast_channel_inherit_variables(const struct ast_channel *parent, struct ast_channel *child)
Inherits channel variable from parent to child channel.
Definition: channel.c:6744
AST_FEATURE_AUTOMIXMON
@ AST_FEATURE_AUTOMIXMON
Definition: channel.h:1089
AST_FEATURE_REDIRECT
@ AST_FEATURE_REDIRECT
Definition: channel.h:1084
AST_FEATURE_PARKCALL
@ AST_FEATURE_PARKCALL
Definition: channel.h:1088
AST_FEATURE_AUTOMON
@ AST_FEATURE_AUTOMON
Definition: channel.h:1087
AST_FEATURE_DISCONNECT
@ AST_FEATURE_DISCONNECT
Definition: channel.h:1085
ast_connected_line_parse_data
int ast_connected_line_parse_data(const unsigned char *data, size_t datalen, struct ast_party_connected_line *connected)
Parse connected line indication frame data.
Definition: channel.c:8799
ast_request_with_stream_topology
struct ast_channel * ast_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *addr, int *cause)
Requests a channel (specifying stream topology)
Definition: channel.c:6327
ast_channel_supports_html
int ast_channel_supports_html(struct ast_channel *channel)
Checks for HTML support on a channel.
Definition: channel.c:6591
ast_check_hangup
int ast_check_hangup(struct ast_channel *chan)
Check to see if a channel is needing hang up.
Definition: channel.c:445
ast_channel_get_stream_topology
struct ast_stream_topology * ast_channel_get_stream_topology(const struct ast_channel *chan)
Retrieve the topology of streams on a channel.
Definition: channel_internal_api.c:1476
ast_channel_hangupcause
int ast_channel_hangupcause(const struct ast_channel *chan)
Definition: channel_internal_api.c:298
ast_channel_get_duration_ms
int64_t ast_channel_get_duration_ms(struct ast_channel *chan)
Obtain how long it's been, in milliseconds, since the channel was created.
Definition: channel.c:2791
ast_channel_dialed
struct ast_party_dialed * ast_channel_dialed(struct ast_channel *chan)
Definition: channel_internal_api.c:762
ast_indicate_data
int ast_indicate_data(struct ast_channel *chan, int condition, const void *data, size_t datalen)
Indicates condition of channel, with payload.
Definition: channel.c:4621
ast_channel_zone
struct ast_tone_zone * ast_channel_zone(const struct ast_channel *chan)
Definition: channel_internal_api.c:603
ast_channel_set_flag
void ast_channel_set_flag(struct ast_channel *chan, unsigned int flag)
Set a flag on a channel.
Definition: channel.c:11041
ast_channel_update_redirecting
void ast_channel_update_redirecting(struct ast_channel *chan, const struct ast_party_redirecting *redirecting, const struct ast_set_party_redirecting *update)
Indicate that the redirecting id has changed.
Definition: channel.c:10298
ast_channel_timingfunc
ast_timing_func_t ast_channel_timingfunc(const struct ast_channel *chan)
Definition: channel_internal_api.c:1089
AST_CHANNEL_NAME
#define AST_CHANNEL_NAME
Definition: channel.h:173
ast_channel_whentohangup
struct timeval * ast_channel_whentohangup(struct ast_channel *chan)
Definition: channel_internal_api.c:786
ast_party_number_free
void ast_party_number_free(struct ast_party_number *doomed)
Destroy the party number contents.
Definition: channel.c:1659
ast_party_connected_line_init
void ast_party_connected_line_init(struct ast_party_connected_line *init)
Initialize the given connected line structure.
Definition: channel.c:1990
ast_channel_sendurl
int ast_channel_sendurl(struct ast_channel *channel, const char *url)
Sends a URL on a given link Send URL on link.
Definition: channel.c:6603
ast_channel_language
const char * ast_channel_language(const struct ast_channel *chan)
ast_cause2str
const char * ast_cause2str(int cause) attribute_pure
Gives the string form of a given cause code.
Definition: channel.c:612
ast_party_redirecting_free
void ast_party_redirecting_free(struct ast_party_redirecting *doomed)
Destroy the redirecting information contents.
Definition: channel.c:2147
ast_channel_context_set
void ast_channel_context_set(struct ast_channel *chan, const char *value)
Definition: channel_internal_api.c:223
ast_channel_sched
struct ast_sched_context * ast_channel_sched(const struct ast_channel *chan)
Definition: channel_internal_api.c:587
ast_connected_line_copy_from_caller
void ast_connected_line_copy_from_caller(struct ast_party_connected_line *dest, const struct ast_party_caller *src)
Copy the caller information to the connected line information.
Definition: channel.c:8307
ast_channel_call_forward
const char * ast_channel_call_forward(const struct ast_channel *chan)
ast_channel_stream
struct ast_filestream * ast_channel_stream(const struct ast_channel *chan)
Definition: channel_internal_api.c:479
ast_pre_call
int ast_pre_call(struct ast_channel *chan, const char *sub_args)
Execute a Gosub call on the channel before a call is placed.
Definition: channel.c:6412
ast_channel_setoption
int ast_channel_setoption(struct ast_channel *channel, int option, void *data, int datalen, int block)
Sets an option on a channel.
Definition: channel.c:7404
ast_channel_caller
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
Definition: channel_internal_api.c:746
ast_party_connected_line_set_init
void ast_party_connected_line_set_init(struct ast_party_connected_line *init, const struct ast_party_connected_line *guide)
Initialize the given connected line structure using the given guide for a set update operation.
Definition: channel.c:2013
ast_channel_transfercapability_set
void ast_channel_transfercapability_set(struct ast_channel *chan, unsigned short value)
Definition: channel_internal_api.c:368
ast_channel_priority_set
void ast_channel_priority_set(struct ast_channel *chan, int value)
Definition: channel_internal_api.c:311
ast_channel_hangupcause_set
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
Definition: channel_internal_api.c:302
ast_autoservice_chan_hangup_peer
void ast_autoservice_chan_hangup_peer(struct ast_channel *chan, struct ast_channel *peer)
Put chan into autoservice while hanging up peer.
Definition: autoservice.c:349
ast_channel_whentohangup_set
void ast_channel_whentohangup_set(struct ast_channel *chan, struct timeval *value)
Definition: channel_internal_api.c:826
ast_answer
int ast_answer(struct ast_channel *chan)
Answer a channel.
Definition: channel.c:2776
ast_channel_adsicpe_set
void ast_channel_adsicpe_set(struct ast_channel *chan, enum ast_channel_adsicpe value)
Definition: channel_internal_api.c:643
ast_channel_early_bridge
int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
Bridge two channels together (early)
Definition: channel.c:7394
ast_indicate
int ast_indicate(struct ast_channel *chan, int condition)
Indicates condition of channel.
Definition: channel.c:4243
ast_channel_exten
const char * ast_channel_exten(const struct ast_channel *chan)
Definition: channel_internal_api.c:232
ast_channel_unlock
#define ast_channel_unlock(chan)
Definition: channel.h:2973
AST_MAX_EXTENSION
#define AST_MAX_EXTENSION
Definition: channel.h:134
ast_party_redirecting_copy
void ast_party_redirecting_copy(struct ast_party_redirecting *dest, const struct ast_party_redirecting *src)
Copy the source redirecting information to the destination redirecting.
Definition: channel.c:2103
ast_channel_datastore_find
struct ast_datastore * ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
Find a datastore on a channel.
Definition: channel.c:2369
ast_channel_state
ast_channel_state
ast_channel states
Definition: channelstate.h:35
AST_STATE_UP
@ AST_STATE_UP
Definition: channelstate.h:42
ast_datastore_alloc
#define ast_datastore_alloc(info, uid)
Definition: datastore.h:85
dial.h
Dialing API.
ast_hangup_cause_to_dial_status
const char * ast_hangup_cause_to_dial_status(int hangup_cause)
Convert a hangup cause to a publishable dial status.
Definition: dial.c:749
dsp.h
Convenient Signal Processing routines.
THRESHOLD_SILENCE
@ THRESHOLD_SILENCE
Definition: dsp.h:73
ast_dsp_get_threshold_from_settings
int ast_dsp_get_threshold_from_settings(enum threshold which)
Get silence threshold from dsp.conf.
Definition: dsp.c:2011
connected
char connected
Definition: eagi_proxy.c:82
buf
char buf[BUFSIZE]
Definition: eagi_proxy.c:66
features.h
Call Parking and Pickup API Includes code and algorithms from the Zapata library.
ast_bridge_call
int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
Bridge a call, optionally allowing redirection.
Definition: features.c:694
ast_bridge_timelimit
int ast_bridge_timelimit(struct ast_channel *chan, struct ast_bridge_config *config, char *parse, struct timeval *calldurationlimit)
parse L option and read associated channel variables to set warning, warning frequency,...
Definition: features.c:866
file.h
Generic File Format Support. Should be included by clients of the file handling routines....
ast_stopstream
int ast_stopstream(struct ast_channel *c)
Stops a stream.
Definition: file.c:223
ast_streamfile
int ast_streamfile(struct ast_channel *c, const char *filename, const char *preflang)
Streams a file.
Definition: file.c:1312
ast_fileexists
int ast_fileexists(const char *filename, const char *fmt, const char *preflang)
Checks for the existence of a given file.
Definition: file.c:1148
ast_filedelete
int ast_filedelete(const char *filename, const char *fmt)
Deletes a file.
Definition: file.c:1160
AST_DIGIT_ANY
#define AST_DIGIT_ANY
Definition: file.h:48
ast_waitstream
int ast_waitstream(struct ast_channel *c, const char *breakon)
Waits for a stream to stop or digit to be pressed.
Definition: file.c:1874
name
static const char name[]
Definition: format_mp3.c:68
framehook.h
FrameHook Architecture.
SCOPE_EXIT_RTN_VALUE
#define SCOPE_EXIT_RTN_VALUE(__return_value,...)
Definition: include/asterisk/logger.h:1034
SCOPE_ENTER
#define SCOPE_ENTER(level,...)
Definition: include/asterisk/logger.h:1006
ast_trace
#define ast_trace(level,...)
Definition: include/asterisk/logger.h:997
ast_channel_publish_dial
void ast_channel_publish_dial(struct ast_channel *caller, struct ast_channel *peer, const char *dialstring, const char *dialstatus)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition: stasis_channels.c:765
ast_channel_stage_snapshot_done
void ast_channel_stage_snapshot_done(struct ast_channel *chan)
Clear flag to indicate channel snapshot is being staged, and publish snapshot.
Definition: stasis_channels.c:1069
ast_channel_publish_snapshot
void ast_channel_publish_snapshot(struct ast_channel *chan)
Publish a ast_channel_snapshot for a channel.
Definition: stasis_channels.c:1081
ast_channel_stage_snapshot
void ast_channel_stage_snapshot(struct ast_channel *chan)
Set flag to indicate channel snapshot is being staged.
Definition: stasis_channels.c:1064
ast_channel_publish_dial_forward
void ast_channel_publish_dial_forward(struct ast_channel *caller, struct ast_channel *peer, struct ast_channel *forwarded, const char *dialstring, const char *dialstatus, const char *forward)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition: stasis_channels.c:715
app.h
Application convenience functions, designed to give consistent look and feel to Asterisk apps.
ast_app_expand_sub_args
const char * ast_app_expand_sub_args(struct ast_channel *chan, const char *args)
Add missing context/exten to subroutine argument string.
Definition: main/app.c:278
AST_APP_ARG
#define AST_APP_ARG(name)
Define an application argument.
Definition: include/asterisk/app.h:1230
ast_sf_stream
int ast_sf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int frequency, int is_external)
Send a string of SF digits to a channel.
Definition: main/app.c:1097
END_OPTIONS
#define END_OPTIONS
Definition: include/asterisk/app.h:1342
AST_APP_OPTIONS
#define AST_APP_OPTIONS(holder, options...)
Declares an array of options for an application.
Definition: include/asterisk/app.h:1391
ast_play_and_record
int ast_play_and_record(struct ast_channel *chan, const char *playfile, const char *recordfile, int maxtime_sec, const char *fmt, int *duration, int *sound_duration, int silencethreshold, int maxsilence_ms, const char *path)
Record a file based on input from a channel. Use default accept and cancel DTMF. This function will p...
Definition: main/app.c:2154
ast_app_group_set_channel
int ast_app_group_set_channel(struct ast_channel *chan, const char *data)
Set the group for a channel, splitting the provided data into group and category, if specified.
Definition: main/app.c:2193
AST_APP_OPTION_ARG
#define AST_APP_OPTION_ARG(option, flagno, argno)
Declares an application option that accepts an argument.
Definition: include/asterisk/app.h:1411
AST_DECLARE_APP_ARGS
#define AST_DECLARE_APP_ARGS(name, arglist)
Declare a structure to hold an application's arguments.
Definition: include/asterisk/app.h:1247
BEGIN_OPTIONS
#define BEGIN_OPTIONS
Definition: include/asterisk/app.h:1341
ast_play_and_wait
int ast_play_and_wait(struct ast_channel *chan, const char *fn)
Play a stream and wait for a digit, returning the digit that was pressed.
Definition: main/app.c:1616
AST_STANDARD_APP_ARGS
#define AST_STANDARD_APP_ARGS(args, parse)
Performs the 'standard' argument separation process for an application.
Definition: include/asterisk/app.h:1283
AST_APP_OPTION
#define AST_APP_OPTION(option, flagno)
Declares an application option that does not accept an argument.
Definition: include/asterisk/app.h:1400
ast_app_exec_sub
int ast_app_exec_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const char *sub_args, int ignore_hangup)
Run a subroutine on a channel, placing an optional second channel into autoservice.
Definition: main/app.c:297
ast_dtmf_stream
int ast_dtmf_stream(struct ast_channel *chan, struct ast_channel *peer, const char *digits, int between, unsigned int duration)
Send a string of DTMF digits to a channel.
Definition: main/app.c:1127
ast_mf_stream
int ast_mf_stream(struct ast_channel *chan, struct ast_channel *peer, struct ast_channel *chan2, const char *digits, int between, unsigned int duration, unsigned int durationkp, unsigned int durationst, int is_external)
Send a string of MF digits to a channel.
Definition: main/app.c:1113
ast_app_parse_options64
int ast_app_parse_options64(const struct ast_app_option *options, struct ast_flags64 *flags, char **args, char *optstr)
Parses a string containing application options and sets flags/arguments.
Definition: main/app.c:3071
config.h
Configuration File Parser.
features_config.h
AST_FEATURE_MAX_LEN
#define AST_FEATURE_MAX_LEN
Definition: include/asterisk/features_config.h:231
ast_get_builtin_feature
int ast_get_builtin_feature(struct ast_channel *chan, const char *feature, char *buf, size_t len)
Get the DTMF code for a builtin feature.
Definition: features_config.c:1305
AST_FRAME_DTMF
#define AST_FRAME_DTMF
Definition: include/asterisk/frame.h:139
AST_OPTION_OPRMODE
#define AST_OPTION_OPRMODE
Definition: include/asterisk/frame.h:463
ast_frfree
#define ast_frfree(fr)
Definition: include/asterisk/frame.h:565
AST_FRAME_VIDEO
@ AST_FRAME_VIDEO
Definition: include/asterisk/frame.h:103
AST_FRAME_HTML
@ AST_FRAME_HTML
Definition: include/asterisk/frame.h:118
AST_FRAME_IMAGE
@ AST_FRAME_IMAGE
Definition: include/asterisk/frame.h:116
AST_FRAME_DTMF_END
@ AST_FRAME_DTMF_END
Definition: include/asterisk/frame.h:99
AST_FRAME_DTMF_BEGIN
@ AST_FRAME_DTMF_BEGIN
Definition: include/asterisk/frame.h:125
AST_FRAME_VOICE
@ AST_FRAME_VOICE
Definition: include/asterisk/frame.h:101
AST_FRAME_CONTROL
@ AST_FRAME_CONTROL
Definition: include/asterisk/frame.h:105
AST_FRAME_TEXT
@ AST_FRAME_TEXT
Definition: include/asterisk/frame.h:114
AST_CONTROL_SRCUPDATE
@ AST_CONTROL_SRCUPDATE
Definition: include/asterisk/frame.h:297
AST_CONTROL_WINK
@ AST_CONTROL_WINK
Definition: include/asterisk/frame.h:287
AST_CONTROL_PROGRESS
@ AST_CONTROL_PROGRESS
Definition: include/asterisk/frame.h:291
AST_CONTROL_OFFHOOK
@ AST_CONTROL_OFFHOOK
Definition: include/asterisk/frame.h:284
AST_CONTROL_BUSY
@ AST_CONTROL_BUSY
Definition: include/asterisk/frame.h:282
AST_CONTROL_UNHOLD
@ AST_CONTROL_UNHOLD
Definition: include/asterisk/frame.h:294
AST_CONTROL_VIDUPDATE
@ AST_CONTROL_VIDUPDATE
Definition: include/asterisk/frame.h:295
AST_CONTROL_PROCEEDING
@ AST_CONTROL_PROCEEDING
Definition: include/asterisk/frame.h:292
AST_CONTROL_REDIRECTING
@ AST_CONTROL_REDIRECTING
Definition: include/asterisk/frame.h:300
AST_CONTROL_CONGESTION
@ AST_CONTROL_CONGESTION
Definition: include/asterisk/frame.h:285
AST_CONTROL_PLAYBACK_BEGIN
@ AST_CONTROL_PLAYBACK_BEGIN
Definition: include/asterisk/frame.h:338
AST_CONTROL_CC
@ AST_CONTROL_CC
Definition: include/asterisk/frame.h:302
AST_CONTROL_ANSWER
@ AST_CONTROL_ANSWER
Definition: include/asterisk/frame.h:281
AST_CONTROL_RINGING
@ AST_CONTROL_RINGING
Definition: include/asterisk/frame.h:280
AST_CONTROL_HANGUP
@ AST_CONTROL_HANGUP
Definition: include/asterisk/frame.h:278
AST_CONTROL_HOLD
@ AST_CONTROL_HOLD
Definition: include/asterisk/frame.h:293
AST_CONTROL_CONNECTED_LINE
@ AST_CONTROL_CONNECTED_LINE
Definition: include/asterisk/frame.h:299
AST_CONTROL_FLASH
@ AST_CONTROL_FLASH
Definition: include/asterisk/frame.h:286
AST_CONTROL_AOC
@ AST_CONTROL_AOC
Definition: include/asterisk/frame.h:305
AST_CONTROL_SRCCHANGE
@ AST_CONTROL_SRCCHANGE
Definition: include/asterisk/frame.h:303
AST_CONTROL_PVT_CAUSE_CODE
@ AST_CONTROL_PVT_CAUSE_CODE
Definition: include/asterisk/frame.h:310
ast_debug
#define ast_debug(level,...)
Log a DEBUG message.
Definition: include/asterisk/logger.h:454
LOG_ERROR
#define LOG_ERROR
Definition: include/asterisk/logger.h:289
ast_verb
#define ast_verb(level,...)
Definition: include/asterisk/logger.h:465
LOG_NOTICE
#define LOG_NOTICE
Definition: include/asterisk/logger.h:267
LOG_WARNING
#define LOG_WARNING
Definition: include/asterisk/logger.h:278
indications.h
Tone Indication Support.
ast_tone_zone_sound_unref
static struct ast_tone_zone_sound * ast_tone_zone_sound_unref(struct ast_tone_zone_sound *ts)
Release a reference to an ast_tone_zone_sound.
Definition: indications.h:227
ast_playtones_start
int ast_playtones_start(struct ast_channel *chan, int vol, const char *tonelist, int interruptible)
Start playing a list of tones on a channel.
Definition: indications.c:302
ast_get_indication_tone
struct ast_tone_zone_sound * ast_get_indication_tone(const struct ast_tone_zone *zone, const char *indication)
Locate a tone zone sound.
Definition: indications.c:461
AST_LIST_HEAD_NOLOCK
#define AST_LIST_HEAD_NOLOCK(name, type)
Defines a structure to be used to hold a list of specified type (with no lock).
Definition: linkedlists.h:225
AST_LIST_TRAVERSE
#define AST_LIST_TRAVERSE(head, var, field)
Loops over (traverses) the entries in a list.
Definition: linkedlists.h:491
AST_LIST_EMPTY
#define AST_LIST_EMPTY(head)
Checks whether the specified list contains any entries.
Definition: linkedlists.h:450
AST_LIST_INSERT_TAIL
#define AST_LIST_INSERT_TAIL(head, elm, field)
Appends a list entry to the tail of a list.
Definition: linkedlists.h:731
AST_LIST_HEAD_NOLOCK_INIT_VALUE
#define AST_LIST_HEAD_NOLOCK_INIT_VALUE
Defines initial values for a declaration of AST_LIST_HEAD_NOLOCK.
Definition: linkedlists.h:252
AST_LIST_ENTRY
#define AST_LIST_ENTRY(type)
Declare a forward link structure inside a list entry.
Definition: linkedlists.h:410
AST_LIST_TRAVERSE_SAFE_END
#define AST_LIST_TRAVERSE_SAFE_END
Closes a safe loop traversal block.
Definition: linkedlists.h:615
AST_LIST_TRAVERSE_SAFE_BEGIN
#define AST_LIST_TRAVERSE_SAFE_BEGIN(head, var, field)
Loops safely over (traverses) the entries in a list.
Definition: linkedlists.h:529
AST_LIST_REMOVE_CURRENT
#define AST_LIST_REMOVE_CURRENT(field)
Removes the current entry from a list during a traversal.
Definition: linkedlists.h:557
AST_LIST_REMOVE_HEAD
#define AST_LIST_REMOVE_HEAD(head, field)
Removes and returns the head entry from a list.
Definition: linkedlists.h:833
AST_LIST_FIRST
#define AST_LIST_FIRST(head)
Returns the first entry contained in a list.
Definition: linkedlists.h:421
AST_LIST_NEXT
#define AST_LIST_NEXT(elm, field)
Returns the next entry in the list after the given entry.
Definition: linkedlists.h:439
lock.h
Asterisk locking-related definitions:
manager.h
The AMI - Asterisk Manager Interface - is a TCP protocol created to manage Asterisk with third-party ...
max_forwards.h
ast_max_forwards_decrement
int ast_max_forwards_decrement(struct ast_channel *chan)
Decrement the max forwards count for a particular channel.
Definition: max_forwards.c:135
ast_max_forwards_get
int ast_max_forwards_get(struct ast_channel *chan)
Get the current max forwards for a particular channel.
Definition: max_forwards.c:121
module.h
Asterisk module definitions.
AST_MODFLAG_DEFAULT
@ AST_MODFLAG_DEFAULT
Definition: module.h:329
AST_MODULE_INFO
#define AST_MODULE_INFO(keystr, flags_to_set, desc, fields...)
Definition: module.h:557
AST_MODULE_SUPPORT_CORE
@ AST_MODULE_SUPPORT_CORE
Definition: module.h:121
ASTERISK_GPL_KEY
#define ASTERISK_GPL_KEY
The text the key() function should return.
Definition: module.h:46
ast_unregister_application
int ast_unregister_application(const char *app)
Unregister an application.
Definition: pbx_app.c:392
ast_register_application_xml
#define ast_register_application_xml(app, execute)
Register an application using XML documentation.
Definition: module.h:640
musiconhold.h
Music on hold handling.
ast_moh_start
int ast_moh_start(struct ast_channel *chan, const char *mclass, const char *interpclass)
Turn on music on hold on a given channel.
Definition: channel.c:7748
ast_moh_stop
void ast_moh_stop(struct ast_channel *chan)
Turn off music on hold on a given channel.
Definition: channel.c:7758
paths.h
Asterisk file paths, configured in asterisk.conf.
ast_config_AST_DATA_DIR
const char * ast_config_AST_DATA_DIR
Definition: options.c:159
pbx.h
Core PBX routines and definitions.
pbx_builtin_getvar_helper
const char * pbx_builtin_getvar_helper(struct ast_channel *chan, const char *name)
Return a pointer to the value of the corresponding channel variable.
Definition: pbx_variables.c:1162
ast_exists_extension
int ast_exists_extension(struct ast_channel *c, const char *context, const char *exten, int priority, const char *callerid)
Determine whether an extension exists.
Definition: pbx.c:4196
AST_PBX_INCOMPLETE
#define AST_PBX_INCOMPLETE
Definition: pbx.h:51
pbx_builtin_setvar_helper
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name.
Definition: pbx_variables.c:1233
ast_goto_if_exists
int ast_goto_if_exists(struct ast_channel *chan, const char *context, const char *exten, int priority)
Definition: pbx.c:8806
ast_pbx_start
enum ast_pbx_result ast_pbx_start(struct ast_channel *c)
Create a new thread and start the PBX.
Definition: pbx.c:4729
ast_get_hint
int ast_get_hint(char *hint, int hintsize, char *name, int namesize, struct ast_channel *c, const char *context, const char *exten)
If an extension hint exists, return non-zero.
Definition: pbx.c:4158
ast_parseable_goto
int ast_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition: pbx.c:8891
privacy.h
Persistent data storage (akin to *doze registry)
AST_PRIVACY_KILL
#define AST_PRIVACY_KILL
Definition: privacy.h:32
AST_PRIVACY_ALLOW
#define AST_PRIVACY_ALLOW
Definition: privacy.h:31
AST_PRIVACY_DENY
#define AST_PRIVACY_DENY
Definition: privacy.h:30
ast_privacy_set
int ast_privacy_set(char *dest, char *cid, int status)
Definition: privacy.c:82
ast_privacy_check
int ast_privacy_check(char *dest, char *cid)
Definition: privacy.c:46
AST_PRIVACY_UNKNOWN
#define AST_PRIVACY_UNKNOWN
Definition: privacy.h:34
AST_PRIVACY_TORTURE
#define AST_PRIVACY_TORTURE
Definition: privacy.h:33
url
static char url[512]
Definition: res_config_ldap.c:74
NULL
#define NULL
Definition: resample.c:96
rtp_engine.h
Pluggable RTP Architecture.
ast_rtp_instance_early_bridge_make_compatible
void ast_rtp_instance_early_bridge_make_compatible(struct ast_channel *c_dst, struct ast_channel *c_src)
Make two channels compatible for early bridging.
Definition: rtp_engine.c:2443
say.h
Say numbers and dates (maybe words one day too)
ast_sched_runq
int ast_sched_runq(struct ast_sched_context *con)
Runs the queue.
Definition: sched.c:786
ast_sched_wait
int ast_sched_wait(struct ast_sched_context *con) attribute_warn_unused_result
Determines number of seconds until the next outstanding event to take place.
Definition: sched.c:433
stasis_channels.h
stream.h
Media Stream API.
AST_STREAM_STATE_RECVONLY
@ AST_STREAM_STATE_RECVONLY
Set when the stream is receiving media only.
Definition: stream.h:90
AST_STREAM_STATE_SENDONLY
@ AST_STREAM_STATE_SENDONLY
Set when the stream is sending media only.
Definition: stream.h:86
ast_stream_set_state
void ast_stream_set_state(struct ast_stream *stream, enum ast_stream_state state)
Set the state of a stream.
Definition: stream.c:380
ast_stream_topology_to_str
const char * ast_stream_topology_to_str(const struct ast_stream_topology *topology, struct ast_str **buf)
Get a string representing the topology for debugging/display purposes.
Definition: stream.c:939
ast_stream_topology_get_stream
struct ast_stream * ast_stream_topology_get_stream(const struct ast_stream_topology *topology, unsigned int position)
Get a specific stream from the topology.
Definition: stream.c:791
ast_stream_topology_get_count
int ast_stream_topology_get_count(const struct ast_stream_topology *topology)
Get the number of streams in a topology.
Definition: stream.c:768
ast_stream_get_state
enum ast_stream_state ast_stream_get_state(const struct ast_stream *stream)
Get the current state of a stream.
Definition: stream.c:373
ast_stream_topology_free
void ast_stream_topology_free(struct ast_stream_topology *topology)
Unreference and destroy a stream topology.
Definition: stream.c:746
ast_stream_topology_clone
struct ast_stream_topology * ast_stream_topology_clone(const struct ast_stream_topology *topology)
Create a deep clone of an existing stream topology.
Definition: stream.c:670
ast_str_append
int ast_str_append(struct ast_str **buf, ssize_t max_len, const char *fmt,...)
Append to a thread local dynamic string.
Definition: strings.h:1139
ast_str_buffer
char * ast_str_buffer(const struct ast_str *buf)
Returns the string buffer within the ast_str buf.
Definition: strings.h:761
S_COR
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition: strings.h:87
ast_strlen_zero
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition: strings.h:65
ast_str_tmp
#define ast_str_tmp(init_len, __expr)
Provides a temporary ast_str and returns a copy of its buffer.
Definition: strings.h:1189
ast_str_alloca
#define ast_str_alloca(init_len)
Definition: strings.h:848
ast_str_reset
void ast_str_reset(struct ast_str *buf)
Reset the content of a dynamic string. Useful before a series of ast_str_append.
Definition: strings.h:693
ast_str_strlen
size_t ast_str_strlen(const struct ast_str *buf)
Returns the current length of the string stored within buf.
Definition: strings.h:730
ast_copy_string
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition: strings.h:425
ast_strip
char * ast_strip(char *s)
Strip leading/trailing whitespace from a string.
Definition: strings.h:223
ast_aoc_decoded
Definition: aoc.c:214
ast_aoc_encoded
Definition: aoc.c:206
ast_bridge_config
bridge configuration
Definition: channel.h:1096
ast_bridge_config::end_bridge_callback_data
void * end_bridge_callback_data
Definition: channel.h:1111
ast_channel
Main Channel structure associated with a channel.
Definition: channel_private.h:48
ast_channel::exten
char exten[AST_MAX_EXTENSION]
Definition: channel_private.h:181
ast_channel::data
const char * data
Definition: channel_private.h:58
ast_channel::tech
const struct ast_channel_tech * tech
Definition: channel_private.h:49
ast_datastore_info
Structure for a data store type.
Definition: datastore.h:31
ast_datastore_info::type
const char * type
Definition: datastore.h:32
ast_datastore
Structure for a data store object.
Definition: datastore.h:64
ast_datastore::data
void * data
Definition: datastore.h:66
ast_flags64
Structure used to handle a large number of boolean flags == used only in app_dial?
Definition: utils.h:222
ast_flags64::flags
uint64_t flags
Definition: utils.h:223
ast_frame_subclass::topology
struct ast_stream_topology * topology
Definition: include/asterisk/frame.h:157
ast_frame_subclass::integer
int integer
Definition: include/asterisk/frame.h:152
ast_frame
Data structure associated with a single frame of data.
Definition: include/asterisk/frame.h:165
ast_frame::subclass
struct ast_frame_subclass subclass
Definition: include/asterisk/frame.h:169
ast_frame::uint32
uint32_t uint32
Definition: include/asterisk/frame.h:183
ast_frame::frametype
enum ast_frame_type frametype
Definition: include/asterisk/frame.h:167
ast_frame::data
union ast_frame::@231 data
ast_frame::datalen
int datalen
Definition: include/asterisk/frame.h:171
ast_frame::ptr
void * ptr
Definition: include/asterisk/frame.h:183
ast_party_caller
Caller Party information.
Definition: channel.h:420
ast_party_caller::id
struct ast_party_id id
Caller party ID.
Definition: channel.h:422
ast_party_connected_line
Connected Line/Party information.
Definition: channel.h:458
ast_party_connected_line::source
int source
Information about the source of an update.
Definition: channel.h:484
ast_party_connected_line::id
struct ast_party_id id
Connected party ID.
Definition: channel.h:460
ast_party_dialed::transit_network_select
int transit_network_select
Transit Network Select.
Definition: channel.h:399
ast_party_id
Information needed to identify an endpoint in a call.
Definition: channel.h:340
ast_party_id::subaddress
struct ast_party_subaddress subaddress
Subscriber subaddress.
Definition: channel.h:346
ast_party_id::tag
char * tag
User-set "tag".
Definition: channel.h:356
ast_party_id::name
struct ast_party_name name
Subscriber name.
Definition: channel.h:342
ast_party_id::number
struct ast_party_number number
Subscriber phone number.
Definition: channel.h:344
ast_party_name::valid
unsigned char valid
TRUE if the name information is valid/present.
Definition: channel.h:281
ast_party_name::str
char * str
Subscriber name (Malloced)
Definition: channel.h:266
ast_party_number::presentation
int presentation
Q.931 presentation-indicator and screening-indicator encoded fields.
Definition: channel.h:297
ast_party_number::valid
unsigned char valid
TRUE if the number information is valid/present.
Definition: channel.h:299
ast_party_number::str
char * str
Subscriber phone number (Malloced)
Definition: channel.h:293
ast_party_redirecting
Redirecting Line information. RDNIS (Redirecting Directory Number Information Service) Where a call d...
Definition: channel.h:524
ast_party_redirecting::from
struct ast_party_id from
Who is redirecting the call (Sent to the party the call is redirected toward)
Definition: channel.h:529
ast_party_redirecting::to
struct ast_party_id to
Call is redirecting to a new party (Sent to the caller)
Definition: channel.h:532
ast_party_subaddress::str
char * str
Malloced subaddress string.
Definition: channel.h:315
ast_str
Support for dynamic strings.
Definition: strings.h:623
ast_stream_topology
Definition: stream.c:123
ast_stream
Definition: stream.c:81
ast_tone_zone_sound
Description of a tone.
Definition: indications.h:35
ast_tone_zone_sound::data
const char * data
Description of a tone.
Definition: indications.h:52
cause_args
Definition: app_dial.c:869
cause_args::congestion
int congestion
Definition: app_dial.c:872
cause_args::nochan
int nochan
Definition: app_dial.c:873
cause_args::chan
struct ast_channel * chan
Definition: app_dial.c:870
cause_args::busy
int busy
Definition: app_dial.c:871
chanlist
List of channel drivers.
Definition: app_dial.c:803
chanlist::number
const char * number
Definition: app_dial.c:811
chanlist::interface
const char * interface
Definition: app_dial.c:807
chanlist::aoc_s_rate_list
struct ast_aoc_decoded * aoc_s_rate_list
Definition: app_dial.c:819
chanlist::connected
struct ast_party_connected_line connected
Definition: app_dial.c:816
chanlist::orig_chan_name
char * orig_chan_name
Definition: app_dial.c:813
chanlist::stuff
char stuff[0]
Definition: app_dial.c:821
chanlist::chan
struct ast_channel * chan
Definition: app_dial.c:805
chanlist::flags
uint64_t flags
Definition: app_dial.c:814
chanlist::tech
const char * tech
Definition: app_dial.c:809
chanlist::node
struct chanlist::@16 node
chanlist::pending_connected_update
unsigned int pending_connected_update
Definition: app_dial.c:818
chans
Definition: astman.c:88
dial_head
Definition: app_dial.c:824
extension
structure to hold extensions
Definition: res_phoneprov.c:293
max_forwards
Channel datastore data for max forwards.
Definition: max_forwards.c:29
node
Definition: test_heap.c:38
number
Number structure.
Definition: app_followme.c:157
oprmode::peer
struct ast_channel * peer
Definition: include/asterisk/frame.h:527
privacy_args
Definition: app_dial.c:1146
privacy_args::status
char status[256]
Definition: app_dial.c:1151
privacy_args::privdb_val
int privdb_val
Definition: app_dial.c:1148
privacy_args::canceled
int canceled
Definition: app_dial.c:1152
privacy_args::privcid
char privcid[256]
Definition: app_dial.c:1149
privacy_args::sentringing
int sentringing
Definition: app_dial.c:1147
privacy_args::privintro
char privintro[1024]
Definition: app_dial.c:1150
done
int done
Definition: test_amihooks.c:48
args
const char * args
Definition: test_func_file.c:46
options
static struct test_options options
Definition: test_http_media_cache.c:63
c
static struct test_val c
Definition: test_linkedlists.c:48
ast_tvzero
int ast_tvzero(const struct timeval t)
Returns true if the argument is 0,0.
Definition: time.h:117
ast_remaining_ms
int ast_remaining_ms(struct timeval start, int max_ms)
Calculate remaining milliseconds given a starting timestamp and upper bound.
Definition: utils.c:2317
ast_tvadd
struct timeval ast_tvadd(struct timeval a, struct timeval b)
Returns the sum of two timevals a + b.
Definition: extconf.c:2280
ast_tvdiff_ms
int64_t ast_tvdiff_ms(struct timeval end, struct timeval start)
Computes the difference (in milliseconds) between two struct timeval instances.
Definition: time.h:107
ast_tvnow
struct timeval ast_tvnow(void)
Returns current timeval. Meant to replace calls to gettimeofday().
Definition: time.h:159
translate.h
Support for translation of data formats. translate.c.
out
FILE * out
Definition: utils/frame.c:33
in
FILE * in
Definition: utils/frame.c:33
utils.h
Utility functions.
ast_test_flag
#define ast_test_flag(p, flag)
Definition: utils.h:63
ast_set2_flag64
#define ast_set2_flag64(p, value, flag)
Definition: utils.h:168
ast_test_flag64
#define ast_test_flag64(p, flag)
Definition: utils.h:137
ast_clear_flag64
#define ast_clear_flag64(p, flag)
Definition: utils.h:151
ast_clear_flag
#define ast_clear_flag(p, flag)
Definition: utils.h:77
ast_mkdir
int ast_mkdir(const char *path, int mode)
Recursively create directory path.
Definition: utils.c:2515
ast_copy_flags64
#define ast_copy_flags64(dest, src, flagz)
Definition: utils.h:158
ast_set_flag64
#define ast_set_flag64(p, flag)
Definition: utils.h:144
ast_set_flag
#define ast_set_flag(p, flag)
Definition: utils.h:70
ast_replace_subargument_delimiter
void ast_replace_subargument_delimiter(char *s)
Replace '^' in a string with ','.
Definition: utils.c:2379
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