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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 <para>The timeouts need not be whole numbers; both arguments accept fractional seconds.</para>
101 </parameter>
102 <parameter name="options" required="false">
103 <optionlist>
104 <option name="A" argsep=":">
105 <argument name="x">
106 <para>The file to play to the called party</para>
107 </argument>
108 <argument name="y">
109 <para>The file to play to the calling party</para>
110 </argument>
111 <para>Play an announcement to the called and/or calling parties, where <replaceable>x</replaceable>
112 is the prompt to be played to the called party and <replaceable>y</replaceable> is the prompt
113 to be played to the caller. The files may be different and will be played to each party
114 simultaneously.</para>
115 </option>
116 <option name="a">
117 <para>Immediately answer the calling channel when the called channel answers in
118 all cases. Normally, the calling channel is answered when the called channel
119 answers, but when options such as <literal>A()</literal> and
120 <literal>M()</literal> are used, the calling channel is
121 not answered until all actions on the called channel (such as playing an
122 announcement) are completed. This option can be used to answer the calling
123 channel before doing anything on the called channel. You will rarely need to use
124 this option, the default behavior is adequate in most cases.</para>
125 </option>
126 <option name="b" argsep="^">
127 <para>Before initiating an outgoing call, <literal>Gosub</literal> to the specified
128 location using the newly created channel. The <literal>Gosub</literal> will be
129 executed for each destination channel.</para>
130 <argument name="context" required="false" />
131 <argument name="exten" required="false" />
132 <argument name="priority" required="true" hasparams="optional" argsep="^">
133 <argument name="arg1" multiple="true" required="true" />
134 <argument name="argN" />
135 </argument>
136 </option>
137 <option name="B" argsep="^">
138 <para>Before initiating the outgoing call(s), <literal>Gosub</literal> to the
139 specified location using the current channel.</para>
140 <argument name="context" required="false" />
141 <argument name="exten" required="false" />
142 <argument name="priority" required="true" hasparams="optional" argsep="^">
143 <argument name="arg1" multiple="true" required="true" />
144 <argument name="argN" />
145 </argument>
146 </option>
147 <option name="C">
148 <para>Reset the call detail record (CDR) for this call.</para>
149 </option>
150 <option name="c">
151 <para>If the Dial() application cancels this call, always set
152 <variable>HANGUPCAUSE</variable> to 'answered elsewhere'</para>
153 </option>
154 <option name="d">
155 <para>Allow the calling user to dial a 1 digit extension while waiting for
156 a call to be answered. Exit to that extension if it exists in the
157 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
158 if it exists.</para>
159 <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
160 connected. If you wish to use this option with these phones, you
161 can use the <literal>Answer</literal> application before dialing.</para>
162 </option>
163 <option name="D" argsep=":">
164 <argument name="called" />
165 <argument name="calling" />
166 <argument name="progress" />
167 <argument name="mfprogress" />
168 <argument name="mfwink" />
169 <argument name="sfprogress" />
170 <argument name="sfwink" />
171 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
172 party has answered, but before the call gets bridged. The
173 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
174 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
175 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
176 to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
177 <para>See <literal>SendDTMF</literal> for valid digits.</para>
178 <para>If <replaceable>mfprogress</replaceable> is specified, its MF is sent
179 to the called party immediately after receiving a <literal>PROGRESS</literal> message.
180 If <replaceable>mfwink</replaceable> is specified, its MF is sent
181 to the called party immediately after receiving a <literal>WINK</literal> message.</para>
182 <para>See <literal>SendMF</literal> for valid digits.</para>
183 <para>If <replaceable>sfprogress</replaceable> is specified, its SF is sent
184 to the called party immediately after receiving a <literal>PROGRESS</literal> message.
185 If <replaceable>sfwink</replaceable> is specified, its SF is sent
186 to the called party immediately after receiving a <literal>WINK</literal> message.</para>
187 <para>See <literal>SendSF</literal> for valid digits.</para>
188 </option>
189 <option name="E">
190 <para>Enable echoing of sent MF or SF digits back to caller (e.g. "hearpulsing").
191 Used in conjunction with the D option.</para>
192 </option>
193 <option name="e">
194 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
195 </option>
196 <option name="f">
197 <argument name="x" required="false" />
198 <para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
199 deflection to the dialplan extension of this <literal>Dial()</literal> using a dialplan <literal>hint</literal>.
200 For example, some PSTNs do not allow CallerID to be set to anything
201 other than the numbers assigned to you.
202 If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
203 </option>
204 <option name="F" argsep="^">
205 <argument name="context" required="false" />
206 <argument name="exten" required="false" />
207 <argument name="priority" required="true" />
208 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
209 to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
210 <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
211 prefixed with one or two underbars ('_').</para>
212 </option>
213 <option name="F">
214 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
215 and <emphasis>start</emphasis> execution at that location.</para>
216 <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
217 prefixed with one or two underbars ('_').</para>
218 <para>NOTE: Using this option from a Gosub() might not make sense as there would be no return points.</para>
219 </option>
220 <option name="g">
221 <para>Proceed with dialplan execution at the next priority in the current extension if the
222 destination channel hangs up.</para>
223 </option>
224 <option name="G" argsep="^">
225 <argument name="context" required="false" />
226 <argument name="exten" required="false" />
227 <argument name="priority" required="true" />
228 <para>If the call is answered, transfer the calling party to
229 the specified <replaceable>priority</replaceable> and the called party to the specified
230 <replaceable>priority</replaceable> plus one.</para>
231 <para>NOTE: You cannot use any additional action post answer options in conjunction with this option.</para>
232 </option>
233 <option name="h">
234 <para>Allow the called party to hang up by sending the DTMF sequence
235 defined for disconnect in <filename>features.conf</filename>.</para>
236 </option>
237 <option name="H">
238 <para>Allow the calling party to hang up by sending the DTMF sequence
239 defined for disconnect in <filename>features.conf</filename>.</para>
240 <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
241 connected. If you wish to allow DTMF disconnect before the dialed
242 party answers with these phones, you can use the <literal>Answer</literal>
243 application before dialing.</para>
244 </option>
245 <option name="i">
246 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
247 </option>
248 <option name="I">
249 <para>Asterisk will ignore any connected line update requests or any redirecting party
250 update requests it may receive on this dial attempt.</para>
251 </option>
252 <option name="j">
253 <para>Use the initial stream topology of the caller for outgoing channels, even if the caller topology has changed.</para>
254 <para>NOTE: For this option to work, it has to be present in all invocations of Dial that the caller channel goes through.</para>
255 </option>
256 <option name="k">
257 <para>Allow the called party to enable parking of the call by sending
258 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
259 </option>
260 <option name="K">
261 <para>Allow the calling party to enable parking of the call by sending
262 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
263 </option>
264 <option name="L" argsep=":">
265 <argument name="x" required="true">
266 <para>Maximum call time, in milliseconds</para>
267 </argument>
268 <argument name="y">
269 <para>Warning time, in milliseconds</para>
270 </argument>
271 <argument name="z">
272 <para>Repeat time, in milliseconds</para>
273 </argument>
274 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
275 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
276 <para>This option is affected by the following variables:</para>
277 <variablelist>
278 <variable name="LIMIT_PLAYAUDIO_CALLER">
279 <value name="yes" default="true" />
280 <value name="no" />
281 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
282 </variable>
283 <variable name="LIMIT_PLAYAUDIO_CALLEE">
284 <value name="yes" />
285 <value name="no" default="true"/>
286 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
287 </variable>
288 <variable name="LIMIT_TIMEOUT_FILE">
289 <value name="filename"/>
290 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
291 If not set, the time remaining will be announced.</para>
292 </variable>
293 <variable name="LIMIT_CONNECT_FILE">
294 <value name="filename"/>
295 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
296 If not set, the time remaining will be announced.</para>
297 </variable>
298 <variable name="LIMIT_WARNING_FILE">
299 <value name="filename"/>
300 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
301 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
302 </variable>
303 </variablelist>
304 </option>
305 <option name="m">
306 <argument name="class" required="false"/>
307 <para>Provide hold music to the calling party until a requested
308 channel answers. A specific music on hold <replaceable>class</replaceable>
309 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
310 </option>
311 <option name="n">
312 <argument name="delete">
313 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
314 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
315 yet answered.</para>
316 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
317 always be deleted.</para>
318 </argument>
319 <para>This option is a modifier for the call screening/privacy mode. (See the
320 <literal>p</literal> and <literal>P</literal> options.) It specifies
321 that no introductions are to be saved in the <directory>priv-callerintros</directory>
322 directory.</para>
323 </option>
324 <option name="N">
325 <para>This option is a modifier for the call screening/privacy mode. It specifies
326 that if CallerID is present, do not screen the call.</para>
327 </option>
328 <option name="o">
329 <argument name="x" required="false" />
330 <para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
331 <emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
332 This was the behavior of Asterisk 1.0 and earlier.
333 If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
334 Note that <literal>o(${CALLERID(all)})</literal> is similar to option <literal>o</literal> without the parameter.</para>
335 </option>
336 <option name="O">
337 <argument name="mode">
338 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
339 the originator hanging up will cause the phone to ring back immediately.</para>
340 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
341 flashes the trunk, it will ring their phone back.</para>
342 </argument>
343 <para>Enables <emphasis>operator services</emphasis> mode. This option only
344 works when bridging a DAHDI channel to another DAHDI channel
345 only. If specified on non-DAHDI interfaces, it will be ignored.
346 When the destination answers (presumably an operator services
347 station), the originator no longer has control of their line.
348 They may hang up, but the switch will not release their line
349 until the destination party (the operator) hangs up.</para>
350 </option>
351 <option name="p">
352 <para>This option enables screening mode. This is basically Privacy mode
353 without memory.</para>
354 </option>
355 <option name="P">
356 <argument name="x" />
357 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
358 it is provided. The current extension is used if a database family/key is not specified.</para>
359 </option>
360 <option name="Q">
361 <argument name="cause" required="true"/>
362 <para>Specify the Q.850/Q.931 <replaceable>cause</replaceable> to send on
363 unanswered channels when another channel answers the call.
364 As with <literal>Hangup()</literal>, <replaceable>cause</replaceable>
365 can be a numeric cause code or a name such as
366 <literal>NO_ANSWER</literal>,
367 <literal>USER_BUSY</literal>,
368 <literal>CALL_REJECTED</literal> or
369 <literal>ANSWERED_ELSEWHERE</literal> (the default if Q isn't specified).
370 You can also specify <literal>0</literal> or <literal>NONE</literal>
371 to send no cause. See the <filename>causes.h</filename> file for the
372 full list of valid causes and names.
373 </para>
374 </option>
375 <option name="r">
376 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
377 party until the called channel has answered.</para>
378 <argument name="tone" required="false">
379 <para>Indicate progress to calling party. Send audio 'tone' from the <filename>indications.conf</filename> tonezone currently in use.</para>
380 </argument>
381 </option>
382 <option name="R">
383 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
384 Allow interruption of the ringback if early media is received on the channel.</para>
385 </option>
386 <option name="S">
387 <argument name="x" required="true" />
388 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
389 answered the call.</para>
390 </option>
391 <option name="s">
392 <argument name="x" required="true" />
393 <para>Force the outgoing CallerID tag parameter to be set to the string <replaceable>x</replaceable>.</para>
394 <para>Works with the <literal>f</literal> option.</para>
395 </option>
396 <option name="t">
397 <para>Allow the called party to transfer the calling party by sending the
398 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
399 transfers initiated by other methods.</para>
400 </option>
401 <option name="T">
402 <para>Allow the calling party to transfer the called party by sending the
403 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
404 transfers initiated by other methods.</para>
405 </option>
406 <option name="U" argsep="^">
407 <argument name="x" required="true">
408 <para>Name of the subroutine context to execute via <literal>Gosub</literal>.
409 The subroutine execution starts in the named context at the s exten and priority 1.</para>
410 </argument>
411 <argument name="arg" multiple="true" required="false">
412 <para>Arguments for the <literal>Gosub</literal> routine</para>
413 </argument>
414 <para>Execute via <literal>Gosub</literal> the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
415 to the calling channel. Arguments can be specified to the <literal>Gosub</literal>
416 using <literal>^</literal> as a delimiter. The <literal>Gosub</literal> routine can set the variable
417 <variable>GOSUB_RESULT</variable> to specify the following actions after the <literal>Gosub</literal> returns.</para>
418 <variablelist>
419 <variable name="GOSUB_RESULT">
420 <value name="ABORT">
421 Hangup both legs of the call.
422 </value>
423 <value name="CONGESTION">
424 Behave as if line congestion was encountered.
425 </value>
426 <value name="BUSY">
427 Behave as if a busy signal was encountered.
428 </value>
429 <value name="CONTINUE">
430 Hangup the called party and allow the calling party
431 to continue dialplan execution at the next priority.
432 </value>
433 <value name="GOTO:[[&lt;context&gt;^]&lt;exten&gt;^]&lt;priority&gt;">
434 Transfer the call to the specified destination.
435 </value>
436 </variable>
437 </variablelist>
438 <para>NOTE: You cannot use any additional action post answer options in conjunction
439 with this option. Also, pbx services are run on the <emphasis>called</emphasis> channel,
440 so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
441 </option>
442 <option name="u">
443 <argument name = "x" required="true">
444 <para>Force the outgoing callerid presentation indicator parameter to be set
445 to one of the values passed in <replaceable>x</replaceable>:
446 <literal>allowed_not_screened</literal>
447 <literal>allowed_passed_screen</literal>
448 <literal>allowed_failed_screen</literal>
449 <literal>allowed</literal>
450 <literal>prohib_not_screened</literal>
451 <literal>prohib_passed_screen</literal>
452 <literal>prohib_failed_screen</literal>
453 <literal>prohib</literal>
454 <literal>unavailable</literal></para>
455 </argument>
456 <para>Works with the <literal>f</literal> option.</para>
457 </option>
458 <option name="w">
459 <para>Allow the called party to enable recording of the call by sending
460 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
461 </option>
462 <option name="W">
463 <para>Allow the calling party to enable recording of the call by sending
464 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
465 </option>
466 <option name="x">
467 <para>Allow the called party to enable recording of the call by sending
468 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
469 </option>
470 <option name="X">
471 <para>Allow the calling party to enable recording of the call by sending
472 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
473 </option>
474 <option name="z">
475 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
476 </option>
477 </optionlist>
478 </parameter>
479 <parameter name="URL">
480 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
481 </parameter>
482 </syntax>
483 <description>
484 <para>This application will place calls to one or more specified channels. As soon
485 as one of the requested channels answers, the originating channel will be
486 answered, if it has not already been answered. These two channels will then
487 be active in a bridged call. All other channels that were requested will then
488 be hung up.</para>
489 <para>Unless there is a timeout specified, the Dial application will wait
490 indefinitely until one of the called channels answers, the user hangs up, or
491 if all of the called channels are busy or unavailable. Dialplan execution will
492 continue if no requested channels can be called, or if the timeout expires.
493 This application will report normal termination if the originating channel
494 hangs up, or if the call is bridged and either of the parties in the bridge
495 ends the call.</para>
496 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
497 application will be put into that group (as in <literal>Set(GROUP()=...</literal>).
498 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
499 application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
500 however, the variable will be unset after use.</para>
501 <example title="Dial with 30 second timeout">
502 same => n,Dial(PJSIP/alice,30)
503 </example>
504 <example title="Parallel dial with 45 second timeout">
505 same => n,Dial(PJSIP/alice&amp;PJIP/bob,45)
506 </example>
507 <example title="Dial with 'g' continuation option">
508 same => n,Dial(PJSIP/alice,,g)
509 same => n,Log(NOTICE, Alice call result: ${DIALSTATUS})
510 </example>
511 <example title="Dial with transfer/recording features for calling party">
512 same => n,Dial(PJSIP/alice,,TX)
513 </example>
514 <example title="Dial with call length limit">
515 same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
516 </example>
517 <example title="Dial alice and bob and send NO_ANSWER to bob instead of ANSWERED_ELSEWHERE when alice answers">
518 same => n,Dial(PJSIP/alice&amp;PJSIP/bob,,Q(NO_ANSWER))
519 </example>
520 <example title="Dial with pre-dial subroutines">
521 [default]
522 exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
523 same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
524 same => n,Return()
525 exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
526 same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
527 same => n,Return()
528 exten => _X.,1,NoOp()
529 same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
530 same => n,Hangup()
531 </example>
532 <example title="Dial with post-answer subroutine executed on outbound channel">
533 [my_gosub_routine]
534 exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
535 same => n,Playback(hello)
536 same => n,Return()
537 [default]
538 exten => _X.,1,NoOp()
539 same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
540 same => n,Hangup()
541 </example>
542 <example title="Dial into ConfBridge using 'G' option">
543 same => n,Dial(PJSIP/alice,,G(jump_to_here))
544 same => n(jump_to_here),Goto(confbridge)
545 same => n,Goto(confbridge)
546 same => n(confbridge),ConfBridge(${EXTEN})
547 </example>
548 <para>This application sets the following channel variables:</para>
549 <variablelist>
550 <variable name="DIALEDTIME">
551 <para>This is the time from dialing a channel until when it is disconnected.</para>
552 </variable>
553 <variable name="DIALEDTIME_MS">
554 <para>This is the milliseconds version of the DIALEDTIME variable.</para>
555 </variable>
556 <variable name="ANSWEREDTIME">
557 <para>This is the amount of time for actual call.</para>
558 </variable>
559 <variable name="ANSWEREDTIME_MS">
560 <para>This is the milliseconds version of the ANSWEREDTIME variable.</para>
561 </variable>
562 <variable name="RINGTIME">
563 <para>This is the time from creating the channel to the first RINGING event received. Empty if there was no ring.</para>
564 </variable>
565 <variable name="RINGTIME_MS">
566 <para>This is the milliseconds version of the RINGTIME variable.</para>
567 </variable>
568 <variable name="PROGRESSTIME">
569 <para>This is the time from creating the channel to the first PROGRESS event received. Empty if there was no such event.</para>
570 </variable>
571 <variable name="PROGRESSTIME_MS">
572 <para>This is the milliseconds version of the PROGRESSTIME variable.</para>
573 </variable>
574 <variable name="DIALEDPEERNAME">
575 <para>The name of the outbound channel that answered the call.</para>
576 </variable>
577 <variable name="DIALEDPEERNUMBER">
578 <para>The number that was dialed for the answered outbound channel.</para>
579 </variable>
580 <variable name="FORWARDERNAME">
581 <para>If a call forward occurred, the name of the forwarded channel.</para>
582 </variable>
583 <variable name="DIALSTATUS">
584 <para>This is the status of the call</para>
585 <value name="CHANUNAVAIL">
586 Either the dialed peer exists but is not currently reachable, e.g.
587 endpoint is not registered, or an attempt was made to call a
588 nonexistent location, e.g. nonexistent DNS hostname.
589 </value>
590 <value name="CONGESTION">
591 Channel or switching congestion occurred when routing the call.
592 This can occur if there is a slow or no response from the remote end.
593 </value>
594 <value name="NOANSWER">
595 Called party did not answer.
596 </value>
597 <value name="BUSY">
598 The called party was busy or indicated a busy status.
599 Note that some SIP devices will respond with 486 Busy if their Do Not Disturb
600 modes are active. In this case, you can use DEVICE_STATUS to check if the
601 endpoint is actually in use, if needed.
602 </value>
603 <value name="ANSWER">
604 The call was answered.
605 Any other result implicitly indicates the call was not answered.
606 </value>
607 <value name="CANCEL">
608 Dial was cancelled before call was answered or reached some other terminating event.
609 </value>
610 <value name="DONTCALL">
611 For the Privacy and Screening Modes.
612 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
613 </value>
614 <value name="TORTURE">
615 For the Privacy and Screening Modes.
616 Will be set if the called party chooses to send the calling party to the 'torture' script.
617 </value>
618 <value name="INVALIDARGS">
619 Dial failed due to invalid syntax.
620 </value>
621 </variable>
622 </variablelist>
623 </description>
624 <see-also>
625 <ref type="application">RetryDial</ref>
626 <ref type="application">SendDTMF</ref>
627 <ref type="application">Gosub</ref>
628 </see-also>
629 </application>
630 <application name="RetryDial" language="en_US">
631 <since>
632 <version>1.2.0</version>
633 </since>
634 <synopsis>
635 Place a call, retrying on failure allowing an optional exit extension.
636 </synopsis>
637 <syntax>
638 <parameter name="announce" required="true">
639 <para>Filename of sound that will be played when no channel can be reached</para>
640 </parameter>
641 <parameter name="sleep" required="true">
642 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
643 </parameter>
644 <parameter name="retries" required="true">
645 <para>Number of retries</para>
646 <para>When this is reached flow will continue at the next priority in the dialplan</para>
647 </parameter>
648 <parameter name="dialargs" required="true">
649 <para>Same format as arguments provided to the Dial application</para>
650 </parameter>
651 </syntax>
652 <description>
653 <para>This application will attempt to place a call using the normal Dial application.
654 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
655 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
656 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
657 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
658 While waiting to retry a call, a 1 digit extension may be dialed. If that
659 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
660 one, The call will jump to that extension immediately.
661 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
662 to the Dial application.</para>
663 </description>
664 <see-also>
665 <ref type="application">Dial</ref>
666 </see-also>
667 </application>
668 ***/
669
670static const char app[] = "Dial";
671static const char rapp[] = "RetryDial";
672
673enum {
674 OPT_ANNOUNCE = (1 << 0),
675 OPT_RESETCDR = (1 << 1),
676 OPT_DTMF_EXIT = (1 << 2),
677 OPT_SENDDTMF = (1 << 3),
678 OPT_FORCECLID = (1 << 4),
679 OPT_GO_ON = (1 << 5),
680 OPT_CALLEE_HANGUP = (1 << 6),
681 OPT_CALLER_HANGUP = (1 << 7),
682 OPT_ORIGINAL_CLID = (1 << 8),
683 OPT_DURATION_LIMIT = (1 << 9),
684 OPT_MUSICBACK = (1 << 10),
685 OPT_SCREEN_NOINTRO = (1 << 12),
686 OPT_SCREEN_NOCALLERID = (1 << 13),
687 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
688 OPT_SCREENING = (1 << 15),
689 OPT_PRIVACY = (1 << 16),
690 OPT_RINGBACK = (1 << 17),
691 OPT_DURATION_STOP = (1 << 18),
692 OPT_CALLEE_TRANSFER = (1 << 19),
693 OPT_CALLER_TRANSFER = (1 << 20),
694 OPT_CALLEE_MONITOR = (1 << 21),
695 OPT_CALLER_MONITOR = (1 << 22),
696 OPT_GOTO = (1 << 23),
697 OPT_OPERMODE = (1 << 24),
698 OPT_CALLEE_PARK = (1 << 25),
699 OPT_CALLER_PARK = (1 << 26),
700 OPT_IGNORE_FORWARDING = (1 << 27),
701 OPT_CALLEE_GOSUB = (1 << 28),
702 OPT_CALLEE_MIXMONITOR = (1 << 29),
703 OPT_CALLER_MIXMONITOR = (1 << 30),
704};
705
706/* flags are now 64 bits, so keep it up! */
707#define DIAL_STILLGOING (1LLU << 31)
708#define DIAL_NOFORWARDHTML (1LLU << 32)
709#define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */
710#define OPT_CANCEL_ELSEWHERE (1LLU << 34)
711#define OPT_PEER_H (1LLU << 35)
712#define OPT_CALLEE_GO_ON (1LLU << 36)
713#define OPT_CANCEL_TIMEOUT (1LLU << 37)
714#define OPT_FORCE_CID_TAG (1LLU << 38)
715#define OPT_FORCE_CID_PRES (1LLU << 39)
716#define OPT_CALLER_ANSWER (1LLU << 40)
717#define OPT_PREDIAL_CALLEE (1LLU << 41)
718#define OPT_PREDIAL_CALLER (1LLU << 42)
719#define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
720#define OPT_HANGUPCAUSE (1LLU << 44)
721#define OPT_HEARPULSING (1LLU << 45)
722#define OPT_TOPOLOGY_PRESERVE (1LLU << 46)
723
724enum {
725 OPT_ARG_ANNOUNCE = 0,
726 OPT_ARG_SENDDTMF,
727 OPT_ARG_GOTO,
728 OPT_ARG_DURATION_LIMIT,
729 OPT_ARG_MUSICBACK,
730 OPT_ARG_RINGBACK,
731 OPT_ARG_CALLEE_GOSUB,
732 OPT_ARG_CALLEE_GO_ON,
733 OPT_ARG_PRIVACY,
734 OPT_ARG_DURATION_STOP,
735 OPT_ARG_OPERMODE,
736 OPT_ARG_SCREEN_NOINTRO,
737 OPT_ARG_ORIGINAL_CLID,
738 OPT_ARG_FORCECLID,
739 OPT_ARG_FORCE_CID_TAG,
740 OPT_ARG_FORCE_CID_PRES,
741 OPT_ARG_PREDIAL_CALLEE,
742 OPT_ARG_PREDIAL_CALLER,
743 OPT_ARG_HANGUPCAUSE,
744 /* note: this entry _MUST_ be the last one in the enum */
745 OPT_ARG_ARRAY_SIZE
746};
747
748AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
749 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
750 AST_APP_OPTION('a', OPT_CALLER_ANSWER),
751 AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
752 AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
753 AST_APP_OPTION('C', OPT_RESETCDR),
754 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
755 AST_APP_OPTION('d', OPT_DTMF_EXIT),
756 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
757 AST_APP_OPTION('E', OPT_HEARPULSING),
758 AST_APP_OPTION('e', OPT_PEER_H),
759 AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
760 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
761 AST_APP_OPTION('g', OPT_GO_ON),
762 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
763 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
764 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
765 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
766 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
767 AST_APP_OPTION('j', OPT_TOPOLOGY_PRESERVE),
768 AST_APP_OPTION('k', OPT_CALLEE_PARK),
769 AST_APP_OPTION('K', OPT_CALLER_PARK),
770 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
771 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
772 AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
773 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
774 AST_APP_OPTION_ARG('o', OPT_ORIGINAL_CLID, OPT_ARG_ORIGINAL_CLID),
775 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
776 AST_APP_OPTION('p', OPT_SCREENING),
777 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
778 AST_APP_OPTION_ARG('Q', OPT_HANGUPCAUSE, OPT_ARG_HANGUPCAUSE),
779 AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
780 AST_APP_OPTION('R', OPT_RING_WITH_EARLY_MEDIA),
781 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
782 AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
783 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
784 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
785 AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
786 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
787 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
788 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
789 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
790 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
791 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
792END_OPTIONS );
793
794#define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
795 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
796 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
797 OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_GOSUB) && \
798 !ast_channel_audiohooks(chan) && !ast_channel_audiohooks(peer) && \
799 ast_framehook_list_is_empty(ast_channel_framehooks(chan)) && ast_framehook_list_is_empty(ast_channel_framehooks(peer)))
800
801/*
802 * The list of active channels
803 */
804struct chanlist {
805 AST_LIST_ENTRY(chanlist) node;
806 struct ast_channel *chan;
807 /*! Channel interface dialing string (is tech/number). (Stored in stuff[]) */
808 const char *interface;
809 /*! Channel technology name. (Stored in stuff[]) */
810 const char *tech;
811 /*! Channel device addressing. (Stored in stuff[]) */
812 const char *number;
813 /*! Original channel name. Must be freed. Could be NULL if allocation failed. */
814 char *orig_chan_name;
815 uint64_t flags;
816 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
817 struct ast_party_connected_line connected;
818 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
819 unsigned int pending_connected_update:1;
820 struct ast_aoc_decoded *aoc_s_rate_list;
821 /*! The interface, tech, and number strings are stuffed here. */
822 char stuff[0];
823};
824
825AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
826
827static void topology_ds_destroy(void *data) {
828 struct ast_stream_topology *top = data;
829 ast_stream_topology_free(top);
830}
831
832static const struct ast_datastore_info topology_ds_info = {
833 .type = "app_dial_topology_preserve",
834 .destroy = topology_ds_destroy,
835};
836
837static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
838
839static void chanlist_free(struct chanlist *outgoing)
840{
841 ast_party_connected_line_free(&outgoing->connected);
842 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
843 ast_free(outgoing->orig_chan_name);
844 ast_free(outgoing);
845}
846
847static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
848{
849 /* Hang up a tree of stuff */
850 struct chanlist *outgoing;
851
852 while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
853 /* Hangup any existing lines we have open */
854 if (outgoing->chan && (outgoing->chan != exception)) {
855 if (hangupcause >= 0) {
856 /* This is for the channel drivers */
857 ast_channel_hangupcause_set(outgoing->chan, hangupcause);
858 }
859 ast_hangup(outgoing->chan);
860 }
861 chanlist_free(outgoing);
862 }
863}
864
865#define AST_MAX_WATCHERS 256
866
867/*
868 * argument to handle_cause() and other functions.
869 */
870struct cause_args {
871 struct ast_channel *chan;
872 int busy;
873 int congestion;
874 int nochan;
875};
876
877static void handle_cause(int cause, struct cause_args *num)
878{
879 switch(cause) {
880 case AST_CAUSE_BUSY:
881 num->busy++;
882 break;
883 case AST_CAUSE_CONGESTION:
884 num->congestion++;
885 break;
886 case AST_CAUSE_NO_ROUTE_DESTINATION:
887 case AST_CAUSE_UNREGISTERED:
888 num->nochan++;
889 break;
890 case AST_CAUSE_NO_ANSWER:
891 case AST_CAUSE_NORMAL_CLEARING:
892 break;
893 default:
894 num->nochan++;
895 break;
896 }
897}
898
899static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
900{
901 char rexten[2] = { exten, '\0' };
902
903 if (context) {
904 if (!ast_goto_if_exists(chan, context, rexten, pri))
905 return 1;
906 } else {
907 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
908 return 1;
909 }
910 return 0;
911}
912
913/* do not call with chan lock held */
914static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
915{
916 const char *context;
917 const char *exten;
918
919 ast_channel_lock(chan);
920 context = ast_strdupa(ast_channel_context(chan));
921 exten = ast_strdupa(ast_channel_exten(chan));
922 ast_channel_unlock(chan);
923
924 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
925}
926
927/*!
928 * helper function for wait_for_answer()
929 *
930 * \param o Outgoing call channel list.
931 * \param num Incoming call channel cause accumulation
932 * \param peerflags Dial option flags
933 * \param single TRUE if there is only one outgoing call.
934 * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
935 * \param to Remaining call timeout time.
936 * \param forced_clid OPT_FORCECLID caller id to send
937 * \param stored_clid Caller id representing the called party if needed
938 *
939 * XXX this code is highly suspicious, as it essentially overwrites
940 * the outgoing channel without properly deleting it.
941 *
942 * \todo eventually this function should be integrated into and replaced by ast_call_forward()
943 */
944static void do_forward(struct chanlist *o, struct cause_args *num,
945 struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
946 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
947{
948 char tmpchan[256];
949 char forwarder[AST_CHANNEL_NAME];
950 struct ast_channel *original = o->chan;
951 struct ast_channel *c = o->chan; /* the winner */
952 struct ast_channel *in = num->chan; /* the input channel */
953 char *stuff;
954 const char *tech;
955 int cause;
956 struct ast_party_caller caller;
957
958 ast_copy_string(forwarder, ast_channel_name(c), sizeof(forwarder));
959 ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
960 if ((stuff = strchr(tmpchan, '/'))) {
961 *stuff++ = '\0';
962 tech = tmpchan;
963 } else {
964 const char *forward_context;
965 ast_channel_lock(c);
966 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
967 if (ast_strlen_zero(forward_context)) {
968 forward_context = NULL;
969 }
970 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
971 ast_channel_unlock(c);
972 stuff = tmpchan;
973 tech = "Local";
974 }
975 if (!strcasecmp(tech, "Local")) {
976 /*
977 * Drop the connected line update block for local channels since
978 * this is going to run dialplan and the user can change his
979 * mind about what connected line information he wants to send.
980 */
981 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
982 }
983
984 /* Before processing channel, go ahead and check for forwarding */
985 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
986 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
987 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
988 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
989 ast_channel_publish_dial_forward(in, original, NULL, NULL, "CANCEL",
990 ast_channel_call_forward(original));
991 c = o->chan = NULL;
992 cause = AST_CAUSE_BUSY;
993 } else {
994 struct ast_stream_topology *topology;
995
996 ast_channel_lock(in);
997 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(in));
998 ast_channel_unlock(in);
999
1000 /* Setup parameters */
1001 c = o->chan = ast_request_with_stream_topology(tech, topology, NULL, in, stuff, &cause);
1002
1003 ast_stream_topology_free(topology);
1004
1005 if (c) {
1006 if (single && !caller_entertained) {
1007 ast_channel_make_compatible(in, o->chan);
1008 }
1009 ast_channel_lock_both(in, o->chan);
1010 ast_channel_inherit_variables(in, o->chan);
1011 ast_channel_datastore_inherit(in, o->chan);
1012 pbx_builtin_setvar_helper(o->chan, "FORWARDERNAME", forwarder);
1013 ast_max_forwards_decrement(o->chan);
1014 ast_channel_unlock(in);
1015 ast_channel_unlock(o->chan);
1016 /* When a call is forwarded, we don't want to track new interfaces
1017 * dialed for CC purposes. Setting the done flag will ensure that
1018 * any Dial operations that happen later won't record CC interfaces.
1019 */
1020 ast_ignore_cc(o->chan);
1021 ast_verb(3, "Not accepting call completion offers from call-forward recipient %s\n",
1022 ast_channel_name(o->chan));
1023 } else
1024 ast_log(LOG_NOTICE,
1025 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
1026 tech, stuff, cause);
1027 }
1028 if (!c) {
1029 ast_channel_publish_dial(in, original, stuff, "BUSY");
1030 ast_clear_flag64(o, DIAL_STILLGOING);
1031 handle_cause(cause, num);
1032 ast_hangup(original);
1033 } else {
1034 ast_channel_lock_both(c, original);
1035 ast_party_redirecting_copy(ast_channel_redirecting(c),
1036 ast_channel_redirecting(original));
1037 ast_channel_unlock(c);
1038 ast_channel_unlock(original);
1039
1040 ast_channel_lock_both(c, in);
1041
1042 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
1043 ast_rtp_instance_early_bridge_make_compatible(c, in);
1044 }
1045
1046 if (!ast_channel_redirecting(c)->from.number.valid
1047 || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
1048 /*
1049 * The call was not previously redirected so it is
1050 * now redirected from this number.
1051 */
1052 ast_party_number_free(&ast_channel_redirecting(c)->from.number);
1053 ast_party_number_init(&ast_channel_redirecting(c)->from.number);
1054 ast_channel_redirecting(c)->from.number.valid = 1;
1055 ast_channel_redirecting(c)->from.number.str =
1056 ast_strdup(ast_channel_exten(in));
1057 }
1058
1059 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
1060
1061 /* Determine CallerID to store in outgoing channel. */
1062 ast_party_caller_set_init(&caller, ast_channel_caller(c));
1063 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1064 caller.id = *stored_clid;
1065 ast_channel_set_caller_event(c, &caller, NULL);
1066 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1067 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
1068 ast_channel_caller(c)->id.number.str, NULL))) {
1069 /*
1070 * The new channel has no preset CallerID number by the channel
1071 * driver. Use the dialplan extension and hint name.
1072 */
1073 caller.id = *stored_clid;
1074 ast_channel_set_caller_event(c, &caller, NULL);
1075 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1076 } else {
1077 ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
1078 }
1079
1080 /* Determine CallerID for outgoing channel to send. */
1081 if (ast_test_flag64(o, OPT_FORCECLID)) {
1082 struct ast_party_connected_line connected;
1083
1084 ast_party_connected_line_init(&connected);
1085 connected.id = *forced_clid;
1086 ast_party_connected_line_copy(ast_channel_connected(c), &connected);
1087 } else {
1088 ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
1089 }
1090
1091 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
1092
1093 ast_channel_appl_set(c, "AppDial");
1094 ast_channel_data_set(c, "(Outgoing Line)");
1095 ast_channel_publish_snapshot(c);
1096
1097 ast_channel_unlock(in);
1098 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1099 struct ast_party_redirecting redirecting;
1100
1101 /*
1102 * Redirecting updates to the caller make sense only on single
1103 * calls.
1104 *
1105 * Need to re-evalute if unlocking is still required here as macro is gone
1106 */
1107 ast_party_redirecting_init(&redirecting);
1108 ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
1109 ast_channel_unlock(c);
1110 if (ast_channel_redirecting_sub(c, in, &redirecting, 0)) {
1111 ast_channel_update_redirecting(in, &redirecting, NULL);
1112 }
1113 ast_party_redirecting_free(&redirecting);
1114 } else {
1115 ast_channel_unlock(c);
1116 }
1117
1118 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1119 *to = -1;
1120 }
1121
1122 if (ast_call(c, stuff, 0)) {
1123 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1124 tech, stuff);
1125 ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1126 ast_clear_flag64(o, DIAL_STILLGOING);
1127 ast_hangup(original);
1128 ast_hangup(c);
1129 c = o->chan = NULL;
1130 num->nochan++;
1131 } else {
1132 ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1133 ast_channel_call_forward(original));
1134
1135 ast_channel_publish_dial(in, c, stuff, NULL);
1136
1137 /* Hangup the original channel now, in case we needed it */
1138 ast_hangup(original);
1139 }
1140 if (single && !caller_entertained) {
1141 ast_indicate(in, -1);
1142 }
1143 }
1144}
1145
1146/* argument used for some functions. */
1147struct privacy_args {
1148 int sentringing;
1149 int privdb_val;
1150 char privcid[256];
1151 char privintro[1024];
1152 char status[256];
1153 int canceled;
1154};
1155
1156static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
1157{
1158 struct chanlist *outgoing;
1159 AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1160 if (!outgoing->chan || outgoing->chan == exception) {
1161 continue;
1162 }
1163 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1164 }
1165}
1166
1167/*!
1168 * \internal
1169 * \brief Update connected line on chan from peer.
1170 * \since 13.6.0
1171 *
1172 * \param chan Channel to get connected line updated.
1173 * \param peer Channel providing connected line information.
1174 * \param is_caller Non-zero if chan is the calling channel.
1175 */
1176static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
1177{
1178 struct ast_party_connected_line connected_caller;
1179
1180 ast_party_connected_line_init(&connected_caller);
1181
1182 ast_channel_lock(peer);
1183 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1184 ast_channel_unlock(peer);
1185 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1186 if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)) {
1187 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1188 }
1189 ast_party_connected_line_free(&connected_caller);
1190}
1191
1192/*!
1193 * \internal
1194 * \pre chan is locked
1195 */
1196static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
1197{
1198 char buf[32];
1199 char full_var_name[128];
1200
1201 snprintf(buf, sizeof(buf), "%" PRId64, duration / 1000);
1202 pbx_builtin_setvar_helper(chan, var_base, buf);
1203
1204 snprintf(full_var_name, sizeof(full_var_name), "%s_MS", var_base);
1205 snprintf(buf, sizeof(buf), "%" PRId64, duration);
1206 pbx_builtin_setvar_helper(chan, full_var_name, buf);
1207}
1208
1209static struct ast_channel *wait_for_answer(struct ast_channel *in,
1210 struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags,
1211 char *opt_args[],
1212 struct privacy_args *pa,
1213 const struct cause_args *num_in, int *result, char *dtmf_progress,
1214 char *mf_progress, char *mf_wink,
1215 char *sf_progress, char *sf_wink,
1216 const int hearpulsing,
1217 const int ignore_cc,
1218 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid,
1219 struct ast_bridge_config *config)
1220{
1221 struct cause_args num = *num_in;
1222 int prestart = num.busy + num.congestion + num.nochan;
1223 int orig_answer_to = *to_answer;
1224 int orig_progress_to = *to_progress;
1225 struct ast_channel *peer = NULL;
1226 struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1227 /* single is set if only one destination is enabled */
1228 int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1229 int caller_entertained = outgoing
1230 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1231 struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1232 int cc_recall_core_id;
1233 int is_cc_recall;
1234 int cc_frame_received = 0;
1235 int num_ringing = 0;
1236 int sent_ring = 0;
1237 int sent_progress = 0, sent_wink = 0;
1238 struct timeval start = ast_tvnow();
1239 SCOPE_ENTER(3, "%s\n", ast_channel_name(in));
1240
1241 if (single) {
1242 /* Turn off hold music, etc */
1243 if (!caller_entertained) {
1244 ast_deactivate_generator(in);
1245 /* If we are calling a single channel, and not providing ringback or music, */
1246 /* then, make them compatible for in-band tone purpose */
1247 if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1248 /* If these channels can not be made compatible,
1249 * there is no point in continuing. The bridge
1250 * will just fail if it gets that far.
1251 */
1252 *to_answer = -1;
1253 strcpy(pa->status, "CONGESTION");
1254 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1255 SCOPE_EXIT_RTN_VALUE(NULL, "%s: can't be made compat with %s\n",
1256 ast_channel_name(in), ast_channel_name(outgoing->chan));
1257 }
1258 }
1259
1260 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1261 && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1262 update_connected_line_from_peer(in, outgoing->chan, 1);
1263 }
1264 }
1265
1266 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1267
1268 while ((*to_answer = ast_remaining_ms(start, orig_answer_to)) && (*to_progress = ast_remaining_ms(start, orig_progress_to)) && !peer) {
1269 struct chanlist *o;
1270 int pos = 0; /* how many channels do we handle */
1271 int numlines = prestart;
1272 struct ast_channel *winner;
1273 struct ast_channel *watchers[AST_MAX_WATCHERS];
1274
1275 watchers[pos++] = in;
1276 AST_LIST_TRAVERSE(out_chans, o, node) {
1277 /* Keep track of important channels */
1278 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1279 watchers[pos++] = o->chan;
1280 numlines++;
1281 }
1282 if (pos == 1) { /* only the input channel is available */
1283 if (numlines == (num.busy + num.congestion + num.nochan)) {
1284 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1285 if (num.busy)
1286 strcpy(pa->status, "BUSY");
1287 else if (num.congestion)
1288 strcpy(pa->status, "CONGESTION");
1289 else if (num.nochan)
1290 strcpy(pa->status, "CHANUNAVAIL");
1291 } else {
1292 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1293 }
1294 *to_answer = 0;
1295 if (is_cc_recall) {
1296 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1297 }
1298 SCOPE_EXIT_RTN_VALUE(NULL, "%s: No outgoing channels available\n", ast_channel_name(in));
1299 }
1300
1301 /* If progress timeout is active, use that if it's the shorter of the 2 timeouts. */
1302 winner = ast_waitfor_n(watchers, pos, *to_progress > 0 && (*to_answer < 0 || *to_progress < *to_answer) ? to_progress : to_answer);
1303
1304 AST_LIST_TRAVERSE(out_chans, o, node) {
1305 int res = 0;
1306 struct ast_frame *f;
1307 struct ast_channel *c = o->chan;
1308
1309 if (c == NULL)
1310 continue;
1311 if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1312 if (!peer) {
1313 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1314 if (o->orig_chan_name
1315 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1316 /*
1317 * The channel name changed so we must generate COLP update.
1318 * Likely because a call pickup channel masqueraded in.
1319 */
1320 update_connected_line_from_peer(in, c, 1);
1321 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1322 if (o->pending_connected_update) {
1323 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1324 ast_channel_update_connected_line(in, &o->connected, NULL);
1325 }
1326 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1327 update_connected_line_from_peer(in, c, 1);
1328 }
1329 }
1330 if (o->aoc_s_rate_list) {
1331 size_t encoded_size;
1332 struct ast_aoc_encoded *encoded;
1333 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1334 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1335 ast_aoc_destroy_encoded(encoded);
1336 }
1337 }
1338 peer = c;
1339 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1340 ast_copy_flags64(peerflags, o,
1341 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1342 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1343 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1344 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1345 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1346 DIAL_NOFORWARDHTML);
1347 ast_channel_dialcontext_set(c, "");
1348 ast_channel_exten_set(c, "");
1349 }
1350 continue;
1351 }
1352 if (c != winner)
1353 continue;
1354 /* here, o->chan == c == winner */
1355 if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1356 pa->sentringing = 0;
1357 if (!ignore_cc && (f = ast_read(c))) {
1358 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1359 /* This channel is forwarding the call, and is capable of CC, so
1360 * be sure to add the new device interface to the list
1361 */
1362 ast_handle_cc_control_frame(in, c, f->data.ptr);
1363 }
1364 ast_frfree(f);
1365 }
1366
1367 if (o->pending_connected_update) {
1368 /*
1369 * Re-seed the chanlist's connected line information with
1370 * previously acquired connected line info from the incoming
1371 * channel. The previously acquired connected line info could
1372 * have been set through the CONNECTED_LINE dialplan function.
1373 */
1374 o->pending_connected_update = 0;
1375 ast_channel_lock(in);
1376 ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1377 ast_channel_unlock(in);
1378 }
1379
1380 do_forward(o, &num, peerflags, single, caller_entertained, &orig_answer_to,
1381 forced_clid, stored_clid);
1382
1383 if (o->chan) {
1384 ast_free(o->orig_chan_name);
1385 o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1386 if (single
1387 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1388 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1389 update_connected_line_from_peer(in, o->chan, 1);
1390 }
1391 }
1392 continue;
1393 }
1394 f = ast_read(winner);
1395 if (!f) {
1396 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1397 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1398 ast_hangup(c);
1399 c = o->chan = NULL;
1400 ast_clear_flag64(o, DIAL_STILLGOING);
1401 handle_cause(ast_channel_hangupcause(in), &num);
1402 continue;
1403 }
1404 switch (f->frametype) {
1405 case AST_FRAME_CONTROL:
1406 switch (f->subclass.integer) {
1407 case AST_CONTROL_ANSWER:
1408 /* This is our guy if someone answered. */
1409 if (!peer) {
1410 ast_trace(-1, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1411 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1412 if (o->orig_chan_name
1413 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1414 /*
1415 * The channel name changed so we must generate COLP update.
1416 * Likely because a call pickup channel masqueraded in.
1417 */
1418 update_connected_line_from_peer(in, c, 1);
1419 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1420 if (o->pending_connected_update) {
1421 if (ast_channel_connected_line_sub(c, in, &o->connected, 0)) {
1422 ast_channel_update_connected_line(in, &o->connected, NULL);
1423 }
1424 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1425 update_connected_line_from_peer(in, c, 1);
1426 }
1427 }
1428 if (o->aoc_s_rate_list) {
1429 size_t encoded_size;
1430 struct ast_aoc_encoded *encoded;
1431 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1432 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1433 ast_aoc_destroy_encoded(encoded);
1434 }
1435 }
1436 peer = c;
1437 /* Answer can optionally include a topology */
1438 if (f->subclass.topology) {
1439 /*
1440 * We need to bump the refcount on the topology to prevent it
1441 * from being cleaned up when the frame is cleaned up.
1442 */
1443 config->answer_topology = ao2_bump(f->subclass.topology);
1444 ast_trace(-1, "%s Found topology in frame: %p %p %s\n",
1445 ast_channel_name(peer), f, config->answer_topology,
1446 ast_str_tmp(256, ast_stream_topology_to_str(config->answer_topology, &STR_TMP)));
1447 }
1448
1449 /* Inform everyone else that they've been canceled.
1450 * The dial end event for the peer will be sent out after
1451 * other Dial options have been handled.
1452 */
1453 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1454 ast_copy_flags64(peerflags, o,
1455 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1456 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1457 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1458 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1459 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1460 DIAL_NOFORWARDHTML);
1461 ast_channel_dialcontext_set(c, "");
1462 ast_channel_exten_set(c, "");
1463 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1464 /* Setup early bridge if appropriate */
1465 ast_channel_early_bridge(in, peer);
1466 }
1467 }
1468 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1469 ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1470 ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1471 break;
1472 case AST_CONTROL_BUSY:
1473 ast_verb(3, "%s is busy\n", ast_channel_name(c));
1474 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1475 ast_channel_publish_dial(in, c, NULL, "BUSY");
1476 ast_hangup(c);
1477 c = o->chan = NULL;
1478 ast_clear_flag64(o, DIAL_STILLGOING);
1479 handle_cause(AST_CAUSE_BUSY, &num);
1480 break;
1481 case AST_CONTROL_CONGESTION:
1482 ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1483 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1484 ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1485 ast_hangup(c);
1486 c = o->chan = NULL;
1487 ast_clear_flag64(o, DIAL_STILLGOING);
1488 handle_cause(AST_CAUSE_CONGESTION, &num);
1489 break;
1490 case AST_CONTROL_RINGING:
1491 /* This is a tricky area to get right when using a native
1492 * CC agent. The reason is that we do the best we can to send only a
1493 * single ringing notification to the caller.
1494 *
1495 * Call completion complicates the logic used here. CCNR is typically
1496 * offered during a ringing message. Let's say that party A calls
1497 * parties B, C, and D. B and C do not support CC requests, but D
1498 * does. If we were to receive a ringing notification from B before
1499 * the others, then we would end up sending a ringing message to
1500 * A with no CCNR offer present.
1501 *
1502 * The approach that we have taken is that if we receive a ringing
1503 * response from a party and no CCNR offer is present, we need to
1504 * wait. Specifically, we need to wait until either a) a called party
1505 * offers CCNR in its ringing response or b) all called parties have
1506 * responded in some way to our call and none offers CCNR.
1507 *
1508 * The drawback to this is that if one of the parties has a delayed
1509 * response or, god forbid, one just plain doesn't respond to our
1510 * outgoing call, then this will result in a significant delay between
1511 * when the caller places the call and hears ringback.
1512 *
1513 * Note also that if CC is disabled for this call, then it is perfectly
1514 * fine for ringing frames to get sent through.
1515 */
1516 ++num_ringing;
1517 *to_progress = -1;
1518 orig_progress_to = -1;
1519 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1520 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1521 /* Setup early media if appropriate */
1522 if (single && !caller_entertained
1523 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1524 ast_channel_early_bridge(in, c);
1525 }
1526 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1527 ast_indicate(in, AST_CONTROL_RINGING);
1528 pa->sentringing++;
1529 }
1530 if (!sent_ring) {
1531 struct timeval now, then;
1532 int64_t diff;
1533
1534 now = ast_tvnow();
1535
1536 ast_channel_lock(in);
1537 ast_channel_stage_snapshot(in);
1538
1539 then = ast_channel_creationtime(c);
1540 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1541 set_duration_var(in, "RINGTIME", diff);
1542
1543 ast_channel_stage_snapshot_done(in);
1544 ast_channel_unlock(in);
1545 sent_ring = 1;
1546 }
1547 }
1548 ast_channel_publish_dial(in, c, NULL, "RINGING");
1549 break;
1550 case AST_CONTROL_PROGRESS:
1551 ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1552 /* Setup early media if appropriate */
1553 if (single && !caller_entertained
1554 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1555 ast_channel_early_bridge(in, c);
1556 }
1557 if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1558 if (single || (!single && !pa->sentringing)) {
1559 ast_indicate(in, AST_CONTROL_PROGRESS);
1560 }
1561 }
1562 *to_progress = -1;
1563 orig_progress_to = -1;
1564 if (!sent_progress) {
1565 struct timeval now, then;
1566 int64_t diff;
1567
1568 now = ast_tvnow();
1569
1570 ast_channel_lock(in);
1571 ast_channel_stage_snapshot(in);
1572
1573 then = ast_channel_creationtime(c);
1574 diff = ast_tvzero(then) ? 0 : ast_tvdiff_ms(now, then);
1575 set_duration_var(in, "PROGRESSTIME", diff);
1576
1577 ast_channel_stage_snapshot_done(in);
1578 ast_channel_unlock(in);
1579 sent_progress = 1;
1580
1581 if (!ast_strlen_zero(mf_progress)) {
1582 ast_verb(3,
1583 "Sending MF '%s' to %s as result of "
1584 "receiving a PROGRESS message.\n",
1585 mf_progress, hearpulsing ? "parties" : "called party");
1586 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1587 (hearpulsing ? in : NULL), mf_progress, 50, 55, 120, 65, 0);
1588 }
1589 if (!ast_strlen_zero(sf_progress)) {
1590 ast_verb(3,
1591 "Sending SF '%s' to %s as result of "
1592 "receiving a PROGRESS message.\n",
1593 sf_progress, (hearpulsing ? "parties" : "called party"));
1594 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1595 (hearpulsing ? in : NULL), sf_progress, 0, 0);
1596 }
1597 if (!ast_strlen_zero(dtmf_progress)) {
1598 ast_verb(3,
1599 "Sending DTMF '%s' to the called party as result of "
1600 "receiving a PROGRESS message.\n",
1601 dtmf_progress);
1602 res |= ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1603 }
1604 if (res) {
1605 ast_log(LOG_WARNING, "Called channel %s hung up post-progress before all digits could be sent\n", ast_channel_name(c));
1606 goto wait_over;
1607 }
1608 }
1609 ast_channel_publish_dial(in, c, NULL, "PROGRESS");
1610 break;
1611 case AST_CONTROL_WINK:
1612 ast_verb(3, "%s winked, passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1613 if (!sent_wink) {
1614 sent_wink = 1;
1615 if (!ast_strlen_zero(mf_wink)) {
1616 ast_verb(3,
1617 "Sending MF '%s' to %s as result of "
1618 "receiving a WINK message.\n",
1619 mf_wink, (hearpulsing ? "parties" : "called party"));
1620 res |= ast_mf_stream(c, (hearpulsing ? NULL : in),
1621 (hearpulsing ? in : NULL), mf_wink, 50, 55, 120, 65, 0);
1622 }
1623 if (!ast_strlen_zero(sf_wink)) {
1624 ast_verb(3,
1625 "Sending SF '%s' to %s as result of "
1626 "receiving a WINK message.\n",
1627 sf_wink, (hearpulsing ? "parties" : "called party"));
1628 res |= ast_sf_stream(c, (hearpulsing ? NULL : in),
1629 (hearpulsing ? in : NULL), sf_wink, 0, 0);
1630 }
1631 if (res) {
1632 ast_log(LOG_WARNING, "Called channel %s hung up post-wink before all digits could be sent\n", ast_channel_name(c));
1633 goto wait_over;
1634 }
1635 }
1636 ast_indicate(in, AST_CONTROL_WINK);
1637 break;
1638 case AST_CONTROL_VIDUPDATE:
1639 case AST_CONTROL_SRCUPDATE:
1640 case AST_CONTROL_SRCCHANGE:
1641 if (!single || caller_entertained) {
1642 break;
1643 }
1644 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1645 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1646 ast_indicate(in, f->subclass.integer);
1647 break;
1648 case AST_CONTROL_CONNECTED_LINE:
1649 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1650 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1651 break;
1652 }
1653 if (!single) {
1654 struct ast_party_connected_line connected;
1655
1656 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1657 ast_channel_name(c), ast_channel_name(in));
1658 ast_party_connected_line_set_init(&connected, &o->connected);
1659 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1660 ast_party_connected_line_set(&o->connected, &connected, NULL);
1661 ast_party_connected_line_free(&connected);
1662 o->pending_connected_update = 1;
1663 break;
1664 }
1665 if (ast_channel_connected_line_sub(c, in, f, 1)) {
1666 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1667 }
1668 break;
1669 case AST_CONTROL_AOC:
1670 {
1671 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1672 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1673 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1674 o->aoc_s_rate_list = decoded;
1675 } else {
1676 ast_aoc_destroy_decoded(decoded);
1677 }
1678 }
1679 break;
1680 case AST_CONTROL_REDIRECTING:
1681 if (!single) {
1682 /*
1683 * Redirecting updates to the caller make sense only on single
1684 * calls.
1685 */
1686 break;
1687 }
1688 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1689 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1690 break;
1691 }
1692 ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1693 ast_channel_name(c), ast_channel_name(in));
1694 if (ast_channel_redirecting_sub(c, in, f, 1)) {
1695 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1696 }
1697 pa->sentringing = 0;
1698 break;
1699 case AST_CONTROL_PROCEEDING:
1700 ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1701 if (single && !caller_entertained
1702 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1703 ast_channel_early_bridge(in, c);
1704 }
1705 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1706 ast_indicate(in, AST_CONTROL_PROCEEDING);
1707 ast_channel_publish_dial(in, c, NULL, "PROCEEDING");
1708 break;
1709 case AST_CONTROL_HOLD:
1710 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1711 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1712 ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1713 break;
1714 case AST_CONTROL_UNHOLD:
1715 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1716 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1717 ast_indicate(in, AST_CONTROL_UNHOLD);
1718 break;
1719 case AST_CONTROL_OFFHOOK:
1720 case AST_CONTROL_FLASH:
1721 /* Ignore going off hook and flash */
1722 break;
1723 case AST_CONTROL_CC:
1724 if (!ignore_cc) {
1725 ast_handle_cc_control_frame(in, c, f->data.ptr);
1726 cc_frame_received = 1;
1727 }
1728 break;
1729 case AST_CONTROL_PVT_CAUSE_CODE:
1730 ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1731 break;
1732 case AST_CONTROL_PLAYBACK_BEGIN:
1733 if (!f->data.ptr) {
1734 ast_log(LOG_WARNING, "Got playback begin directive without filename on %s\n", ast_channel_name(c));
1735 } else {
1736 const char *filename = f->data.ptr;
1737 ast_verb(3, "Playing audio file %s on %s\n", filename, ast_channel_name(in));
1738 ast_streamfile(in, filename, ast_channel_language(in));
1739 }
1740 break;
1741 case -1:
1742 if (single && !caller_entertained) {
1743 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1744 ast_indicate(in, -1);
1745 pa->sentringing = 0;
1746 }
1747 break;
1748 default:
1749 ast_debug(1, "Dunno what to do with control type %d on %s\n", f->subclass.integer, ast_channel_name(in));
1750 break;
1751 }
1752 break;
1753 case AST_FRAME_VIDEO:
1754 case AST_FRAME_VOICE:
1755 case AST_FRAME_IMAGE:
1756 case AST_FRAME_DTMF_BEGIN:
1757 case AST_FRAME_DTMF_END:
1758 if (caller_entertained) {
1759 break;
1760 }
1761 *to_progress = -1;
1762 orig_progress_to = -1;
1763 /* Fall through */
1764 case AST_FRAME_TEXT:
1765 if (single && ast_write(in, f)) {
1766 ast_log(LOG_WARNING, "Unable to write frametype %u on %s\n",
1767 f->frametype, ast_channel_name(in));
1768 }
1769 break;
1770 case AST_FRAME_HTML:
1771 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1772 && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1773 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(in));
1774 }
1775 break;
1776 default:
1777 break;
1778 }
1779 ast_frfree(f);
1780 } /* end for */
1781 if (winner == in) {
1782 struct ast_frame *f = ast_read(in);
1783 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1784 /* Got hung up */
1785 *to_answer = -1;
1786 strcpy(pa->status, "CANCEL");
1787 pa->canceled = 1;
1788 publish_dial_end_event(in, out_chans, NULL, pa->status);
1789 if (f) {
1790 if (f->data.uint32) {
1791 ast_channel_hangupcause_set(in, f->data.uint32);
1792 }
1793 ast_frfree(f);
1794 }
1795 if (is_cc_recall) {
1796 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1797 }
1798 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller hung up\n", ast_channel_name(in));
1799 }
1800
1801 /* now f is guaranteed non-NULL */
1802 if (f->frametype == AST_FRAME_DTMF) {
1803 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1804 const char *context;
1805 ast_channel_lock(in);
1806 if ((context = pbx_builtin_getvar_helper(in, "EXITCONTEXT"))) {
1807 context = ast_strdupa(context);
1808 }
1809 ast_channel_unlock(in);
1810 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1811 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1812 *to_answer = 0;
1813 *result = f->subclass.integer;
1814 strcpy(pa->status, "CANCEL");
1815 pa->canceled = 1;
1816 publish_dial_end_event(in, out_chans, NULL, pa->status);
1817 ast_frfree(f);
1818 if (is_cc_recall) {
1819 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1820 }
1821 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller pressed %c to end call\n",
1822 ast_channel_name(in), f->subclass.integer);
1823 }
1824 }
1825
1826 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1827 detect_disconnect(in, f->subclass.integer, &featurecode)) {
1828 ast_verb(3, "User requested call disconnect.\n");
1829 *to_answer = 0;
1830 strcpy(pa->status, "CANCEL");
1831 pa->canceled = 1;
1832 publish_dial_end_event(in, out_chans, NULL, pa->status);
1833 ast_frfree(f);
1834 if (is_cc_recall) {
1835 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1836 }
1837 SCOPE_EXIT_RTN_VALUE(NULL, "%s: Caller requested disconnect\n",
1838 ast_channel_name(in));
1839 }
1840 }
1841
1842 /* Send the frame from the in channel to all outgoing channels. */
1843 AST_LIST_TRAVERSE(out_chans, o, node) {
1844 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1845 /* This outgoing channel has died so don't send the frame to it. */
1846 continue;
1847 }
1848 switch (f->frametype) {
1849 case AST_FRAME_HTML:
1850 /* Forward HTML stuff */
1851 if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1852 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1853 ast_log(LOG_WARNING, "Unable to send URL on %s\n", ast_channel_name(o->chan));
1854 }
1855 break;
1856 case AST_FRAME_VIDEO:
1857 case AST_FRAME_VOICE:
1858 case AST_FRAME_IMAGE:
1859 if (!single || caller_entertained) {
1860 /*
1861 * We are calling multiple parties or caller is being
1862 * entertained and has thus not been made compatible.
1863 * No need to check any other called parties.
1864 */
1865 goto skip_frame;
1866 }
1867 /* Fall through */
1868 case AST_FRAME_TEXT:
1869 case AST_FRAME_DTMF_BEGIN:
1870 case AST_FRAME_DTMF_END:
1871 if (ast_write(o->chan, f)) {
1872 ast_log(LOG_WARNING, "Unable to forward frametype %u on %s\n",
1873 f->frametype, ast_channel_name(o->chan));
1874 }
1875 break;
1876 case AST_FRAME_CONTROL:
1877 switch (f->subclass.integer) {
1878 case AST_CONTROL_HOLD:
1879 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1880 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1881 break;
1882 case AST_CONTROL_UNHOLD:
1883 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1884 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1885 break;
1886 case AST_CONTROL_FLASH:
1887 ast_verb(3, "Hook flash on %s\n", ast_channel_name(o->chan));
1888 ast_indicate(o->chan, AST_CONTROL_FLASH);
1889 break;
1890 case AST_CONTROL_VIDUPDATE:
1891 case AST_CONTROL_SRCUPDATE:
1892 case AST_CONTROL_SRCCHANGE:
1893 if (!single || caller_entertained) {
1894 /*
1895 * We are calling multiple parties or caller is being
1896 * entertained and has thus not been made compatible.
1897 * No need to check any other called parties.
1898 */
1899 goto skip_frame;
1900 }
1901 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1902 ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1903 ast_indicate(o->chan, f->subclass.integer);
1904 break;
1905 case AST_CONTROL_CONNECTED_LINE:
1906 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1907 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
1908 break;
1909 }
1910 if (ast_channel_connected_line_sub(in, o->chan, f, 1)) {
1911 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1912 }
1913 break;
1914 case AST_CONTROL_REDIRECTING:
1915 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1916 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
1917 break;
1918 }
1919 if (ast_channel_redirecting_sub(in, o->chan, f, 1)) {
1920 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1921 }
1922 break;
1923 default:
1924 /* We are not going to do anything with this frame. */
1925 goto skip_frame;
1926 }
1927 break;
1928 default:
1929 /* We are not going to do anything with this frame. */
1930 goto skip_frame;
1931 }
1932 }
1933skip_frame:;
1934 ast_frfree(f);
1935 }
1936 }
1937
1938wait_over:
1939 if (!*to_answer || ast_check_hangup(in)) {
1940 ast_verb(3, "Nobody picked up in %d ms\n", orig_answer_to);
1941 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1942 } else if (!*to_progress) {
1943 ast_verb(3, "No early media received in %d ms\n", orig_progress_to);
1944 publish_dial_end_event(in, out_chans, NULL, "CHANUNAVAIL");
1945 strcpy(pa->status, "CHANUNAVAIL");
1946 *to_answer = 0; /* Reset to prevent hangup */
1947 }
1948
1949 if (is_cc_recall) {
1950 ast_cc_completed(in, "Recall completed!");
1951 }
1952 SCOPE_EXIT_RTN_VALUE(peer, "%s: %s%s\n", ast_channel_name(in),
1953 peer ? "Answered by " : "No answer", peer ? ast_channel_name(peer) : "");
1954}
1955
1956static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1957{
1958 char disconnect_code[AST_FEATURE_MAX_LEN];
1959 int res;
1960
1961 ast_str_append(featurecode, 1, "%c", code);
1962
1963 res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1964 if (res) {
1965 ast_str_reset(*featurecode);
1966 return 0;
1967 }
1968
1969 if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1970 /* Could be a partial match, anyway */
1971 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1972 ast_str_reset(*featurecode);
1973 }
1974 return 0;
1975 }
1976
1977 if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
1978 ast_str_reset(*featurecode);
1979 return 0;
1980 }
1981
1982 return 1;
1983}
1984
1985/* returns true if there is a valid privacy reply */
1986static int valid_priv_reply(struct ast_flags64 *opts, int res)
1987{
1988 if (res < '1')
1989 return 0;
1990 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1991 return 1;
1992 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1993 return 1;
1994 return 0;
1995}
1996
1997static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1998 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1999{
2000
2001 int res2;
2002 int loopcount = 0;
2003
2004 /* Get the user's intro, store it in priv-callerintros/$CID,
2005 unless it is already there-- this should be done before the
2006 call is actually dialed */
2007
2008 /* all ring indications and moh for the caller has been halted as soon as the
2009 target extension was picked up. We are going to have to kill some
2010 time and make the caller believe the peer hasn't picked up yet */
2011
2012 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2013 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2014 ast_indicate(chan, -1);
2015 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2016 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2017 ast_channel_musicclass_set(chan, original_moh);
2018 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2019 ast_indicate(chan, AST_CONTROL_RINGING);
2020 pa->sentringing++;
2021 }
2022
2023 /* Start autoservice on the other chan ?? */
2024 res2 = ast_autoservice_start(chan);
2025 /* Now Stream the File */
2026 for (loopcount = 0; loopcount < 3; loopcount++) {
2027 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
2028 break;
2029 if (!res2) /* on timeout, play the message again */
2030 res2 = ast_play_and_wait(peer, "priv-callpending");
2031 if (!valid_priv_reply(opts, res2))
2032 res2 = 0;
2033 /* priv-callpending script:
2034 "I have a caller waiting, who introduces themselves as:"
2035 */
2036 if (!res2)
2037 res2 = ast_play_and_wait(peer, pa->privintro);
2038 if (!valid_priv_reply(opts, res2))
2039 res2 = 0;
2040 /* now get input from the called party, as to their choice */
2041 if (!res2) {
2042 /* XXX can we have both, or they are mutually exclusive ? */
2043 if (ast_test_flag64(opts, OPT_PRIVACY))
2044 res2 = ast_play_and_wait(peer, "priv-callee-options");
2045 if (ast_test_flag64(opts, OPT_SCREENING))
2046 res2 = ast_play_and_wait(peer, "screen-callee-options");
2047 }
2048
2049 /*! \page DialPrivacy Dial Privacy scripts
2050 * \par priv-callee-options script:
2051 * \li Dial 1 if you wish this caller to reach you directly in the future,
2052 * and immediately connect to their incoming call.
2053 * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
2054 * \li Dial 3 to send this caller to the torture menus, now and forevermore.
2055 * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
2056 * \li Dial 5 to allow this caller to come straight thru to you in the future,
2057 * but right now, just this once, send them to voicemail.
2058 *
2059 * \par screen-callee-options script:
2060 * \li Dial 1 if you wish to immediately connect to the incoming call
2061 * \li Dial 2 if you wish to send this caller to voicemail.
2062 * \li Dial 3 to send this caller to the torture menus.
2063 * \li Dial 4 to send this caller to a simple "go away" menu.
2064 */
2065 if (valid_priv_reply(opts, res2))
2066 break;
2067 /* invalid option */
2068 res2 = ast_play_and_wait(peer, "vm-sorry");
2069 }
2070
2071 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
2072 ast_moh_stop(chan);
2073 } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
2074 ast_indicate(chan, -1);
2075 pa->sentringing = 0;
2076 }
2077 ast_autoservice_stop(chan);
2078 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
2079 /* map keypresses to various things, the index is res2 - '1' */
2080 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
2081 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
2082 int i = res2 - '1';
2083 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
2084 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
2085 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
2086 }
2087 switch (res2) {
2088 case '1':
2089 break;
2090 case '2':
2091 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2092 break;
2093 case '3':
2094 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2095 break;
2096 case '4':
2097 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2098 break;
2099 case '5':
2100 if (ast_test_flag64(opts, OPT_PRIVACY)) {
2101 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2102 break;
2103 }
2104 /* if not privacy, then 5 is the same as "default" case */
2105 default: /* bad input or -1 if failure to start autoservice */
2106 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
2107 /* well, there seems basically two choices. Just patch the caller thru immediately,
2108 or,... put 'em thru to voicemail. */
2109 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
2110 ast_verb(3, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
2111 /* XXX should we set status to DENY ? */
2112 /* XXX what about the privacy flags ? */
2113 break;
2114 }
2115
2116 if (res2 == '1') { /* the only case where we actually connect */
2117 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
2118 just clog things up, and it's not useful information, not being tied to a CID */
2119 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
2120 ast_filedelete(pa->privintro, NULL);
2121 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2122 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2123 else
2124 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2125 }
2126 return 0; /* the good exit path */
2127 } else {
2128 return -1;
2129 }
2130}
2131
2132/*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
2133static int setup_privacy_args(struct privacy_args *pa,
2134 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
2135{
2136 char callerid[60];
2137 int res;
2138 char *l;
2139
2140 if (ast_channel_caller(chan)->id.number.valid
2141 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2142 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2143 ast_shrink_phone_number(l);
2144 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
2145 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
2146 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
2147 } else {
2148 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
2149 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2150 }
2151 } else {
2152 char *tnam, *tn2;
2153
2154 tnam = ast_strdupa(ast_channel_name(chan));
2155 /* clean the channel name so slashes don't try to end up in disk file name */
2156 for (tn2 = tnam; *tn2; tn2++) {
2157 if (*tn2 == '/') /* any other chars to be afraid of? */
2158 *tn2 = '=';
2159 }
2160 ast_verb(3, "Privacy-- callerid is empty\n");
2161
2162 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
2163 l = callerid;
2164 pa->privdb_val = AST_PRIVACY_UNKNOWN;
2165 }
2166
2167 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
2168
2169 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
2170 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
2171 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
2172 pa->privdb_val = AST_PRIVACY_ALLOW;
2173 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
2174 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
2175 }
2176
2177 if (pa->privdb_val == AST_PRIVACY_DENY) {
2178 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
2179 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
2180 return 0;
2181 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
2182 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
2183 return 0; /* Is this right? */
2184 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
2185 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
2186 return 0; /* is this right??? */
2187 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
2188 /* Get the user's intro, store it in priv-callerintros/$CID,
2189 unless it is already there-- this should be done before the
2190 call is actually dialed */
2191
2192 /* make sure the priv-callerintros dir actually exists */
2193 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
2194 if ((res = ast_mkdir(pa->privintro, 0755))) {
2195 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
2196 return -1;
2197 }
2198
2199 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
2200 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
2201 /* the DELUX version of this code would allow this caller the
2202 option to hear and retape their previously recorded intro.
2203 */
2204 } else {
2205 int duration; /* for feedback from play_and_wait */
2206 /* the file doesn't exist yet. Let the caller submit his
2207 vocal intro for posterity */
2208 /* priv-recordintro script:
2209 "At the tone, please say your name:"
2210 */
2211 int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
2212 ast_answer(chan);
2213 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 */
2214 /* don't think we'll need a lock removed, we took care of
2215 conflicts by naming the pa.privintro file */
2216 if (res == -1) {
2217 /* Delete the file regardless since they hung up during recording */
2218 ast_filedelete(pa->privintro, NULL);
2219 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
2220 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
2221 else
2222 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
2223 return -1;
2224 }
2225 if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
2226 ast_waitstream(chan, "");
2227 }
2228 }
2229 return 1; /* success */
2230}
2231
2232static void end_bridge_callback(void *data)
2233{
2234 struct ast_channel *chan = data;
2235
2236 ast_channel_lock(chan);
2237 ast_channel_stage_snapshot(chan);
2238 set_duration_var(chan, "ANSWEREDTIME", ast_channel_get_up_time_ms(chan));
2239 set_duration_var(chan, "DIALEDTIME", ast_channel_get_duration_ms(chan));
2240 ast_channel_stage_snapshot_done(chan);
2241 ast_channel_unlock(chan);
2242}
2243
2244static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2245 bconfig->end_bridge_callback_data = originator;
2246}
2247
2248static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2249{
2250 struct ast_tone_zone_sound *ts = NULL;
2251 int res;
2252 const char *str = data;
2253
2254 if (ast_strlen_zero(str)) {
2255 ast_debug(1,"Nothing to play\n");
2256 return -1;
2257 }
2258
2259 ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2260
2261 if (ts && ts->data[0]) {
2262 res = ast_playtones_start(chan, 0, ts->data, 0);
2263 } else {
2264 res = -1;
2265 }
2266
2267 if (ts) {
2268 ts = ast_tone_zone_sound_unref(ts);
2269 }
2270
2271 if (res) {
2272 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2273 }
2274
2275 return res;
2276}
2277
2278/*!
2279 * \internal
2280 * \brief Setup the after bridge goto location on the peer.
2281 * \since 12.0.0
2282 *
2283 * \param chan Calling channel for bridge.
2284 * \param peer Peer channel for bridge.
2285 * \param opts Dialing option flags.
2286 * \param opt_args Dialing option argument strings.
2287 */
2288static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
2289{
2290 const char *context;
2291 const char *extension;
2292 int priority;
2293
2294 if (ast_test_flag64(opts, OPT_PEER_H)) {
2295 ast_channel_lock(chan);
2296 context = ast_strdupa(ast_channel_context(chan));
2297 ast_channel_unlock(chan);
2298 ast_bridge_set_after_h(peer, context);
2299 } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2300 ast_channel_lock(chan);
2301 context = ast_strdupa(ast_channel_context(chan));
2302 extension = ast_strdupa(ast_channel_exten(chan));
2303 priority = ast_channel_priority(chan);
2304 ast_channel_unlock(chan);
2305 ast_bridge_set_after_go_on(peer, context, extension, priority,
2306 opt_args[OPT_ARG_CALLEE_GO_ON]);
2307 }
2308}
2309
2310static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2311{
2312 int res = -1; /* default: error */
2313 char *rest, *cur; /* scan the list of destinations */
2314 struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2315 struct chanlist *outgoing;
2316 struct chanlist *tmp;
2317 struct ast_channel *peer = NULL;
2318 int to_answer, to_progress; /* timeouts */
2319 struct cause_args num = { chan, 0, 0, 0 };
2320 int cause, hanguptreecause = -1;
2321
2322 struct ast_bridge_config config = { { 0, } };
2323 struct timeval calldurationlimit = { 0, };
2324 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress = NULL;
2325 char *mf_progress = NULL, *mf_wink = NULL;
2326 char *sf_progress = NULL, *sf_wink = NULL;
2327 struct privacy_args pa = {
2328 .sentringing = 0,
2329 .privdb_val = 0,
2330 .status = "INVALIDARGS",
2331 .canceled = 0,
2332 };
2333 int sentringing = 0, moh = 0;
2334 const char *outbound_group = NULL;
2335 int result = 0;
2336 char *parse;
2337 int opermode = 0;
2338 int delprivintro = 0;
2339 AST_DECLARE_APP_ARGS(args,
2340 AST_APP_ARG(peers);
2341 AST_APP_ARG(timeout);
2342 AST_APP_ARG(options);
2343 AST_APP_ARG(url);
2344 );
2345 struct ast_flags64 opts = { 0, };
2346 char *opt_args[OPT_ARG_ARRAY_SIZE];
2347 int fulldial = 0, num_dialed = 0;
2348 int ignore_cc = 0;
2349 char device_name[AST_CHANNEL_NAME];
2350 char forced_clid_name[AST_MAX_EXTENSION];
2351 char stored_clid_name[AST_MAX_EXTENSION];
2352 int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2353 /*!
2354 * \brief Forced CallerID party information to send.
2355 * \note This will not have any malloced strings so do not free it.
2356 */
2357 struct ast_party_id forced_clid;
2358 /*!
2359 * \brief Stored CallerID information if needed.
2360 *
2361 * \note If OPT_ORIGINAL_CLID set then this is the o option
2362 * CallerID. Otherwise it is the dialplan extension and hint
2363 * name.
2364 *
2365 * \note This will not have any malloced strings so do not free it.
2366 */
2367 struct ast_party_id stored_clid;
2368 /*!
2369 * \brief CallerID party information to store.
2370 * \note This will not have any malloced strings so do not free it.
2371 */
2372 struct ast_party_caller caller;
2373 int max_forwards;
2374 struct ast_datastore *topology_ds = NULL;
2375 SCOPE_ENTER(1, "%s: Data: %s\n", ast_channel_name(chan), data);
2376
2377 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2378 ast_channel_lock(chan);
2379 ast_channel_stage_snapshot(chan);
2380 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2381 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2382 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2383 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2384 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME_MS", "");
2385 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2386 pbx_builtin_setvar_helper(chan, "DIALEDTIME_MS", "");
2387 pbx_builtin_setvar_helper(chan, "RINGTIME", "");
2388 pbx_builtin_setvar_helper(chan, "RINGTIME_MS", "");
2389 pbx_builtin_setvar_helper(chan, "PROGRESSTIME", "");
2390 pbx_builtin_setvar_helper(chan, "PROGRESSTIME_MS", "");
2391 ast_channel_stage_snapshot_done(chan);
2392 max_forwards = ast_max_forwards_get(chan);
2393 ast_channel_unlock(chan);
2394
2395 if (max_forwards <= 0) {
2396 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2397 ast_channel_name(chan));
2398 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2399 SCOPE_EXIT_RTN_VALUE(-1, "%s: Max forwards exceeded\n", ast_channel_name(chan));
2400 }
2401
2402 if (ast_check_hangup_locked(chan)) {
2403 /*
2404 * Caller hung up before we could dial. If dial is executed
2405 * within an AGI then the AGI has likely eaten all queued
2406 * frames before executing the dial in DeadAGI mode. With
2407 * the caller hung up and no pending frames from the caller's
2408 * read queue, dial would not know that the call has hung up
2409 * until a called channel answers. It is rather annoying to
2410 * whoever just answered the non-existent call.
2411 *
2412 * Dial should not continue execution in DeadAGI mode, hangup
2413 * handlers, or the h exten.
2414 */
2415 ast_verb(3, "Caller hung up before dial.\n");
2416 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2417 SCOPE_EXIT_RTN_VALUE(-1, "%s: Caller hung up before dial\n", ast_channel_name(chan));
2418 }
2419
2420 parse = ast_strdupa(data ?: "");
2421
2422 AST_STANDARD_APP_ARGS(args, parse);
2423
2424 if (!ast_strlen_zero(args.options) &&
2425 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2426 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2427 goto done;
2428 }
2429
2430 if (ast_cc_call_init(chan, &ignore_cc)) {
2431 goto done;
2432 }
2433
2434 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2435 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2436
2437 if (delprivintro < 0 || delprivintro > 1) {
2438 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2439 delprivintro = 0;
2440 }
2441 }
2442
2443 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2444 opt_args[OPT_ARG_RINGBACK] = NULL;
2445 }
2446
2447 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2448 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2449 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2450 }
2451
2452 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2453 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2454 if (!calldurationlimit.tv_sec) {
2455 ast_log(LOG_WARNING, "Dial does not accept S(%s)\n", opt_args[OPT_ARG_DURATION_STOP]);
2456 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2457 goto done;
2458 }
2459 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2460 }
2461
2462 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2463 sf_wink = opt_args[OPT_ARG_SENDDTMF];
2464 dtmfcalled = strsep(&sf_wink, ":");
2465 dtmfcalling = strsep(&sf_wink, ":");
2466 dtmf_progress = strsep(&sf_wink, ":");
2467 mf_progress = strsep(&sf_wink, ":");
2468 mf_wink = strsep(&sf_wink, ":");
2469 sf_progress = strsep(&sf_wink, ":");
2470 }
2471
2472 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2473 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2474 goto done;
2475 }
2476
2477 /* Setup the forced CallerID information to send if used. */
2478 ast_party_id_init(&forced_clid);
2479 force_forwards_only = 0;
2480 if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2481 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2482 ast_channel_lock(chan);
2483 forced_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2484 ast_channel_unlock(chan);
2485 forced_clid_name[0] = '\0';
2486 forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2487 sizeof(forced_clid_name), chan);
2488 force_forwards_only = 1;
2489 } else {
2490 /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2491 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2492 &forced_clid.number.str);
2493 }
2494 if (!ast_strlen_zero(forced_clid.name.str)) {
2495 forced_clid.name.valid = 1;
2496 }
2497 if (!ast_strlen_zero(forced_clid.number.str)) {
2498 forced_clid.number.valid = 1;
2499 }
2500 }
2501 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2502 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2503 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2504 }
2505 forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2506 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2507 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2508 int pres;
2509
2510 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2511 if (0 <= pres) {
2512 forced_clid.number.presentation = pres;
2513 }
2514 }
2515
2516 /* Setup the stored CallerID information if needed. */
2517 ast_party_id_init(&stored_clid);
2518 if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2519 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2520 ast_channel_lock(chan);
2521 ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2522 if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2523 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2524 }
2525 if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2526 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2527 }
2528 if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2529 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2530 }
2531 if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2532 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2533 }
2534 ast_channel_unlock(chan);
2535 } else {
2536 /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2537 ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2538 &stored_clid.number.str);
2539 if (!ast_strlen_zero(stored_clid.name.str)) {
2540 stored_clid.name.valid = 1;
2541 }
2542 if (!ast_strlen_zero(stored_clid.number.str)) {
2543 stored_clid.number.valid = 1;
2544 }
2545 }
2546 } else {
2547 /*
2548 * In case the new channel has no preset CallerID number by the
2549 * channel driver, setup the dialplan extension and hint name.
2550 */
2551 stored_clid_name[0] = '\0';
2552 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2553 sizeof(stored_clid_name), chan);
2554 if (ast_strlen_zero(stored_clid.name.str)) {
2555 stored_clid.name.str = NULL;
2556 } else {
2557 stored_clid.name.valid = 1;
2558 }
2559 ast_channel_lock(chan);
2560 stored_clid.number.str = ast_strdupa(ast_channel_exten(chan));
2561 stored_clid.number.valid = 1;
2562 ast_channel_unlock(chan);
2563 }
2564
2565 if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2566 ast_cdr_reset(ast_channel_name(chan), 0);
2567 }
2568 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2569 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2570
2571 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2572 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2573 if (res <= 0)
2574 goto out;
2575 res = -1; /* reset default */
2576 }
2577
2578 if (continue_exec)
2579 *continue_exec = 0;
2580
2581 /* If a channel group has been specified, get it for use when we create peer channels */
2582
2583 ast_channel_lock(chan);
2584 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2585 outbound_group = ast_strdupa(outbound_group);
2586 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2587 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2588 outbound_group = ast_strdupa(outbound_group);
2589 }
2590 ast_channel_unlock(chan);
2591
2592 /* Set per dial instance flags. These flags are also passed back to RetryDial. */
2593 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2594 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2595 | OPT_ANNOUNCE | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2596
2597 /* PREDIAL: Run gosub on the caller's channel */
2598 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2599 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2600 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2601 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2602 }
2603
2604 /* loop through the list of dial destinations */
2605 rest = args.peers;
2606 while ((cur = strsep(&rest, "&"))) {
2607 struct ast_channel *tc; /* channel for this destination */
2608 char *number;
2609 char *tech;
2610 int i;
2611 size_t tech_len;
2612 size_t number_len;
2613 struct ast_stream_topology *topology;
2614 struct ast_stream *stream;
2615
2616 cur = ast_strip(cur);
2617 if (ast_strlen_zero(cur)) {
2618 /* No tech/resource in this position. */
2619 continue;
2620 }
2621
2622 /* Get a technology/resource pair */
2623 number = cur;
2624 tech = strsep(&number, "/");
2625
2626 num_dialed++;
2627 if (ast_strlen_zero(number)) {
2628 ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2629 goto out;
2630 }
2631
2632 tech_len = strlen(tech) + 1;
2633 number_len = strlen(number) + 1;
2634 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2635 if (!tmp) {
2636 goto out;
2637 }
2638
2639 /* Save tech, number, and interface. */
2640 cur = tmp->stuff;
2641 strcpy(cur, tech);
2642 tmp->tech = cur;
2643 cur += tech_len;
2644 strcpy(cur, tech);
2645 cur[tech_len - 1] = '/';
2646 tmp->interface = cur;
2647 cur += tech_len;
2648 strcpy(cur, number);
2649 tmp->number = cur;
2650
2651 if (opts.flags) {
2652 /* Set per outgoing call leg options. */
2653 ast_copy_flags64(tmp, &opts,
2654 OPT_CANCEL_ELSEWHERE |
2655 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2656 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2657 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2658 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2659 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2660 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2661 OPT_RING_WITH_EARLY_MEDIA);
2662 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2663 }
2664
2665 /* Request the peer */
2666
2667 ast_channel_lock(chan);
2668 /*
2669 * Seed the chanlist's connected line information with previously
2670 * acquired connected line info from the incoming channel. The
2671 * previously acquired connected line info could have been set
2672 * through the CONNECTED_LINE dialplan function.
2673 */
2674 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2675
2676 if (ast_test_flag64(&opts, OPT_TOPOLOGY_PRESERVE)) {
2677 topology_ds = ast_channel_datastore_find(chan, &topology_ds_info, NULL);
2678
2679 if (!topology_ds && (topology_ds = ast_datastore_alloc(&topology_ds_info, NULL))) {
2680 topology_ds->data = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2681 ast_channel_datastore_add(chan, topology_ds);
2682 }
2683 }
2684
2685 if (topology_ds) {
2686 ao2_ref(topology_ds->data, +1);
2687 topology = topology_ds->data;
2688 } else {
2689 topology = ast_stream_topology_clone(ast_channel_get_stream_topology(chan));
2690 }
2691
2692 ast_channel_unlock(chan);
2693
2694 for (i = 0; i < ast_stream_topology_get_count(topology); ++i) {
2695 stream = ast_stream_topology_get_stream(topology, i);
2696 /* For both recvonly and sendonly the stream state reflects our state, that is we
2697 * are receiving only and we are sending only. Since we are requesting a
2698 * channel for the peer, we need to swap this to reflect what we will be doing.
2699 * That is, if we are receiving from Alice then we want to be sending to Bob,
2700 * so swap recvonly to sendonly and vice versa.
2701 */
2702 if (ast_stream_get_state(stream) == AST_STREAM_STATE_RECVONLY) {
2703 ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
2704 } else if (ast_stream_get_state(stream) == AST_STREAM_STATE_SENDONLY) {
2705 ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
2706 }
2707 }
2708
2709 tc = ast_request_with_stream_topology(tmp->tech, topology, NULL, chan, tmp->number, &cause);
2710
2711 ast_stream_topology_free(topology);
2712
2713 if (!tc) {
2714 /* If we can't, just go on to the next call */
2715 /* Failure doesn't necessarily mean user error. DAHDI channels could be busy. */
2716 ast_log(LOG_NOTICE, "Unable to create channel of type '%s' (cause %d - %s)\n",
2717 tmp->tech, cause, ast_cause2str(cause));
2718 handle_cause(cause, &num);
2719 if (!rest) {
2720 /* we are on the last destination */
2721 ast_channel_hangupcause_set(chan, cause);
2722 }
2723 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2724 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2725 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2726 }
2727 }
2728 chanlist_free(tmp);
2729 continue;
2730 }
2731
2732 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2733 if (!ignore_cc) {
2734 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2735 }
2736
2737 ast_channel_lock_both(tc, chan);
2738 ast_channel_stage_snapshot(tc);
2739
2740 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2741
2742 /* Setup outgoing SDP to match incoming one */
2743 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2744 /* We are on the only destination. */
2745 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2746 }
2747
2748 /* Inherit specially named variables from parent channel */
2749 ast_channel_inherit_variables(chan, tc);
2750 ast_channel_datastore_inherit(chan, tc);
2751 ast_max_forwards_decrement(tc);
2752
2753 ast_channel_appl_set(tc, "AppDial");
2754 ast_channel_data_set(tc, "(Outgoing Line)");
2755
2756 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2757
2758 /* Determine CallerID to store in outgoing channel. */
2759 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2760 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2761 caller.id = stored_clid;
2762 ast_channel_set_caller_event(tc, &caller, NULL);
2763 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2764 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2765 ast_channel_caller(tc)->id.number.str, NULL))) {
2766 /*
2767 * The new channel has no preset CallerID number by the channel
2768 * driver. Use the dialplan extension and hint name.
2769 */
2770 caller.id = stored_clid;
2771 if (!caller.id.name.valid
2772 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2773 ast_channel_connected(chan)->id.name.str, NULL))) {
2774 /*
2775 * No hint name available. We have a connected name supplied by
2776 * the dialplan we can use instead.
2777 */
2778 caller.id.name.valid = 1;
2779 caller.id.name = ast_channel_connected(chan)->id.name;
2780 }
2781 ast_channel_set_caller_event(tc, &caller, NULL);
2782 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2783 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2784 NULL))) {
2785 /* The new channel has no preset CallerID name by the channel driver. */
2786 if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2787 ast_channel_connected(chan)->id.name.str, NULL))) {
2788 /*
2789 * We have a connected name supplied by the dialplan we can
2790 * use instead.
2791 */
2792 caller.id.name.valid = 1;
2793 caller.id.name = ast_channel_connected(chan)->id.name;
2794 ast_channel_set_caller_event(tc, &caller, NULL);
2795 }
2796 }
2797
2798 /* Determine CallerID for outgoing channel to send. */
2799 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2800 struct ast_party_connected_line connected;
2801
2802 ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2803 connected.id = forced_clid;
2804 ast_channel_set_connected_line(tc, &connected, NULL);
2805 } else {
2806 ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2807 }
2808
2809 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2810
2811 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2812
2813 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2814 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2815 ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
2816 }
2817
2818 /* Pass ADSI CPE and transfer capability */
2819 ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
2820 ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
2821
2822 /* If we have an outbound group, set this peer channel to it */
2823 if (outbound_group)
2824 ast_app_group_set_channel(tc, outbound_group);
2825 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2826 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
2827 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2828
2829 /* Check if we're forced by configuration */
2830 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2831 ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
2832
2833
2834 /* Inherit context and extension */
2835 ast_channel_dialcontext_set(tc, ast_channel_context(chan));
2836 ast_channel_exten_set(tc, ast_channel_exten(chan));
2837
2838 ast_channel_stage_snapshot_done(tc);
2839
2840 /* Save the original channel name to detect call pickup masquerading in. */
2841 tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));
2842
2843 ast_channel_unlock(tc);
2844 ast_channel_unlock(chan);
2845
2846 /* Put channel in the list of outgoing thingies. */
2847 tmp->chan = tc;
2848 AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
2849 }
2850
2851 /* As long as we attempted to dial valid peers, don't throw a warning. */
2852 /* If a DAHDI peer is busy, out_chans will be empty so checking list size is misleading. */
2853 if (!num_dialed) {
2854 ast_verb(3, "No devices or endpoints to dial (technology/resource)\n");
2855 if (continue_exec) {
2856 /* There is no point in having RetryDial try again */
2857 *continue_exec = 1;
2858 }
2859 strcpy(pa.status, "CHANUNAVAIL");
2860 res = 0;
2861 goto out;
2862 }
2863
2864 /*
2865 * PREDIAL: Run gosub on all of the callee channels
2866 *
2867 * We run the callee predial before ast_call() in case the user
2868 * wishes to do something on the newly created channels before
2869 * the channel does anything important.
2870 *
2871 * Inside the target gosub we will be able to do something with
2872 * the newly created channel name ie: now the calling channel
2873 * can know what channel will be used to call the destination
2874 * ex: now we will know that SIP/abc-123 is calling SIP/def-124
2875 */
2876 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
2877 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
2878 && !AST_LIST_EMPTY(&out_chans)) {
2879 const char *predial_callee;
2880
2881 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
2882 predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
2883 if (predial_callee) {
2884 ast_autoservice_start(chan);
2885 AST_LIST_TRAVERSE(&out_chans, tmp, node) {
2886 ast_pre_call(tmp->chan, predial_callee);
2887 }
2888 ast_autoservice_stop(chan);
2889 ast_free((char *) predial_callee);
2890 }
2891 }
2892
2893 /* Start all outgoing calls */
2894 AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
2895 res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
2896 ast_channel_lock(chan);
2897
2898 /* check the results of ast_call */
2899 if (res) {
2900 /* Again, keep going even if there's an error */
2901 ast_debug(1, "ast call on peer returned %d\n", res);
2902 ast_verb(3, "Couldn't call %s\n", tmp->interface);
2903 if (ast_channel_hangupcause(tmp->chan)) {
2904 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
2905 }
2906 ast_channel_unlock(chan);
2907 ast_cc_call_failed(chan, tmp->chan, tmp->interface);
2908 ast_hangup(tmp->chan);
2909 tmp->chan = NULL;
2910 AST_LIST_REMOVE_CURRENT(node);
2911 chanlist_free(tmp);
2912 continue;
2913 }
2914
2915 ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
2916 ast_channel_unlock(chan);
2917
2918 ast_verb(3, "Called %s\n", tmp->interface);
2919 ast_set_flag64(tmp, DIAL_STILLGOING);
2920
2921 /* If this line is up, don't try anybody else */
2922 if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
2923 break;
2924 }
2925 }
2926 AST_LIST_TRAVERSE_SAFE_END;
2927
2928 if (ast_strlen_zero(args.timeout)) {
2929 to_answer = -1;
2930 to_progress = -1;
2931 } else {
2932 double tmp;
2933 char *anstimeout = strsep(&args.timeout, "^");
2934 if (!ast_strlen_zero(anstimeout)) {
2935 if (sscanf(anstimeout, "%30lf", &tmp) == 1 && tmp > 0) {
2936 to_answer = tmp * 1000;
2937 ast_debug(3, "Dial timeout set to %d ms\n", to_answer);
2938 } else {
2939 ast_log(LOG_WARNING, "Invalid answer timeout specified: '%s'. Setting timeout to infinite\n", anstimeout);
2940 to_answer = -1;
2941 }
2942 } else {
2943 to_answer = -1;
2944 }
2945 if (!ast_strlen_zero(args.timeout)) {
2946 if (sscanf(args.timeout, "%30lf", &tmp) == 1 && tmp > 0) {
2947 to_progress = tmp * 1000;
2948 ast_debug(3, "Dial progress timeout set to %d ms\n", to_progress);
2949 } else {
2950 ast_log(LOG_WARNING, "Invalid progress timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2951 to_progress = -1;
2952 }
2953 } else {
2954 to_progress = -1;
2955 }
2956 }
2957
2958 outgoing = AST_LIST_FIRST(&out_chans);
2959 if (!outgoing) {
2960 strcpy(pa.status, "CHANUNAVAIL");
2961 if (fulldial == num_dialed) {
2962 res = -1;
2963 goto out;
2964 }
2965 } else {
2966 /* Our status will at least be NOANSWER */
2967 strcpy(pa.status, "NOANSWER");
2968 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2969 moh = 1;
2970 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2971 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
2972 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
2973 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2974 ast_channel_musicclass_set(chan, original_moh);
2975 } else {
2976 ast_moh_start(chan, NULL, NULL);
2977 }
2978 ast_indicate(chan, AST_CONTROL_PROGRESS);
2979 } else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
2980 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
2981 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
2982 ast_indicate(chan, AST_CONTROL_RINGING);
2983 sentringing++;
2984 } else {
2985 ast_indicate(chan, AST_CONTROL_PROGRESS);
2986 }
2987 } else {
2988 ast_indicate(chan, AST_CONTROL_RINGING);
2989 sentringing++;
2990 }
2991 }
2992 }
2993
2994 peer = wait_for_answer(chan, &out_chans, &to_answer, &to_progress, peerflags, opt_args, &pa, &num, &result,
2995 dtmf_progress, mf_progress, mf_wink, sf_progress, sf_wink,
2996 (ast_test_flag64(&opts, OPT_HEARPULSING) ? 1 : 0),
2997 ignore_cc, &forced_clid, &stored_clid, &config);
2998
2999 if (!peer) {
3000 if (result) {
3001 res = result;
3002 } else if (to_answer) { /* Musta gotten hung up */
3003 res = -1;
3004 } else { /* Nobody answered, next please? */
3005 res = 0;
3006 }
3007 } else {
3008 const char *number;
3009 const char *name;
3010 int dial_end_raised = 0;
3011 int cause = -1;
3012
3013 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
3014 ast_answer(chan);
3015 }
3016
3017 /* Ah ha! Someone answered within the desired timeframe. Of course after this
3018 we will always return with -1 so that it is hung up properly after the
3019 conversation. */
3020
3021 if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
3022 && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
3023 cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
3024 if (cause <= 0) {
3025 if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
3026 cause = 0;
3027 } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
3028 || cause < 0) {
3029 ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
3030 opt_args[OPT_ARG_HANGUPCAUSE]);
3031 cause = -1;
3032 }
3033 }
3034 }
3035 hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
3036
3037 /* If appropriate, log that we have a destination channel and set the answer time */
3038
3039 ast_channel_lock(peer);
3040 name = ast_strdupa(ast_channel_name(peer));
3041
3042 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
3043 if (ast_strlen_zero(number)) {
3044 number = NULL;
3045 } else {
3046 number = ast_strdupa(number);
3047 }
3048 ast_channel_unlock(peer);
3049
3050 ast_channel_lock(chan);
3051 ast_channel_stage_snapshot(chan);
3052
3053 strcpy(pa.status, "ANSWER");
3054 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3055
3056 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
3057 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
3058
3059 ast_channel_stage_snapshot_done(chan);
3060 ast_channel_unlock(chan);
3061
3062 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
3063 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
3064 ast_channel_sendurl( peer, args.url );
3065 }
3066 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
3067 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
3068 ast_channel_publish_dial(chan, peer, NULL, pa.status);
3069 /* hang up on the callee -- he didn't want to talk anyway! */
3070 ast_autoservice_chan_hangup_peer(chan, peer);
3071 res = 0;
3072 goto out;
3073 }
3074 }
3075 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
3076 res = 0;
3077 } else {
3078 int digit = 0;
3079 struct ast_channel *chans[2];
3080 struct ast_channel *active_chan;
3081 char *calledfile = NULL, *callerfile = NULL;
3082 int calledstream = 0, callerstream = 0;
3083
3084 chans[0] = chan;
3085 chans[1] = peer;
3086
3087 /* we need to stream the announcement(s) when the OPT_ARG_ANNOUNCE (-A) is set */
3088 callerfile = opt_args[OPT_ARG_ANNOUNCE];
3089 calledfile = strsep(&callerfile, ":");
3090
3091 /* stream the file(s) */
3092 if (!ast_strlen_zero(calledfile)) {
3093 res = ast_streamfile(peer, calledfile, ast_channel_language(peer));
3094 if (res) {
3095 res = 0;
3096 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", calledfile);
3097 } else {
3098 calledstream = 1;
3099 }
3100 }
3101 if (!ast_strlen_zero(callerfile)) {
3102 res = ast_streamfile(chan, callerfile, ast_channel_language(chan));
3103 if (res) {
3104 res = 0;
3105 ast_log(LOG_ERROR, "error streaming file '%s' to caller\n", callerfile);
3106 } else {
3107 callerstream = 1;
3108 }
3109 }
3110
3111 /* can't use ast_waitstream, because we're streaming two files at once, and can't block
3112 We'll need to handle both channels at once. */
3113
3114 ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
3115 while (ast_channel_stream(peer) || ast_channel_stream(chan)) {
3116 int mspeer, mschan;
3117
3118 mspeer = ast_sched_wait(ast_channel_sched(peer));
3119 mschan = ast_sched_wait(ast_channel_sched(chan));
3120
3121 if (calledstream) {
3122 if (mspeer < 0 && !ast_channel_timingfunc(peer)) {
3123 ast_stopstream(peer);
3124 calledstream = 0;
3125 }
3126 }
3127 if (callerstream) {
3128 if (mschan < 0 && !ast_channel_timingfunc(chan)) {
3129 ast_stopstream(chan);
3130 callerstream = 0;
3131 }
3132 }
3133
3134 if (!calledstream && !callerstream) {
3135 break;
3136 }
3137
3138 if (mspeer < 0)
3139 mspeer = 1000;
3140
3141 if (mschan < 0)
3142 mschan = 1000;
3143
3144 /* wait for the lowest maximum of the two */
3145 active_chan = ast_waitfor_n(chans, 2, (mspeer > mschan ? &mschan : &mspeer));
3146 if (active_chan) {
3147 struct ast_channel *other_chan;
3148 struct ast_frame *fr = ast_read(active_chan);
3149
3150 if (!fr) {
3151 ast_autoservice_chan_hangup_peer(chan, peer);
3152 res = -1;
3153 goto done;
3154 }
3155 switch (fr->frametype) {
3156 case AST_FRAME_DTMF_END:
3157 digit = fr->subclass.integer;
3158 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
3159 ast_stopstream(peer);
3160 res = ast_senddigit(chan, digit, 0);
3161 }
3162 break;
3163 case AST_FRAME_CONTROL:
3164 switch (fr->subclass.integer) {
3165 case AST_CONTROL_HANGUP:
3166 ast_frfree(fr);
3167 ast_autoservice_chan_hangup_peer(chan, peer);
3168 res = -1;
3169 goto done;
3170 case AST_CONTROL_CONNECTED_LINE:
3171 /* Pass COLP update to the other channel. */
3172 if (active_chan == chan) {
3173 other_chan = peer;
3174 } else {
3175 other_chan = chan;
3176 }
3177 if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)) {
3178 ast_indicate_data(other_chan, fr->subclass.integer,
3179 fr->data.ptr, fr->datalen);
3180 }
3181 break;
3182 default:
3183 break;
3184 }
3185 break;
3186 default:
3187 /* Ignore all others */
3188 break;
3189 }
3190 ast_frfree(fr);
3191 }
3192 ast_sched_runq(ast_channel_sched(peer));
3193 ast_sched_runq(ast_channel_sched(chan));
3194 }
3195 ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
3196 }
3197
3198 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
3199 /* chan and peer are going into the PBX; as such neither are considered
3200 * outgoing channels any longer */
3201 ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);
3202
3203 ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
3204 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
3205 /* peer goes to the same context and extension as chan, so just copy info from chan*/
3206 ast_channel_lock(peer);
3207 ast_channel_stage_snapshot(peer);
3208 ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
3209 ast_channel_context_set(peer, ast_channel_context(chan));
3210 ast_channel_exten_set(peer, ast_channel_exten(chan));
3211 ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
3212 ast_channel_stage_snapshot_done(peer);
3213 ast_channel_unlock(peer);
3214 if (ast_pbx_start(peer)) {
3215 ast_autoservice_chan_hangup_peer(chan, peer);
3216 }
3217 if (continue_exec)
3218 *continue_exec = 1;
3219 res = 0;
3220 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3221 goto done;
3222 }
3223
3224 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
3225 const char *gosub_result_peer;
3226 char *gosub_argstart;
3227 char *gosub_args = NULL;
3228 int gosub_res = -1;
3229
3230 ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
3231 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
3232 if (gosub_argstart) {
3233 const char *what_is_s = "s";
3234 *gosub_argstart = 0;
3235 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)) &&
3236 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))) {
3237 what_is_s = "~~s~~";
3238 }
3239 if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
3240 gosub_args = NULL;
3241 }
3242 *gosub_argstart = ',';
3243 } else {
3244 const char *what_is_s = "s";
3245 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)) &&
3246 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))) {
3247 what_is_s = "~~s~~";
3248 }
3249 if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
3250 gosub_args = NULL;
3251 }
3252 }
3253 if (gosub_args) {
3254 gosub_res = ast_app_exec_sub(chan, peer, gosub_args, 0);
3255 ast_free(gosub_args);
3256 } else {
3257 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
3258 }
3259
3260 ast_channel_lock_both(chan, peer);
3261
3262 if (!gosub_res && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
3263 char *gosub_transfer_dest;
3264 char *gosub_result = ast_strdupa(gosub_result_peer);
3265 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
3266
3267 /* Inherit return value from the peer, so it can be used in the master */
3268 if (gosub_retval) {
3269 pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
3270 }
3271
3272 ast_channel_unlock(peer);
3273 ast_channel_unlock(chan);
3274
3275 if (!strcasecmp(gosub_result, "BUSY")) {
3276 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
3277 ast_set_flag64(peerflags, OPT_GO_ON);
3278 gosub_res = -1;
3279 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
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, "CONTINUE")) {
3284 /* Hangup peer and continue with the next extension priority. */
3285 ast_set_flag64(peerflags, OPT_GO_ON);
3286 gosub_res = -1;
3287 } else if (!strcasecmp(gosub_result, "ABORT")) {
3288 /* Hangup both ends unless the caller has the g flag */
3289 gosub_res = -1;
3290 } else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
3291 gosub_transfer_dest = gosub_result + 5;
3292 gosub_res = -1;
3293 /* perform a transfer to a new extension */
3294 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
3295 ast_replace_subargument_delimiter(gosub_transfer_dest);
3296 }
3297 if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
3298 ast_set_flag64(peerflags, OPT_GO_ON);
3299 }
3300 }
3301 if (gosub_res) {
3302 res = gosub_res;
3303 if (!dial_end_raised) {
3304 ast_channel_publish_dial(chan, peer, NULL, gosub_result);
3305 dial_end_raised = 1;
3306 }
3307 }
3308 } else {
3309 ast_channel_unlock(peer);
3310 ast_channel_unlock(chan);
3311 }
3312 }
3313
3314 if (!res) {
3315
3316 /* None of the Dial options changed our status; inform
3317 * everyone that this channel answered
3318 */
3319 if (!dial_end_raised) {
3320 ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
3321 dial_end_raised = 1;
3322 }
3323
3324 if (!ast_tvzero(calldurationlimit)) {
3325 struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
3326 ast_channel_lock(peer);
3327 ast_channel_whentohangup_set(peer, &whentohangup);
3328 ast_channel_unlock(peer);
3329 }
3330 if (!ast_strlen_zero(dtmfcalled)) {
3331 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
3332 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
3333 }
3334 if (!ast_strlen_zero(dtmfcalling)) {
3335 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
3336 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
3337 }
3338 }
3339
3340 if (res) { /* some error */
3341 if (!ast_check_hangup(chan) && ast_check_hangup(peer)) {
3342 ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
3343 }
3344 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3345 if (ast_bridge_setup_after_goto(peer)
3346 || ast_pbx_start(peer)) {
3347 ast_autoservice_chan_hangup_peer(chan, peer);
3348 }
3349 res = -1;
3350 } else {
3351 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
3352 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
3353 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
3354 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
3355 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
3356 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
3357 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
3358 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
3359 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
3360 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
3361 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
3362 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
3363 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
3364 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
3365 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
3366 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
3367 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
3368 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
3369 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
3370 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
3371
3372 config.end_bridge_callback = end_bridge_callback;
3373 config.end_bridge_callback_data = chan;
3374 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
3375
3376 if (moh) {
3377 moh = 0;
3378 ast_moh_stop(chan);
3379 } else if (sentringing) {
3380 sentringing = 0;
3381 ast_indicate(chan, -1);
3382 }
3383 /* Be sure no generators are left on it and reset the visible indication */
3384 ast_deactivate_generator(chan);
3385 ast_channel_visible_indication_set(chan, 0);
3386 /* Make sure channels are compatible */
3387 res = ast_channel_make_compatible(chan, peer);
3388 if (res < 0) {
3389 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));
3390 ast_autoservice_chan_hangup_peer(chan, peer);
3391 res = -1;
3392 goto done;
3393 }
3394 if (opermode) {
3395 struct oprmode oprmode;
3396
3397 oprmode.peer = peer;
3398 oprmode.mode = opermode;
3399
3400 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
3401 }
3402 setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
3403
3404 res = ast_bridge_call(chan, peer, &config);
3405 }
3406 }
3407out:
3408 if (moh) {
3409 moh = 0;
3410 ast_moh_stop(chan);
3411 } else if (sentringing) {
3412 sentringing = 0;
3413 ast_indicate(chan, -1);
3414 }
3415
3416 if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3417 ast_filedelete(pa.privintro, NULL);
3418 if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
3419 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
3420 } else {
3421 ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
3422 }
3423 }
3424
3425 ast_channel_early_bridge(chan, NULL);
3426 /* forward 'answered elsewhere' if we received it */
3427 if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)) {
3428 hanguptreecause = AST_CAUSE_ANSWERED_ELSEWHERE;
3429 } else if (pa.canceled) { /* Caller canceled */
3430 if (ast_channel_hangupcause(chan))
3431 hanguptreecause = ast_channel_hangupcause(chan);
3432 else
3433 hanguptreecause = AST_CAUSE_NORMAL_CLEARING;
3434 }
3435 hanguptree(&out_chans, NULL, hanguptreecause);
3436 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
3437 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
3438
3439 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
3440 if (!ast_tvzero(calldurationlimit))
3441 memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
3442 res = 0;
3443 }
3444
3445done:
3446 if (config.answer_topology) {
3447 ast_trace(2, "%s Cleaning up topology: %p %s\n",
3448 peer ? ast_channel_name(peer) : "<no channel>", &config.answer_topology,
3449 ast_str_tmp(256, ast_stream_topology_to_str(config.answer_topology, &STR_TMP)));
3450
3451 /*
3452 * At this point, the channel driver that answered should have bumped the
3453 * topology refcount for itself. Here we're cleaning up the reference we added
3454 * in wait_for_answer().
3455 */
3456 ast_stream_topology_free(config.answer_topology);
3457 }
3458 if (config.warning_sound) {
3459 ast_free((char *)config.warning_sound);
3460 }
3461 if (config.end_sound) {
3462 ast_free((char *)config.end_sound);
3463 }
3464 if (config.start_sound) {
3465 ast_free((char *)config.start_sound);
3466 }
3467 ast_ignore_cc(chan);
3468 SCOPE_EXIT_RTN_VALUE(res, "%s: Done\n", ast_channel_name(chan));
3469}
3470
3471static int dial_exec(struct ast_channel *chan, const char *data)
3472{
3473 struct ast_flags64 peerflags;
3474
3475 memset(&peerflags, 0, sizeof(peerflags));
3476
3477 return dial_exec_full(chan, data, &peerflags, NULL);
3478}
3479
3480static int retrydial_exec(struct ast_channel *chan, const char *data)
3481{
3482 char *parse;
3483 const char *context = NULL;
3484 int sleepms = 0, loops = 0, res = -1;
3485 struct ast_flags64 peerflags = { 0, };
3486 AST_DECLARE_APP_ARGS(args,
3487 AST_APP_ARG(announce);
3488 AST_APP_ARG(sleep);
3489 AST_APP_ARG(retries);
3490 AST_APP_ARG(dialdata);
3491 );
3492
3493 if (ast_strlen_zero(data)) {
3494 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
3495 return -1;
3496 }
3497
3498 parse = ast_strdupa(data);
3499 AST_STANDARD_APP_ARGS(args, parse);
3500
3501 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
3502 sleepms *= 1000;
3503
3504 if (!ast_strlen_zero(args.retries)) {
3505 loops = atoi(args.retries);
3506 }
3507
3508 if (!args.dialdata) {
3509 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
3510 goto done;
3511 }
3512
3513 if (sleepms < 1000)
3514 sleepms = 10000;
3515
3516 if (!loops)
3517 loops = -1; /* run forever */
3518
3519 ast_channel_lock(chan);
3520 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
3521 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
3522 ast_channel_unlock(chan);
3523
3524 res = 0;
3525 while (loops) {
3526 int continue_exec;
3527
3528 ast_channel_data_set(chan, "Retrying");
3529 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3530 ast_moh_stop(chan);
3531
3532 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
3533 if (continue_exec)
3534 break;
3535
3536 if (res == 0) {
3537 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
3538 if (!ast_strlen_zero(args.announce)) {
3539 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3540 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3541 ast_waitstream(chan, AST_DIGIT_ANY);
3542 } else
3543 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3544 }
3545 if (!res && sleepms) {
3546 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3547 ast_moh_start(chan, NULL, NULL);
3548 res = ast_waitfordigit(chan, sleepms);
3549 }
3550 } else {
3551 if (!ast_strlen_zero(args.announce)) {
3552 if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
3553 if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
3554 res = ast_waitstream(chan, "");
3555 } else
3556 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
3557 }
3558 if (sleepms) {
3559 if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3560 ast_moh_start(chan, NULL, NULL);
3561 if (!res)
3562 res = ast_waitfordigit(chan, sleepms);
3563 }
3564 }
3565 }
3566
3567 if (res < 0 || res == AST_PBX_INCOMPLETE) {
3568 break;
3569 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
3570 if (onedigit_goto(chan, context, (char) res, 1)) {
3571 res = 0;
3572 break;
3573 }
3574 }
3575 loops--;
3576 }
3577 if (loops == 0)
3578 res = 0;
3579 else if (res == 1)
3580 res = 0;
3581
3582 if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
3583 ast_moh_stop(chan);
3584 done:
3585 return res;
3586}
3587
3588static int unload_module(void)
3589{
3590 int res;
3591
3592 res = ast_unregister_application(app);
3593 res |= ast_unregister_application(rapp);
3594
3595 return res;
3596}
3597
3598static int load_module(void)
3599{
3600 int res;
3601
3602 res = ast_register_application_xml(app, dial_exec);
3603 res |= ast_register_application_xml(rapp, retrydial_exec);
3604
3605 return res;
3606}
3607
3608AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Dialing Application",
3609 .support_level = AST_MODULE_SUPPORT_CORE,
3610 .load = load_module,
3611 .unload = unload_module,
3612);
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:827
DIAL_STILLGOING
#define DIAL_STILLGOING
Definition app_dial.c:707
dial_exec_full
static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
Definition app_dial.c:2310
OPT_PREDIAL_CALLER
#define OPT_PREDIAL_CALLER
Definition app_dial.c:718
OPT_CANCEL_ELSEWHERE
#define OPT_CANCEL_ELSEWHERE
Definition app_dial.c:710
get_cid_name
static const char * get_cid_name(char *name, int namelen, struct ast_channel *chan)
Definition app_dial.c:914
app
static const char app[]
Definition app_dial.c:670
dial_exec_options
static const struct ast_app_option dial_exec_options[128]
Definition app_dial.c:792
OPT_PEER_H
#define OPT_PEER_H
Definition app_dial.c:711
do_forward
static void do_forward(struct chanlist *o, struct cause_args *num, struct ast_flags64 *peerflags, int single, int caller_entertained, int *to, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
Definition app_dial.c:944
OPT_PREDIAL_CALLEE
#define OPT_PREDIAL_CALLEE
Definition app_dial.c:717
DIAL_CALLERID_ABSENT
#define DIAL_CALLERID_ABSENT
Definition app_dial.c:709
OPT_FORCE_CID_PRES
#define OPT_FORCE_CID_PRES
Definition app_dial.c:715
setup_peer_after_bridge_goto
static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
Definition app_dial.c:2288
CAN_EARLY_BRIDGE
#define CAN_EARLY_BRIDGE(flags, chan, peer)
Definition app_dial.c:794
OPT_TOPOLOGY_PRESERVE
#define OPT_TOPOLOGY_PRESERVE
Definition app_dial.c:722
OPT_RING_WITH_EARLY_MEDIA
#define OPT_RING_WITH_EARLY_MEDIA
Definition app_dial.c:719
OPT_FORCE_CID_TAG
#define OPT_FORCE_CID_TAG
Definition app_dial.c:714
OPT_HEARPULSING
#define OPT_HEARPULSING
Definition app_dial.c:721
dial_exec
static int dial_exec(struct ast_channel *chan, const char *data)
Definition app_dial.c:3471
DIAL_NOFORWARDHTML
#define DIAL_NOFORWARDHTML
Definition app_dial.c:708
AST_MAX_WATCHERS
#define AST_MAX_WATCHERS
Definition app_dial.c:865
OPT_CANCEL_TIMEOUT
#define OPT_CANCEL_TIMEOUT
Definition app_dial.c:713
chanlist_free
static void chanlist_free(struct chanlist *outgoing)
Definition app_dial.c:839
publish_dial_end_event
static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
Definition app_dial.c:1156
onedigit_goto
static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
Definition app_dial.c:899
rapp
static const char rapp[]
Definition app_dial.c:671
handle_cause
static void handle_cause(int cause, struct cause_args *num)
Definition app_dial.c:877
setup_privacy_args
static int setup_privacy_args(struct privacy_args *pa, struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
returns 1 if successful, 0 or <0 if the caller should 'goto out'
Definition app_dial.c:2133
set_duration_var
static void set_duration_var(struct ast_channel *chan, const char *var_base, int64_t duration)
Definition app_dial.c:1196
update_connected_line_from_peer
static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
Definition app_dial.c:1176
OPT_ARG_CALLEE_GO_ON
@ OPT_ARG_CALLEE_GO_ON
Definition app_dial.c:732
OPT_ARG_SENDDTMF
@ OPT_ARG_SENDDTMF
Definition app_dial.c:726
OPT_ARG_DURATION_STOP
@ OPT_ARG_DURATION_STOP
Definition app_dial.c:734
OPT_ARG_PREDIAL_CALLEE
@ OPT_ARG_PREDIAL_CALLEE
Definition app_dial.c:741
OPT_ARG_RINGBACK
@ OPT_ARG_RINGBACK
Definition app_dial.c:730
OPT_ARG_MUSICBACK
@ OPT_ARG_MUSICBACK
Definition app_dial.c:729
OPT_ARG_CALLEE_GOSUB
@ OPT_ARG_CALLEE_GOSUB
Definition app_dial.c:731
OPT_ARG_HANGUPCAUSE
@ OPT_ARG_HANGUPCAUSE
Definition app_dial.c:743
OPT_ARG_FORCE_CID_PRES
@ OPT_ARG_FORCE_CID_PRES
Definition app_dial.c:740
OPT_ARG_ANNOUNCE
@ OPT_ARG_ANNOUNCE
Definition app_dial.c:725
OPT_ARG_GOTO
@ OPT_ARG_GOTO
Definition app_dial.c:727
OPT_ARG_DURATION_LIMIT
@ OPT_ARG_DURATION_LIMIT
Definition app_dial.c:728
OPT_ARG_ORIGINAL_CLID
@ OPT_ARG_ORIGINAL_CLID
Definition app_dial.c:737
OPT_ARG_OPERMODE
@ OPT_ARG_OPERMODE
Definition app_dial.c:735
OPT_ARG_FORCECLID
@ OPT_ARG_FORCECLID
Definition app_dial.c:738
OPT_ARG_PREDIAL_CALLER
@ OPT_ARG_PREDIAL_CALLER
Definition app_dial.c:742
OPT_ARG_ARRAY_SIZE
@ OPT_ARG_ARRAY_SIZE
Definition app_dial.c:745
OPT_ARG_PRIVACY
@ OPT_ARG_PRIVACY
Definition app_dial.c:733
OPT_ARG_SCREEN_NOINTRO
@ OPT_ARG_SCREEN_NOINTRO
Definition app_dial.c:736
OPT_ARG_FORCE_CID_TAG
@ OPT_ARG_FORCE_CID_TAG
Definition app_dial.c:739
detect_disconnect
static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
Definition app_dial.c:1956
OPT_HANGUPCAUSE
#define OPT_HANGUPCAUSE
Definition app_dial.c:720
hanguptree
static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
Definition app_dial.c:847
OPT_RESETCDR
@ OPT_RESETCDR
Definition app_dial.c:675
OPT_SCREEN_NOINTRO
@ OPT_SCREEN_NOINTRO
Definition app_dial.c:685
OPT_DTMF_EXIT
@ OPT_DTMF_EXIT
Definition app_dial.c:676
OPT_ANNOUNCE
@ OPT_ANNOUNCE
Definition app_dial.c:674
OPT_CALLEE_PARK
@ OPT_CALLEE_PARK
Definition app_dial.c:698
OPT_DURATION_LIMIT
@ OPT_DURATION_LIMIT
Definition app_dial.c:683
OPT_SCREEN_NOCALLERID
@ OPT_SCREEN_NOCALLERID
Definition app_dial.c:686
OPT_IGNORE_FORWARDING
@ OPT_IGNORE_FORWARDING
Definition app_dial.c:700
OPT_OPERMODE
@ OPT_OPERMODE
Definition app_dial.c:697
OPT_DURATION_STOP
@ OPT_DURATION_STOP
Definition app_dial.c:691
OPT_GO_ON
@ OPT_GO_ON
Definition app_dial.c:679
OPT_RINGBACK
@ OPT_RINGBACK
Definition app_dial.c:690
OPT_GOTO
@ OPT_GOTO
Definition app_dial.c:696
OPT_IGNORE_CONNECTEDLINE
@ OPT_IGNORE_CONNECTEDLINE
Definition app_dial.c:687
OPT_CALLEE_TRANSFER
@ OPT_CALLEE_TRANSFER
Definition app_dial.c:692
OPT_SENDDTMF
@ OPT_SENDDTMF
Definition app_dial.c:677
OPT_CALLER_MIXMONITOR
@ OPT_CALLER_MIXMONITOR
Definition app_dial.c:703
OPT_CALLER_PARK
@ OPT_CALLER_PARK
Definition app_dial.c:699
OPT_CALLER_MONITOR
@ OPT_CALLER_MONITOR
Definition app_dial.c:695
OPT_CALLEE_MONITOR
@ OPT_CALLEE_MONITOR
Definition app_dial.c:694
OPT_CALLEE_GOSUB
@ OPT_CALLEE_GOSUB
Definition app_dial.c:701
OPT_CALLER_HANGUP
@ OPT_CALLER_HANGUP
Definition app_dial.c:681
OPT_FORCECLID
@ OPT_FORCECLID
Definition app_dial.c:678
OPT_CALLEE_HANGUP
@ OPT_CALLEE_HANGUP
Definition app_dial.c:680
OPT_SCREENING
@ OPT_SCREENING
Definition app_dial.c:688
OPT_MUSICBACK
@ OPT_MUSICBACK
Definition app_dial.c:684
OPT_CALLER_TRANSFER
@ OPT_CALLER_TRANSFER
Definition app_dial.c:693
OPT_CALLEE_MIXMONITOR
@ OPT_CALLEE_MIXMONITOR
Definition app_dial.c:702
OPT_ORIGINAL_CLID
@ OPT_ORIGINAL_CLID
Definition app_dial.c:682
OPT_PRIVACY
@ OPT_PRIVACY
Definition app_dial.c:689
topology_ds_info
static const struct ast_datastore_info topology_ds_info
Definition app_dial.c:832
load_module
static int load_module(void)
Definition app_dial.c:3598
retrydial_exec
static int retrydial_exec(struct ast_channel *chan, const char *data)
Definition app_dial.c:3480
dial_handle_playtones
static int dial_handle_playtones(struct ast_channel *chan, const char *data)
Definition app_dial.c:2248
end_bridge_callback
static void end_bridge_callback(void *data)
Definition app_dial.c:2232
unload_module
static int unload_module(void)
Definition app_dial.c:3588
OPT_CALLER_ANSWER
#define OPT_CALLER_ANSWER
Definition app_dial.c:716
wait_for_answer
static struct ast_channel * wait_for_answer(struct ast_channel *in, struct dial_head *out_chans, int *to_answer, int *to_progress, struct ast_flags64 *peerflags, char *opt_args[], struct privacy_args *pa, const struct cause_args *num_in, int *result, char *dtmf_progress, char *mf_progress, char *mf_wink, char *sf_progress, char *sf_wink, const int hearpulsing, const int ignore_cc, struct ast_party_id *forced_clid, struct ast_party_id *stored_clid, struct ast_bridge_config *config)
Definition app_dial.c:1209
end_bridge_callback_data_fixup
static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator)
Definition app_dial.c:2244
valid_priv_reply
static int valid_priv_reply(struct ast_flags64 *opts, int res)
Definition app_dial.c:1986
OPT_CALLEE_GO_ON
#define OPT_CALLEE_GO_ON
Definition app_dial.c:712
status
jack_status_t status
Definition app_jack.c:149
str
const char * str
Definition app_jack.c:150
silencethreshold
static int silencethreshold
Definition app_voicemail.c:1109
strsep
char * strsep(char **str, const char *delims)
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:4261
ast_ignore_cc
void ast_ignore_cc(struct ast_channel *chan)
Mark the channel to ignore further CC activity.
Definition ccss.c:3747
ast_cc_is_recall
int ast_cc_is_recall(struct ast_channel *chan, int *core_id, const char *const monitor_type)
Decide if a call to a particular channel is a CC recall.
Definition ccss.c:3459
ast_handle_cc_control_frame
void ast_handle_cc_control_frame(struct ast_channel *inbound, struct ast_channel *outbound, void *frame_data)
Properly react to a CC control frame.
Definition ccss.c:2334
ast_cc_completed
int ast_cc_completed(struct ast_channel *chan, const char *const debug,...)
Indicate recall has been acknowledged.
Definition ccss.c:3885
ast_cc_busy_interface
void ast_cc_busy_interface(struct ast_channel *inbound, struct ast_cc_config_params *cc_params, const char *monitor_type, const char *const device_name, const char *const dialstring, void *private_data)
Callback made from ast_cc_callback for certain channel types.
Definition ccss.c:4296
ast_cc_extension_monitor_add_dialstring
void ast_cc_extension_monitor_add_dialstring(struct ast_channel *incoming, const char *const dialstring, const char *const device_name)
Add a child dialstring to an extension monitor.
Definition ccss.c:2022
ast_cc_call_init
int ast_cc_call_init(struct ast_channel *chan, int *ignore_cc)
Start the CC process on a call.
Definition ccss.c:2429
ast_cc_failed
int ast_cc_failed(int core_id, const char *const debug,...)
Indicate failure has occurred.
Definition ccss.c:3924
ast_cc_callback
int ast_cc_callback(struct ast_channel *inbound, const char *const tech, const char *const dest, ast_cc_callback_fn callback)
Run a callback for potential matching destinations.
Definition ccss.c:4311
ast_cdr_reset
int ast_cdr_reset(const char *channel_name, int keep_variables)
Reset the detail record.
Definition cdr.c:3746
priority
static int priority
Definition cdr_beanstalkd.c:76
connected
static int connected
Definition cdr_pgsql.c:73
result
static PGresult * result
Definition cel_pgsql.c:84
config
static const char config[]
Definition chan_ooh323.c:111
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:3191
ast_str2cause
int ast_str2cause(const char *name) attribute_pure
Convert the string form of a cause code to a number.
Definition channel.c:626
ast_channel_name
const char * ast_channel_name(const struct ast_channel *chan)
ast_autoservice_stop
int ast_autoservice_stop(struct ast_channel *chan)
Stop servicing a channel for us...
Definition autoservice.c:266
ast_channel_appl_set
void ast_channel_appl_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:192
ast_channel_visible_indication_set
void ast_channel_visible_indication_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:353
ast_channel_get_device_name
int ast_channel_get_device_name(struct ast_channel *chan, char *device_name, size_t name_buffer_length)
Get a device name given its channel structure.
Definition channel.c:10584
ast_party_redirecting_init
void ast_party_redirecting_init(struct ast_party_redirecting *init)
Initialize the given redirecting structure.
Definition channel.c:2109
ast_call
int ast_call(struct ast_channel *chan, const char *addr, int timeout)
Make a call.
Definition channel.c:6496
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:11122
ast_channel_datastore_add
int ast_channel_datastore_add(struct ast_channel *chan, struct ast_datastore *datastore)
Add a datastore to a channel.
Definition channel.c:2376
ast_party_id_init
void ast_party_id_init(struct ast_party_id *init)
Initialize the given party id structure.
Definition channel.c:1744
ast_channel_connected_line_sub
int ast_channel_connected_line_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *connected_info, int frame)
Run a connected line interception subroutine and update a channel's connected line information.
Definition channel.c:10422
ast_party_number_init
void ast_party_number_init(struct ast_party_number *init)
Initialize the given number structure.
Definition channel.c:1631
ast_hangup
void ast_hangup(struct ast_channel *chan)
Hang up a channel.
Definition channel.c:2552
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:8392
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:6663
ast_party_connected_line_free
void ast_party_connected_line_free(struct ast_party_connected_line *doomed)
Destroy the connected line information contents.
Definition channel.c:2059
ast_channel_get_up_time_ms
int64_t ast_channel_get_up_time_ms(struct ast_channel *chan)
Obtain how long it has been since the channel was answered in ms.
Definition channel.c:2847
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:7424
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:3173
ast_senddigit
int ast_senddigit(struct ast_channel *chan, char digit, unsigned int duration)
Send a DTMF digit to a channel.
Definition channel.c:5009
ast_channel_lock
#define ast_channel_lock(chan)
Definition channel.h:2982
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:6755
ast_channel_data_set
void ast_channel_data_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:209
AST_FEATURE_AUTOMIXMON
@ AST_FEATURE_AUTOMIXMON
Definition channel.h:1089
AST_FEATURE_REDIRECT
@ AST_FEATURE_REDIRECT
Definition channel.h:1084
AST_FEATURE_PARKCALL
@ AST_FEATURE_PARKCALL
Definition channel.h:1088
AST_FEATURE_AUTOMON
@ AST_FEATURE_AUTOMON
Definition channel.h:1087
AST_FEATURE_DISCONNECT
@ AST_FEATURE_DISCONNECT
Definition channel.h:1085
ast_channel_redirecting
struct ast_party_redirecting * ast_channel_redirecting(struct ast_channel *chan)
Definition channel_internal_api.c:775
ast_channel_transfercapability
unsigned short ast_channel_transfercapability(const struct ast_channel *chan)
Definition channel_internal_api.c:373
ast_party_connected_line_copy
void ast_party_connected_line_copy(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src)
Copy the source connected line information to the destination connected line.
Definition channel.c:2018
ast_channel_flags
struct ast_flags * ast_channel_flags(struct ast_channel *chan)
Definition channel_internal_api.c:1127
ast_party_connected_line_set
void ast_party_connected_line_set(struct ast_party_connected_line *dest, const struct ast_party_connected_line *src, const struct ast_set_party_connected_line *update)
Set the connected line information based on another connected line source.
Definition channel.c:2041
ast_channel_priority
int ast_channel_priority(const struct ast_channel *chan)
Definition channel_internal_api.c:316
ast_channel_lock_both
#define ast_channel_lock_both(chan1, chan2)
Lock two channels.
Definition channel.h:2989
ast_channel_connected
struct ast_party_connected_line * ast_channel_connected(struct ast_channel *chan)
Definition channel_internal_api.c:759
ast_channel_datastore_inherit
int ast_channel_datastore_inherit(struct ast_channel *from, struct ast_channel *to)
Inherit datastores from a parent to a child.
Definition channel.c:2359
ast_channel_req_accountcodes
void ast_channel_req_accountcodes(struct ast_channel *chan, const struct ast_channel *requestor, enum ast_channel_requestor_relationship relationship)
Setup new channel accountcodes from the requestor channel after ast_request().
Definition channel.c:6469
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:2904
ast_check_hangup_locked
int ast_check_hangup_locked(struct ast_channel *chan)
Definition channel.c:460
ast_write
int ast_write(struct ast_channel *chan, struct ast_frame *frame)
Write a frame to a channel This function writes the given frame to the indicated channel.
Definition channel.c:5179
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:4290
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:9177
ast_channel_adsicpe
ast_channel_adsicpe
Definition channel.h:888
ast_party_caller_set_init
void ast_party_caller_set_init(struct ast_party_caller *init, const struct ast_party_caller *guide)
Initialize the given caller structure using the given guide for a set update operation.
Definition channel.c:1986
ast_party_id_set_init
void ast_party_id_set_init(struct ast_party_id *init, const struct ast_party_id *guide)
Initialize the given party id structure using the given guide for a set update operation.
Definition channel.c:1767
ast_channel_creationtime
struct timeval ast_channel_creationtime(struct ast_channel *chan)
Definition channel_internal_api.c:843
ast_channel_redirecting_sub
int ast_channel_redirecting_sub(struct ast_channel *autoservice_chan, struct ast_channel *sub_chan, const void *redirecting_info, int is_frame)
Run a redirecting interception subroutine and update a channel's redirecting information.
Definition channel.c:10467
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:6811
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:8869
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:6394
ast_channel_supports_html
int ast_channel_supports_html(struct ast_channel *channel)
Checks for HTML support on a channel.
Definition channel.c:6658
ast_check_hangup
int ast_check_hangup(struct ast_channel *chan)
Check to see if a channel is needing hang up.
Definition channel.c:446
ast_channel_get_stream_topology
struct ast_stream_topology * ast_channel_get_stream_topology(const struct ast_channel *chan)
Retrieve the topology of streams on a channel.
Definition channel_internal_api.c:1499
ast_channel_hangupcause
int ast_channel_hangupcause(const struct ast_channel *chan)
Definition channel_internal_api.c:298
ast_channel_get_duration_ms
int64_t ast_channel_get_duration_ms(struct ast_channel *chan)
Obtain how long it's been, in milliseconds, since the channel was created.
Definition channel.c:2832
ast_channel_dialed
struct ast_party_dialed * ast_channel_dialed(struct ast_channel *chan)
Definition channel_internal_api.c:771
ast_indicate_data
int ast_indicate_data(struct ast_channel *chan, int condition, const void *data, size_t datalen)
Indicates condition of channel, with payload.
Definition channel.c:4688
ast_channel_zone
struct ast_tone_zone * ast_channel_zone(const struct ast_channel *chan)
Definition channel_internal_api.c:612
ast_channel_set_flag
void ast_channel_set_flag(struct ast_channel *chan, unsigned int flag)
Set a flag on a channel.
Definition channel.c:11115
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:10368
ast_channel_timingfunc
ast_timing_func_t ast_channel_timingfunc(const struct ast_channel *chan)
Definition channel_internal_api.c:1098
AST_CHANNEL_NAME
#define AST_CHANNEL_NAME
Definition channel.h:173
ast_channel_whentohangup
struct timeval * ast_channel_whentohangup(struct ast_channel *chan)
Definition channel_internal_api.c:795
ast_party_number_free
void ast_party_number_free(struct ast_party_number *doomed)
Destroy the party number contents.
Definition channel.c:1678
ast_party_connected_line_init
void ast_party_connected_line_init(struct ast_party_connected_line *init)
Initialize the given connected line structure.
Definition channel.c:2009
ast_channel_sendurl
int ast_channel_sendurl(struct ast_channel *channel, const char *url)
Sends a URL on a given link Send URL on link.
Definition channel.c:6670
ast_channel_language
const char * ast_channel_language(const struct ast_channel *chan)
ast_cause2str
const char * ast_cause2str(int cause) attribute_pure
Gives the string form of a given cause code.
Definition channel.c:613
ast_party_redirecting_free
void ast_party_redirecting_free(struct ast_party_redirecting *doomed)
Destroy the redirecting information contents.
Definition channel.c:2166
ast_channel_context_set
void ast_channel_context_set(struct ast_channel *chan, const char *value)
Definition channel_internal_api.c:223
ast_channel_sched
struct ast_sched_context * ast_channel_sched(const struct ast_channel *chan)
Definition channel_internal_api.c:596
ast_connected_line_copy_from_caller
void ast_connected_line_copy_from_caller(struct ast_party_connected_line *dest, const struct ast_party_caller *src)
Copy the caller information to the connected line information.
Definition channel.c:8377
AST_FLAG_OUTGOING
@ AST_FLAG_OUTGOING
Definition channel.h:1019
AST_FLAG_END_DTMF_ONLY
@ AST_FLAG_END_DTMF_ONLY
Definition channel.h:1027
AST_FLAG_MOH
@ AST_FLAG_MOH
Definition channel.h:1011
ast_channel_call_forward
const char * ast_channel_call_forward(const struct ast_channel *chan)
ast_channel_stream
struct ast_filestream * ast_channel_stream(const struct ast_channel *chan)
Definition channel_internal_api.c:488
ast_pre_call
int ast_pre_call(struct ast_channel *chan, const char *sub_args)
Execute a Gosub call on the channel before a call is placed.
Definition channel.c:6479
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:7474
ast_channel_caller
struct ast_party_caller * ast_channel_caller(struct ast_channel *chan)
Definition channel_internal_api.c:755
ast_party_connected_line_set_init
void ast_party_connected_line_set_init(struct ast_party_connected_line *init, const struct ast_party_connected_line *guide)
Initialize the given connected line structure using the given guide for a set update operation.
Definition channel.c:2032
ast_channel_transfercapability_set
void ast_channel_transfercapability_set(struct ast_channel *chan, unsigned short value)
Definition channel_internal_api.c:377
ast_channel_priority_set
void ast_channel_priority_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:320
ast_channel_hangupcause_set
void ast_channel_hangupcause_set(struct ast_channel *chan, int value)
Definition channel_internal_api.c:302
ast_autoservice_chan_hangup_peer
void ast_autoservice_chan_hangup_peer(struct ast_channel *chan, struct ast_channel *peer)
Put chan into autoservice while hanging up peer.
Definition autoservice.c:349
ast_channel_whentohangup_set
void ast_channel_whentohangup_set(struct ast_channel *chan, struct timeval *value)
Definition channel_internal_api.c:835
ast_answer
int ast_answer(struct ast_channel *chan)
Answer a channel.
Definition channel.c:2817
ast_channel_adsicpe_set
void ast_channel_adsicpe_set(struct ast_channel *chan, enum ast_channel_adsicpe value)
Definition channel_internal_api.c:652
ast_channel_early_bridge
int ast_channel_early_bridge(struct ast_channel *c0, struct ast_channel *c1)
Bridge two channels together (early)
Definition channel.c:7464
ast_indicate
int ast_indicate(struct ast_channel *chan, int condition)
Indicates condition of channel.
Definition channel.c:4310
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:2983
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:2122
ast_channel_datastore_find
struct ast_datastore * ast_channel_datastore_find(struct ast_channel *chan, const struct ast_datastore_info *info, const char *uid)
Find a datastore on a channel.
Definition channel.c:2390
ast_channel_state
ast_channel_state
ast_channel states
Definition channelstate.h:35
AST_STATE_UP
@ AST_STATE_UP
Definition channelstate.h:42
ast_datastore_alloc
#define ast_datastore_alloc(info, uid)
Definition datastore.h:85
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:752
dsp.h
Convenient Signal Processing routines.
THRESHOLD_SILENCE
@ THRESHOLD_SILENCE
Definition dsp.h:75
ast_dsp_get_threshold_from_settings
int ast_dsp_get_threshold_from_settings(enum threshold which)
Get silence threshold from dsp.conf.
Definition dsp.c:2196
buf
char buf[BUFSIZE]
Definition eagi_proxy.c:66
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:1036
SCOPE_ENTER
#define SCOPE_ENTER(level,...)
Definition include/asterisk/logger.h:1008
ast_trace
#define ast_trace(level,...)
Definition include/asterisk/logger.h:999
ast_channel_publish_dial
void ast_channel_publish_dial(struct ast_channel *caller, struct ast_channel *peer, const char *dialstring, const char *dialstatus)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition stasis_channels.c:767
ast_channel_stage_snapshot_done
void ast_channel_stage_snapshot_done(struct ast_channel *chan)
Clear flag to indicate channel snapshot is being staged, and publish snapshot.
Definition stasis_channels.c:1071
ast_channel_publish_snapshot
void ast_channel_publish_snapshot(struct ast_channel *chan)
Publish a ast_channel_snapshot for a channel.
Definition stasis_channels.c:1083
ast_channel_stage_snapshot
void ast_channel_stage_snapshot(struct ast_channel *chan)
Set flag to indicate channel snapshot is being staged.
Definition stasis_channels.c:1066
ast_channel_publish_dial_forward
void ast_channel_publish_dial_forward(struct ast_channel *caller, struct ast_channel *peer, struct ast_channel *forwarded, const char *dialstring, const char *dialstatus, const char *forward)
Publish in the ast_channel_topic or ast_channel_topic_all topics a stasis message for the channels in...
Definition stasis_channels.c:717
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:2155
ast_app_group_set_channel
int ast_app_group_set_channel(struct ast_channel *chan, const char *data)
Set the group for a channel, splitting the provided data into group and category, if specified.
Definition main/app.c:2194
AST_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:1617
AST_STANDARD_APP_ARGS
#define AST_STANDARD_APP_ARGS(args, parse)
Performs the 'standard' argument separation process for an application.
Definition include/asterisk/app.h:1283
AST_APP_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:3072
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_DTMF_END
@ AST_FRAME_DTMF_END
Definition include/asterisk/frame.h:99
AST_FRAME_DTMF_BEGIN
@ AST_FRAME_DTMF_BEGIN
Definition include/asterisk/frame.h:125
AST_FRAME_CONTROL
@ AST_FRAME_CONTROL
Definition include/asterisk/frame.h:105
AST_CONTROL_SRCUPDATE
@ AST_CONTROL_SRCUPDATE
Definition include/asterisk/frame.h:297
AST_CONTROL_PROGRESS
@ AST_CONTROL_PROGRESS
Definition include/asterisk/frame.h:291
AST_CONTROL_OFFHOOK
@ AST_CONTROL_OFFHOOK
Definition include/asterisk/frame.h:284
AST_CONTROL_UNHOLD
@ AST_CONTROL_UNHOLD
Definition include/asterisk/frame.h:294
AST_CONTROL_VIDUPDATE
@ AST_CONTROL_VIDUPDATE
Definition include/asterisk/frame.h:295
AST_CONTROL_PROCEEDING
@ AST_CONTROL_PROCEEDING
Definition include/asterisk/frame.h:292
AST_CONTROL_REDIRECTING
@ AST_CONTROL_REDIRECTING
Definition include/asterisk/frame.h:300
AST_CONTROL_CONGESTION
@ AST_CONTROL_CONGESTION
Definition include/asterisk/frame.h:285
AST_CONTROL_PLAYBACK_BEGIN
@ AST_CONTROL_PLAYBACK_BEGIN
Definition include/asterisk/frame.h:338
AST_CONTROL_ANSWER
@ AST_CONTROL_ANSWER
Definition include/asterisk/frame.h:281
AST_CONTROL_RINGING
@ AST_CONTROL_RINGING
Definition include/asterisk/frame.h:280
AST_CONTROL_HANGUP
@ AST_CONTROL_HANGUP
Definition include/asterisk/frame.h:278
AST_CONTROL_CONNECTED_LINE
@ AST_CONTROL_CONNECTED_LINE
Definition include/asterisk/frame.h:299
AST_CONTROL_FLASH
@ AST_CONTROL_FLASH
Definition include/asterisk/frame.h:286
AST_CONTROL_SRCCHANGE
@ AST_CONTROL_SRCCHANGE
Definition include/asterisk/frame.h:303
AST_CONTROL_PVT_CAUSE_CODE
@ AST_CONTROL_PVT_CAUSE_CODE
Definition include/asterisk/frame.h:310
ast_debug
#define ast_debug(level,...)
Log a DEBUG message.
Definition include/asterisk/logger.h:456
LOG_ERROR
#define LOG_ERROR
Definition include/asterisk/logger.h:291
ast_verb
#define ast_verb(level,...)
Definition include/asterisk/logger.h:467
LOG_NOTICE
#define LOG_NOTICE
Definition include/asterisk/logger.h:269
LOG_WARNING
#define LOG_WARNING
Definition include/asterisk/logger.h:280
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:404
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:7818
ast_moh_stop
void ast_moh_stop(struct ast_channel *chan)
Turn off music on hold on a given channel.
Definition channel.c:7828
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:4211
AST_PBX_INCOMPLETE
#define AST_PBX_INCOMPLETE
Definition pbx.h:51
pbx_builtin_setvar_helper
int pbx_builtin_setvar_helper(struct ast_channel *chan, const char *name, const char *value)
Add a variable to the channel variable stack, removing the most recently set value for the same name.
Definition pbx_variables.c:1233
ast_goto_if_exists
int ast_goto_if_exists(struct ast_channel *chan, const char *context, const char *exten, int priority)
Definition pbx.c:8825
ast_pbx_start
enum ast_pbx_result ast_pbx_start(struct ast_channel *c)
Create a new thread and start the PBX.
Definition pbx.c:4744
ast_get_hint
int ast_get_hint(char *hint, int hintsize, char *name, int namesize, struct ast_channel *c, const char *context, const char *exten)
If an extension hint exists, return non-zero.
Definition pbx.c:4173
ast_parseable_goto
int ast_parseable_goto(struct ast_channel *chan, const char *goto_string)
Definition pbx.c:8910
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
args
static struct @521 args
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_strlen
size_t attribute_pure ast_str_strlen(const struct ast_str *buf)
Returns the current length of the string stored within buf.
Definition strings.h:730
S_COR
#define S_COR(a, b, c)
returns the equivalent of logic or for strings, with an additional boolean check: second one if not e...
Definition strings.h:87
ast_strlen_zero
static force_inline int attribute_pure ast_strlen_zero(const char *s)
Definition strings.h:65
ast_str_tmp
#define ast_str_tmp(init_len, __expr)
Provides a temporary ast_str and returns a copy of its buffer.
Definition strings.h:1189
ast_str_alloca
#define ast_str_alloca(init_len)
Definition strings.h:848
ast_str_reset
void ast_str_reset(struct ast_str *buf)
Reset the content of a dynamic string. Useful before a series of ast_str_append.
Definition strings.h:693
ast_str_buffer
char *attribute_pure ast_str_buffer(const struct ast_str *buf)
Returns the string buffer within the ast_str buf.
Definition strings.h:761
ast_copy_string
void ast_copy_string(char *dst, const char *src, size_t size)
Size-limited null-terminating string copy.
Definition strings.h:425
ast_strip
char * ast_strip(char *s)
Strip leading/trailing whitespace from a string.
Definition strings.h:223
ast_aoc_decoded
Definition aoc.c:214
ast_aoc_encoded
Definition aoc.c:206
ast_bridge_config
bridge configuration
Definition channel.h:1096
ast_bridge_config::end_bridge_callback_data
void * end_bridge_callback_data
Definition channel.h:1111
ast_channel
Main Channel structure associated with a channel.
Definition channel_private.h:48
ast_channel::exten
char exten[AST_MAX_EXTENSION]
Definition channel_private.h:184
ast_channel::data
const char * data
Definition channel_private.h:58
ast_channel::tech
const struct ast_channel_tech * tech
Definition channel_private.h:49
ast_channel::context
char context[AST_MAX_CONTEXT]
Definition channel_private.h:183
ast_datastore_info
Structure for a data store type.
Definition datastore.h:31
ast_datastore_info::type
const char * type
Definition datastore.h:32
ast_datastore
Structure for a data store object.
Definition datastore.h:64
ast_datastore::data
void * data
Definition datastore.h:66
ast_flags64
Structure used to handle a large number of boolean flags == used only in app_dial?
Definition utils.h:225
ast_flags64::flags
uint64_t flags
Definition utils.h:226
ast_frame_subclass::topology
struct ast_stream_topology * topology
Definition include/asterisk/frame.h:157
ast_frame_subclass::integer
int integer
Definition include/asterisk/frame.h:152
ast_frame
Data structure associated with a single frame of data.
Definition include/asterisk/frame.h:165
ast_frame::subclass
struct ast_frame_subclass subclass
Definition include/asterisk/frame.h:169
ast_frame::uint32
uint32_t uint32
Definition include/asterisk/frame.h:183
ast_frame::frametype
enum ast_frame_type frametype
Definition include/asterisk/frame.h:167
ast_frame::data
union ast_frame::@235 data
ast_frame::datalen
int datalen
Definition include/asterisk/frame.h:171
ast_frame::ptr
void * ptr
Definition include/asterisk/frame.h:183
ast_party_caller
Caller Party information.
Definition channel.h:420
ast_party_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:870
cause_args::congestion
int congestion
Definition app_dial.c:873
cause_args::nochan
int nochan
Definition app_dial.c:874
cause_args::chan
struct ast_channel * chan
Definition app_dial.c:871
cause_args::busy
int busy
Definition app_dial.c:872
chanlist
List of channel drivers.
Definition app_dial.c:804
chanlist::number
const char * number
Definition app_dial.c:812
chanlist::interface
const char * interface
Definition app_dial.c:808
chanlist::aoc_s_rate_list
struct ast_aoc_decoded * aoc_s_rate_list
Definition app_dial.c:820
chanlist::connected
struct ast_party_connected_line connected
Definition app_dial.c:817
chanlist::node
struct chanlist::@17 node
chanlist::orig_chan_name
char * orig_chan_name
Definition app_dial.c:814
chanlist::stuff
char stuff[0]
Definition app_dial.c:822
chanlist::chan
struct ast_channel * chan
Definition app_dial.c:806
chanlist::flags
uint64_t flags
Definition app_dial.c:815
chanlist::tech
const char * tech
Definition app_dial.c:810
chanlist::pending_connected_update
unsigned int pending_connected_update
Definition app_dial.c:819
chans
Definition astman.c:88
dial_head
Definition app_dial.c:825
extension
structure to hold extensions
Definition res_phoneprov.c:293
max_forwards
Channel datastore data for max forwards.
Definition max_forwards.c:29
number
Number structure.
Definition app_followme.c:157
oprmode::peer
struct ast_channel * peer
Definition include/asterisk/frame.h:527
privacy_args
Definition app_dial.c:1147
privacy_args::status
char status[256]
Definition app_dial.c:1152
privacy_args::privdb_val
int privdb_val
Definition app_dial.c:1149
privacy_args::canceled
int canceled
Definition app_dial.c:1153
privacy_args::privcid
char privcid[256]
Definition app_dial.c:1150
privacy_args::sentringing
int sentringing
Definition app_dial.c:1148
privacy_args::privintro
char privintro[1024]
Definition app_dial.c:1151
done
int done
Definition test_amihooks.c:48
options
static struct test_options options
Definition test_http_media_cache.c:63
c
static struct test_val c
Definition test_linkedlists.c:48
ast_tvzero
int ast_tvzero(const struct timeval t)
Returns true if the argument is 0,0.
Definition time.h:117
ast_remaining_ms
int ast_remaining_ms(struct timeval start, int max_ms)
Calculate remaining milliseconds given a starting timestamp and upper bound.
Definition utils.c:2315
ast_tvadd
struct timeval ast_tvadd(struct timeval a, struct timeval b)
Returns the sum of two timevals a + b.
Definition extconf.c:2280
ast_tvdiff_ms
int64_t ast_tvdiff_ms(struct timeval end, struct timeval start)
Computes the difference (in milliseconds) between two struct timeval instances.
Definition time.h:107
ast_tvnow
struct timeval ast_tvnow(void)
Returns current timeval. Meant to replace calls to gettimeofday().
Definition time.h:159
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:64
ast_set2_flag64
#define ast_set2_flag64(p, value, flag)
Definition utils.h:171
ast_test_flag64
#define ast_test_flag64(p, flag)
Definition utils.h:140
ast_clear_flag64
#define ast_clear_flag64(p, flag)
Definition utils.h:154
ast_clear_flag
#define ast_clear_flag(p, flag)
Definition utils.h:78
ast_mkdir
int ast_mkdir(const char *path, int mode)
Recursively create directory path.
Definition utils.c:2513
ast_copy_flags64
#define ast_copy_flags64(dest, src, flagz)
Definition utils.h:161
ast_set_flag64
#define ast_set_flag64(p, flag)
Definition utils.h:147
ast_set_flag
#define ast_set_flag(p, flag)
Definition utils.h:71
ast_replace_subargument_delimiter
void ast_replace_subargument_delimiter(char *s)
Replace '^' in a string with ','.
Definition utils.c:2377
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