Fixes to include signal.h
[asterisk/asterisk.git] / apps / app_dial.c
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 ASTERISK_REGISTER_FILE()
36
37 #include <sys/time.h>
38 #include <signal.h>
39 #include <sys/stat.h>
40 #include <netinet/in.h>
41
42 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
43 #include "asterisk/lock.h"
44 #include "asterisk/file.h"
45 #include "asterisk/channel.h"
46 #include "asterisk/pbx.h"
47 #include "asterisk/module.h"
48 #include "asterisk/translate.h"
49 #include "asterisk/say.h"
50 #include "asterisk/config.h"
51 #include "asterisk/features.h"
52 #include "asterisk/musiconhold.h"
53 #include "asterisk/callerid.h"
54 #include "asterisk/utils.h"
55 #include "asterisk/app.h"
56 #include "asterisk/causes.h"
57 #include "asterisk/rtp_engine.h"
58 #include "asterisk/manager.h"
59 #include "asterisk/privacy.h"
60 #include "asterisk/stringfields.h"
61 #include "asterisk/dsp.h"
62 #include "asterisk/aoc.h"
63 #include "asterisk/ccss.h"
64 #include "asterisk/indications.h"
65 #include "asterisk/framehook.h"
66 #include "asterisk/dial.h"
67 #include "asterisk/stasis_channels.h"
68 #include "asterisk/bridge_after.h"
69 #include "asterisk/features_config.h"
70 #include "asterisk/max_forwards.h"
71
72 /*** DOCUMENTATION
73         <application name="Dial" language="en_US">
74                 <synopsis>
75                         Attempt to connect to another device or endpoint and bridge the call.
76                 </synopsis>
77                 <syntax>
78                         <parameter name="Technology/Resource" required="true" argsep="&amp;">
79                                 <argument name="Technology/Resource" required="true">
80                                         <para>Specification of the device(s) to dial.  These must be in the format of
81                                         <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
82                                         represents a particular channel driver, and <replaceable>Resource</replaceable>
83                                         represents a resource available to that particular channel driver.</para>
84                                 </argument>
85                                 <argument name="Technology2/Resource2" required="false" multiple="true">
86                                         <para>Optional extra devices to dial in parallel</para>
87                                         <para>If you need more than one enter them as
88                                         Technology2/Resource2&amp;Technology3/Resource3&amp;.....</para>
89                                 </argument>
90                         </parameter>
91                         <parameter name="timeout" required="false">
92                                 <para>Specifies the number of seconds we attempt to dial the specified devices.</para>
93                                 <para>If not specified, this defaults to 136 years.</para>
94                         </parameter>
95                         <parameter name="options" required="false">
96                                 <optionlist>
97                                 <option name="A">
98                                         <argument name="x" required="true">
99                                                 <para>The file to play to the called party</para>
100                                         </argument>
101                                         <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
102                                 </option>
103                                 <option name="a">
104                                         <para>Immediately answer the calling channel when the called channel answers in
105                                         all cases. Normally, the calling channel is answered when the called channel
106                                         answers, but when options such as A() and M() are used, the calling channel is
107                                         not answered until all actions on the called channel (such as playing an
108                                         announcement) are completed.  This option can be used to answer the calling
109                                         channel before doing anything on the called channel. You will rarely need to use
110                                         this option, the default behavior is adequate in most cases.</para>
111                                 </option>
112                                 <option name="b" argsep="^">
113                                         <para>Before initiating an outgoing call, Gosub to the specified
114                                         location using the newly created channel.  The Gosub will be
115                                         executed for each destination channel.</para>
116                                         <argument name="context" required="false" />
117                                         <argument name="exten" required="false" />
118                                         <argument name="priority" required="true" hasparams="optional" argsep="^">
119                                                 <argument name="arg1" multiple="true" required="true" />
120                                                 <argument name="argN" />
121                                         </argument>
122                                 </option>
123                                 <option name="B" argsep="^">
124                                         <para>Before initiating the outgoing call(s), Gosub to the specified
125                                         location using the current channel.</para>
126                                         <argument name="context" required="false" />
127                                         <argument name="exten" required="false" />
128                                         <argument name="priority" required="true" hasparams="optional" argsep="^">
129                                                 <argument name="arg1" multiple="true" required="true" />
130                                                 <argument name="argN" />
131                                         </argument>
132                                 </option>
133                                 <option name="C">
134                                         <para>Reset the call detail record (CDR) for this call.</para>
135                                 </option>
136                                 <option name="c">
137                                         <para>If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere'</para>
138                                 </option>
139                                 <option name="d">
140                                         <para>Allow the calling user to dial a 1 digit extension while waiting for
141                                         a call to be answered. Exit to that extension if it exists in the
142                                         current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
143                                         if it exists.</para>
144                                         <note>
145                                                 <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
146                                                 connected.  If you wish to use this option with these phones, you
147                                                 can use the <literal>Answer</literal> application before dialing.</para>
148                                         </note>
149                                 </option>
150                                 <option name="D" argsep=":">
151                                         <argument name="called" />
152                                         <argument name="calling" />
153                                         <argument name="progress" />
154                                         <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
155                                         party has answered, but before the call gets bridged.  The
156                                         <replaceable>called</replaceable> DTMF string is sent to the called party, and the
157                                         <replaceable>calling</replaceable> DTMF string is sent to the calling party.  Both arguments
158                                         can be used alone.  If <replaceable>progress</replaceable> is specified, its DTMF is sent
159                                         to the called party immediately after receiving a PROGRESS message.</para>
160                                         <para>See SendDTMF for valid digits.</para>
161                                 </option>
162                                 <option name="e">
163                                         <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
164                                 </option>
165                                 <option name="f">
166                                         <argument name="x" required="false" />
167                                         <para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
168                                         deflection to the dialplan extension of this Dial() using a dialplan <literal>hint</literal>.
169                                         For example, some PSTNs do not allow CallerID to be set to anything
170                                         other than the numbers assigned to you.
171                                         If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
172                                 </option>
173                                 <option name="F" argsep="^">
174                                         <argument name="context" required="false" />
175                                         <argument name="exten" required="false" />
176                                         <argument name="priority" required="true" />
177                                         <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
178                                         to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
179                                         <note>
180                                                 <para>Any channel variables you want the called channel to inherit from the caller channel must be
181                                                 prefixed with one or two underbars ('_').</para>
182                                         </note>
183                                 </option>
184                                 <option name="F">
185                                         <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
186                                         and <emphasis>start</emphasis> execution at that location.</para>
187                                         <note>
188                                                 <para>Any channel variables you want the called channel to inherit from the caller channel must be
189                                                 prefixed with one or two underbars ('_').</para>
190                                         </note>
191                                         <note>
192                                                 <para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
193                                         </note>
194                                 </option>
195                                 <option name="g">
196                                         <para>Proceed with dialplan execution at the next priority in the current extension if the
197                                         destination channel hangs up.</para>
198                                 </option>
199                                 <option name="G" argsep="^">
200                                         <argument name="context" required="false" />
201                                         <argument name="exten" required="false" />
202                                         <argument name="priority" required="true" />
203                                         <para>If the call is answered, transfer the calling party to
204                                         the specified <replaceable>priority</replaceable> and the called party to the specified
205                                         <replaceable>priority</replaceable> plus one.</para>
206                                         <note>
207                                                 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
208                                         </note>
209                                 </option>
210                                 <option name="h">
211                                         <para>Allow the called party to hang up by sending the DTMF sequence
212                                         defined for disconnect in <filename>features.conf</filename>.</para>
213                                 </option>
214                                 <option name="H">
215                                         <para>Allow the calling party to hang up by sending the DTMF sequence
216                                         defined for disconnect in <filename>features.conf</filename>.</para>
217                                         <note>
218                                                 <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
219                                                 connected.  If you wish to allow DTMF disconnect before the dialed
220                                                 party answers with these phones, you can use the <literal>Answer</literal>
221                                                 application before dialing.</para>
222                                         </note>
223                                 </option>
224                                 <option name="i">
225                                         <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
226                                 </option>
227                                 <option name="I">
228                                         <para>Asterisk will ignore any connected line update requests or any redirecting party
229                                         update requests it may receive on this dial attempt.</para>
230                                 </option>
231                                 <option name="k">
232                                         <para>Allow the called party to enable parking of the call by sending
233                                         the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
234                                 </option>
235                                 <option name="K">
236                                         <para>Allow the calling party to enable parking of the call by sending
237                                         the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
238                                 </option>
239                                 <option name="L" argsep=":">
240                                         <argument name="x" required="true">
241                                                 <para>Maximum call time, in milliseconds</para>
242                                         </argument>
243                                         <argument name="y">
244                                                 <para>Warning time, in milliseconds</para>
245                                         </argument>
246                                         <argument name="z">
247                                                 <para>Repeat time, in milliseconds</para>
248                                         </argument>
249                                         <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
250                                         left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
251                                         <para>This option is affected by the following variables:</para>
252                                         <variablelist>
253                                                 <variable name="LIMIT_PLAYAUDIO_CALLER">
254                                                         <value name="yes" default="true" />
255                                                         <value name="no" />
256                                                         <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
257                                                 </variable>
258                                                 <variable name="LIMIT_PLAYAUDIO_CALLEE">
259                                                         <value name="yes" />
260                                                         <value name="no" default="true"/>
261                                                         <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
262                                                 </variable>
263                                                 <variable name="LIMIT_TIMEOUT_FILE">
264                                                         <value name="filename"/>
265                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
266                                                         If not set, the time remaining will be announced.</para>
267                                                 </variable>
268                                                 <variable name="LIMIT_CONNECT_FILE">
269                                                         <value name="filename"/>
270                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
271                                                         If not set, the time remaining will be announced.</para>
272                                                 </variable>
273                                                 <variable name="LIMIT_WARNING_FILE">
274                                                         <value name="filename"/>
275                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
276                                                         a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
277                                                 </variable>
278                                         </variablelist>
279                                 </option>
280                                 <option name="m">
281                                         <argument name="class" required="false"/>
282                                         <para>Provide hold music to the calling party until a requested
283                                         channel answers. A specific music on hold <replaceable>class</replaceable>
284                                         (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
285                                 </option>
286                                 <option name="M" argsep="^">
287                                         <argument name="macro" required="true">
288                                                 <para>Name of the macro that should be executed.</para>
289                                         </argument>
290                                         <argument name="arg" multiple="true">
291                                                 <para>Macro arguments</para>
292                                         </argument>
293                                         <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
294                                         before connecting to the calling channel. Arguments can be specified to the Macro
295                                         using <literal>^</literal> as a delimiter. The macro can set the variable
296                                         <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
297                                         finished executing:</para>
298                                         <variablelist>
299                                                 <variable name="MACRO_RESULT">
300                                                         <para>If set, this action will be taken after the macro finished executing.</para>
301                                                         <value name="ABORT">
302                                                                 Hangup both legs of the call
303                                                         </value>
304                                                         <value name="CONGESTION">
305                                                                 Behave as if line congestion was encountered
306                                                         </value>
307                                                         <value name="BUSY">
308                                                                 Behave as if a busy signal was encountered
309                                                         </value>
310                                                         <value name="CONTINUE">
311                                                                 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
312                                                         </value>
313                                                         <value name="GOTO:[[&lt;context&gt;^]&lt;exten&gt;^]&lt;priority&gt;">
314                                                                 Transfer the call to the specified destination.
315                                                         </value>
316                                                 </variable>
317                                         </variablelist>
318                                         <note>
319                                                 <para>You cannot use any additional action post answer options in conjunction
320                                                 with this option. Also, pbx services are run on the peer (called) channel,
321                                                 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
322                                         </note>
323                                         <warning><para>Be aware of the limitations that macros have, specifically with regards to use of
324                                         the <literal>WaitExten</literal> application. For more information, see the documentation for
325                                         Macro()</para></warning>
326                                 </option>
327                                 <option name="n">
328                                         <argument name="delete">
329                                                 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
330                                                 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
331                                                 yet answered.</para>
332                                                 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
333                                                 always be deleted.</para>
334                                         </argument>
335                                         <para>This option is a modifier for the call screening/privacy mode. (See the
336                                         <literal>p</literal> and <literal>P</literal> options.) It specifies
337                                         that no introductions are to be saved in the <directory>priv-callerintros</directory>
338                                         directory.</para>
339                                 </option>
340                                 <option name="N">
341                                         <para>This option is a modifier for the call screening/privacy mode. It specifies
342                                         that if Caller*ID is present, do not screen the call.</para>
343                                 </option>
344                                 <option name="o">
345                                         <argument name="x" required="false" />
346                                         <para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
347                                         <emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
348                                         This was the behavior of Asterisk 1.0 and earlier.
349                                         If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
350                                         Note that o(${CALLERID(all)}) is similar to option o without the parameter.</para>
351                                 </option>
352                                 <option name="O">
353                                         <argument name="mode">
354                                                 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
355                                                 the originator hanging up will cause the phone to ring back immediately.</para>
356                                                 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
357                                                 flashes the trunk, it will ring their phone back.</para>
358                                         </argument>
359                                         <para>Enables <emphasis>operator services</emphasis> mode.  This option only
360                                         works when bridging a DAHDI channel to another DAHDI channel
361                                         only. if specified on non-DAHDI interfaces, it will be ignored.
362                                         When the destination answers (presumably an operator services
363                                         station), the originator no longer has control of their line.
364                                         They may hang up, but the switch will not release their line
365                                         until the destination party (the operator) hangs up.</para>
366                                 </option>
367                                 <option name="p">
368                                         <para>This option enables screening mode. This is basically Privacy mode
369                                         without memory.</para>
370                                 </option>
371                                 <option name="P">
372                                         <argument name="x" />
373                                         <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
374                                         it is provided. The current extension is used if a database family/key is not specified.</para>
375                                 </option>
376                                 <option name="r">
377                                         <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
378                                         party until the called channel has answered.</para>
379                                         <argument name="tone" required="false">
380                                                 <para>Indicate progress to calling party. Send audio 'tone' from the indications.conf tonezone currently in use.</para>
381                                         </argument>
382                                 </option>
383                                 <option name="R">
384                                         <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. 
385                                         Allow interruption of the ringback if early media is received on the channel.</para>
386                                 </option>
387                                 <option name="S">
388                                         <argument name="x" required="true" />
389                                         <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
390                                         answered the call.</para>
391                                 </option>
392                                 <option name="s">
393                                         <argument name="x" required="true" />
394                                         <para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable>.</para>
395                                         <para>Works with the f option.</para>
396                                 </option>
397                                 <option name="t">
398                                         <para>Allow the called party to transfer the calling party by sending the
399                                         DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
400                                         transfers initiated by other methods.</para>
401                                 </option>
402                                 <option name="T">
403                                         <para>Allow the calling party to transfer the called party by sending the
404                                         DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
405                                         transfers initiated by other methods.</para>
406                                 </option>
407                                 <option name="U" argsep="^">
408                                         <argument name="x" required="true">
409                                                 <para>Name of the subroutine to execute via Gosub</para>
410                                         </argument>
411                                         <argument name="arg" multiple="true" required="false">
412                                                 <para>Arguments for the Gosub routine</para>
413                                         </argument>
414                                         <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
415                                         to the calling channel. Arguments can be specified to the Gosub
416                                         using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
417                                         <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub 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                                         <note>
439                                                 <para>You cannot use any additional action post answer options in conjunction
440                                                 with this option. Also, pbx services are run on the peer (called) channel,
441                                                 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
442                                         </note>
443                                 </option>
444                                 <option name="u">
445                                         <argument name = "x" required="true">
446                                                 <para>Force the outgoing callerid presentation indicator parameter to be set
447                                                 to one of the values passed in <replaceable>x</replaceable>:
448                                                 <literal>allowed_not_screened</literal>
449                                                 <literal>allowed_passed_screen</literal>
450                                                 <literal>allowed_failed_screen</literal>
451                                                 <literal>allowed</literal>
452                                                 <literal>prohib_not_screened</literal>
453                                                 <literal>prohib_passed_screen</literal>
454                                                 <literal>prohib_failed_screen</literal>
455                                                 <literal>prohib</literal>
456                                                 <literal>unavailable</literal></para>
457                                         </argument>
458                                         <para>Works with the f option.</para>
459                                 </option>
460                                 <option name="w">
461                                         <para>Allow the called party to enable recording of the call by sending
462                                         the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
463                                 </option>
464                                 <option name="W">
465                                         <para>Allow the calling party to enable recording of the call by sending
466                                         the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
467                                 </option>
468                                 <option name="x">
469                                         <para>Allow the called party to enable recording of the call by sending
470                                         the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
471                                 </option>
472                                 <option name="X">
473                                         <para>Allow the calling party to enable recording of the call by sending
474                                         the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
475                                 </option>
476                                 <option name="z">
477                                         <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
478                                 </option>
479                                 </optionlist>
480                         </parameter>
481                         <parameter name="URL">
482                                 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
483                         </parameter>
484                 </syntax>
485                 <description>
486                         <para>This application will place calls to one or more specified channels. As soon
487                         as one of the requested channels answers, the originating channel will be
488                         answered, if it has not already been answered. These two channels will then
489                         be active in a bridged call. All other channels that were requested will then
490                         be hung up.</para>
491
492                         <para>Unless there is a timeout specified, the Dial application will wait
493                         indefinitely until one of the called channels answers, the user hangs up, or
494                         if all of the called channels are busy or unavailable. Dialplan execution will
495                         continue if no requested channels can be called, or if the timeout expires.
496                         This application will report normal termination if the originating channel
497                         hangs up, or if the call is bridged and either of the parties in the bridge
498                         ends the call.</para>
499                         <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
500                         application will be put into that group (as in Set(GROUP()=...).
501                         If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
502                         application will be put into that group (as in Set(GROUP()=...). Unlike <variable>OUTBOUND_GROUP</variable>,
503                         however, the variable will be unset after use.</para>
504
505                         <para>This application sets the following channel variables:</para>
506                         <variablelist>
507                                 <variable name="DIALEDTIME">
508                                         <para>This is the time from dialing a channel until when it is disconnected.</para>
509                                 </variable>
510                                 <variable name="ANSWEREDTIME">
511                                         <para>This is the amount of time for actual call.</para>
512                                 </variable>
513                                 <variable name="DIALSTATUS">
514                                         <para>This is the status of the call</para>
515                                         <value name="CHANUNAVAIL" />
516                                         <value name="CONGESTION" />
517                                         <value name="NOANSWER" />
518                                         <value name="BUSY" />
519                                         <value name="ANSWER" />
520                                         <value name="CANCEL" />
521                                         <value name="DONTCALL">
522                                                 For the Privacy and Screening Modes.
523                                                 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
524                                         </value>
525                                         <value name="TORTURE">
526                                                 For the Privacy and Screening Modes.
527                                                 Will be set if the called party chooses to send the calling party to the 'torture' script.
528                                         </value>
529                                         <value name="INVALIDARGS" />
530                                 </variable>
531                         </variablelist>
532                 </description>
533         </application>
534         <application name="RetryDial" language="en_US">
535                 <synopsis>
536                         Place a call, retrying on failure allowing an optional exit extension.
537                 </synopsis>
538                 <syntax>
539                         <parameter name="announce" required="true">
540                                 <para>Filename of sound that will be played when no channel can be reached</para>
541                         </parameter>
542                         <parameter name="sleep" required="true">
543                                 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
544                         </parameter>
545                         <parameter name="retries" required="true">
546                                 <para>Number of retries</para>
547                                 <para>When this is reached flow will continue at the next priority in the dialplan</para>
548                         </parameter>
549                         <parameter name="dialargs" required="true">
550                                 <para>Same format as arguments provided to the Dial application</para>
551                         </parameter>
552                 </syntax>
553                 <description>
554                         <para>This application will attempt to place a call using the normal Dial application.
555                         If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
556                         Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
557                         After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
558                         If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
559                         While waiting to retry a call, a 1 digit extension may be dialed. If that
560                         extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
561                         one, The call will jump to that extension immediately.
562                         The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
563                         to the Dial application.</para>
564                 </description>
565         </application>
566  ***/
567
568 static const char app[] = "Dial";
569 static const char rapp[] = "RetryDial";
570
571 enum {
572         OPT_ANNOUNCE =          (1 << 0),
573         OPT_RESETCDR =          (1 << 1),
574         OPT_DTMF_EXIT =         (1 << 2),
575         OPT_SENDDTMF =          (1 << 3),
576         OPT_FORCECLID =         (1 << 4),
577         OPT_GO_ON =             (1 << 5),
578         OPT_CALLEE_HANGUP =     (1 << 6),
579         OPT_CALLER_HANGUP =     (1 << 7),
580         OPT_ORIGINAL_CLID =     (1 << 8),
581         OPT_DURATION_LIMIT =    (1 << 9),
582         OPT_MUSICBACK =         (1 << 10),
583         OPT_CALLEE_MACRO =      (1 << 11),
584         OPT_SCREEN_NOINTRO =    (1 << 12),
585         OPT_SCREEN_NOCALLERID = (1 << 13),
586         OPT_IGNORE_CONNECTEDLINE = (1 << 14),
587         OPT_SCREENING =         (1 << 15),
588         OPT_PRIVACY =           (1 << 16),
589         OPT_RINGBACK =          (1 << 17),
590         OPT_DURATION_STOP =     (1 << 18),
591         OPT_CALLEE_TRANSFER =   (1 << 19),
592         OPT_CALLER_TRANSFER =   (1 << 20),
593         OPT_CALLEE_MONITOR =    (1 << 21),
594         OPT_CALLER_MONITOR =    (1 << 22),
595         OPT_GOTO =              (1 << 23),
596         OPT_OPERMODE =          (1 << 24),
597         OPT_CALLEE_PARK =       (1 << 25),
598         OPT_CALLER_PARK =       (1 << 26),
599         OPT_IGNORE_FORWARDING = (1 << 27),
600         OPT_CALLEE_GOSUB =      (1 << 28),
601         OPT_CALLEE_MIXMONITOR = (1 << 29),
602         OPT_CALLER_MIXMONITOR = (1 << 30),
603 };
604
605 /* flags are now 64 bits, so keep it up! */
606 #define DIAL_STILLGOING      (1LLU << 31)
607 #define DIAL_NOFORWARDHTML   (1LLU << 32)
608 #define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */
609 #define OPT_CANCEL_ELSEWHERE (1LLU << 34)
610 #define OPT_PEER_H           (1LLU << 35)
611 #define OPT_CALLEE_GO_ON     (1LLU << 36)
612 #define OPT_CANCEL_TIMEOUT   (1LLU << 37)
613 #define OPT_FORCE_CID_TAG    (1LLU << 38)
614 #define OPT_FORCE_CID_PRES   (1LLU << 39)
615 #define OPT_CALLER_ANSWER    (1LLU << 40)
616 #define OPT_PREDIAL_CALLEE   (1LLU << 41)
617 #define OPT_PREDIAL_CALLER   (1LLU << 42)
618 #define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
619
620 enum {
621         OPT_ARG_ANNOUNCE = 0,
622         OPT_ARG_SENDDTMF,
623         OPT_ARG_GOTO,
624         OPT_ARG_DURATION_LIMIT,
625         OPT_ARG_MUSICBACK,
626         OPT_ARG_CALLEE_MACRO,
627         OPT_ARG_RINGBACK,
628         OPT_ARG_CALLEE_GOSUB,
629         OPT_ARG_CALLEE_GO_ON,
630         OPT_ARG_PRIVACY,
631         OPT_ARG_DURATION_STOP,
632         OPT_ARG_OPERMODE,
633         OPT_ARG_SCREEN_NOINTRO,
634         OPT_ARG_ORIGINAL_CLID,
635         OPT_ARG_FORCECLID,
636         OPT_ARG_FORCE_CID_TAG,
637         OPT_ARG_FORCE_CID_PRES,
638         OPT_ARG_PREDIAL_CALLEE,
639         OPT_ARG_PREDIAL_CALLER,
640         /* note: this entry _MUST_ be the last one in the enum */
641         OPT_ARG_ARRAY_SIZE
642 };
643
644 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
645         AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
646         AST_APP_OPTION('a', OPT_CALLER_ANSWER),
647         AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
648         AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
649         AST_APP_OPTION('C', OPT_RESETCDR),
650         AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
651         AST_APP_OPTION('d', OPT_DTMF_EXIT),
652         AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
653         AST_APP_OPTION('e', OPT_PEER_H),
654         AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
655         AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
656         AST_APP_OPTION('g', OPT_GO_ON),
657         AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
658         AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
659         AST_APP_OPTION('H', OPT_CALLER_HANGUP),
660         AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
661         AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
662         AST_APP_OPTION('k', OPT_CALLEE_PARK),
663         AST_APP_OPTION('K', OPT_CALLER_PARK),
664         AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
665         AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
666         AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
667         AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
668         AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
669         AST_APP_OPTION_ARG('o', OPT_ORIGINAL_CLID, OPT_ARG_ORIGINAL_CLID),
670         AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
671         AST_APP_OPTION('p', OPT_SCREENING),
672         AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
673         AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
674         AST_APP_OPTION('R', OPT_RING_WITH_EARLY_MEDIA),
675         AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
676         AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
677         AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
678         AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
679         AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
680         AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
681         AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
682         AST_APP_OPTION('W', OPT_CALLER_MONITOR),
683         AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
684         AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
685         AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
686 END_OPTIONS );
687
688 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
689         OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
690         OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK |  \
691         OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
692         !ast_channel_audiohooks(chan) && !ast_channel_audiohooks(peer) && \
693         ast_framehook_list_is_empty(ast_channel_framehooks(chan)) && ast_framehook_list_is_empty(ast_channel_framehooks(peer)))
694
695 /*
696  * The list of active channels
697  */
698 struct chanlist {
699         AST_LIST_ENTRY(chanlist) node;
700         struct ast_channel *chan;
701         /*! Channel interface dialing string (is tech/number).  (Stored in stuff[]) */
702         const char *interface;
703         /*! Channel technology name.  (Stored in stuff[]) */
704         const char *tech;
705         /*! Channel device addressing.  (Stored in stuff[]) */
706         const char *number;
707         /*! Original channel name.  Must be freed.  Could be NULL if allocation failed. */
708         char *orig_chan_name;
709         uint64_t flags;
710         /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
711         struct ast_party_connected_line connected;
712         /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
713         unsigned int pending_connected_update:1;
714         struct ast_aoc_decoded *aoc_s_rate_list;
715         /*! The interface, tech, and number strings are stuffed here. */
716         char stuff[0];
717 };
718
719 AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
720
721 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
722
723 static void chanlist_free(struct chanlist *outgoing)
724 {
725         ast_party_connected_line_free(&outgoing->connected);
726         ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
727         ast_free(outgoing->orig_chan_name);
728         ast_free(outgoing);
729 }
730
731 static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int answered_elsewhere)
732 {
733         /* Hang up a tree of stuff */
734         struct chanlist *outgoing;
735
736         while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
737                 /* Hangup any existing lines we have open */
738                 if (outgoing->chan && (outgoing->chan != exception)) {
739                         if (answered_elsewhere) {
740                                 /* This is for the channel drivers */
741                                 ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
742                         }
743                         ast_hangup(outgoing->chan);
744                 }
745                 chanlist_free(outgoing);
746         }
747 }
748
749 #define AST_MAX_WATCHERS 256
750
751 /*
752  * argument to handle_cause() and other functions.
753  */
754 struct cause_args {
755         struct ast_channel *chan;
756         int busy;
757         int congestion;
758         int nochan;
759 };
760
761 static void handle_cause(int cause, struct cause_args *num)
762 {
763         switch(cause) {
764         case AST_CAUSE_BUSY:
765                 num->busy++;
766                 break;
767         case AST_CAUSE_CONGESTION:
768                 num->congestion++;
769                 break;
770         case AST_CAUSE_NO_ROUTE_DESTINATION:
771         case AST_CAUSE_UNREGISTERED:
772                 num->nochan++;
773                 break;
774         case AST_CAUSE_NO_ANSWER:
775         case AST_CAUSE_NORMAL_CLEARING:
776                 break;
777         default:
778                 num->nochan++;
779                 break;
780         }
781 }
782
783 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
784 {
785         char rexten[2] = { exten, '\0' };
786
787         if (context) {
788                 if (!ast_goto_if_exists(chan, context, rexten, pri))
789                         return 1;
790         } else {
791                 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
792                         return 1;
793                 else if (!ast_strlen_zero(ast_channel_macrocontext(chan))) {
794                         if (!ast_goto_if_exists(chan, ast_channel_macrocontext(chan), rexten, pri))
795                                 return 1;
796                 }
797         }
798         return 0;
799 }
800
801 /* do not call with chan lock held */
802 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
803 {
804         const char *context;
805         const char *exten;
806
807         ast_channel_lock(chan);
808         context = ast_strdupa(S_OR(ast_channel_macrocontext(chan), ast_channel_context(chan)));
809         exten = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
810         ast_channel_unlock(chan);
811
812         return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
813 }
814
815 /*!
816  * helper function for wait_for_answer()
817  *
818  * \param o Outgoing call channel list.
819  * \param num Incoming call channel cause accumulation
820  * \param peerflags Dial option flags
821  * \param single TRUE if there is only one outgoing call.
822  * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
823  * \param to Remaining call timeout time.
824  * \param forced_clid OPT_FORCECLID caller id to send
825  * \param stored_clid Caller id representing the called party if needed
826  *
827  * XXX this code is highly suspicious, as it essentially overwrites
828  * the outgoing channel without properly deleting it.
829  *
830  * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
831  */
832 static void do_forward(struct chanlist *o, struct cause_args *num,
833         struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
834         struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
835 {
836         char tmpchan[256];
837         struct ast_channel *original = o->chan;
838         struct ast_channel *c = o->chan; /* the winner */
839         struct ast_channel *in = num->chan; /* the input channel */
840         char *stuff;
841         char *tech;
842         int cause;
843         struct ast_party_caller caller;
844
845         ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
846         if ((stuff = strchr(tmpchan, '/'))) {
847                 *stuff++ = '\0';
848                 tech = tmpchan;
849         } else {
850                 const char *forward_context;
851                 ast_channel_lock(c);
852                 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
853                 if (ast_strlen_zero(forward_context)) {
854                         forward_context = NULL;
855                 }
856                 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
857                 ast_channel_unlock(c);
858                 stuff = tmpchan;
859                 tech = "Local";
860         }
861         if (!strcasecmp(tech, "Local")) {
862                 /*
863                  * Drop the connected line update block for local channels since
864                  * this is going to run dialplan and the user can change his
865                  * mind about what connected line information he wants to send.
866                  */
867                 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
868         }
869
870         /* Before processing channel, go ahead and check for forwarding */
871         ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
872         /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
873         if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
874                 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
875                 ast_channel_publish_dial_forward(in, original, NULL, NULL, "CANCEL",
876                         ast_channel_call_forward(original));
877                 c = o->chan = NULL;
878                 cause = AST_CAUSE_BUSY;
879         } else {
880                 struct ast_format_cap *nativeformats;
881
882                 ast_channel_lock(in);
883                 nativeformats = ao2_bump(ast_channel_nativeformats(in));
884                 ast_channel_unlock(in);
885
886                 /* Setup parameters */
887                 c = o->chan = ast_request(tech, nativeformats, NULL, in, stuff, &cause);
888
889                 ao2_cleanup(nativeformats);
890
891                 if (c) {
892                         if (single && !caller_entertained) {
893                                 ast_channel_make_compatible(in, o->chan);
894                         }
895                         ast_channel_lock_both(in, o->chan);
896                         ast_channel_inherit_variables(in, o->chan);
897                         ast_channel_datastore_inherit(in, o->chan);
898                         ast_max_forwards_decrement(o->chan);
899                         ast_channel_unlock(in);
900                         ast_channel_unlock(o->chan);
901                         /* When a call is forwarded, we don't want to track new interfaces
902                          * dialed for CC purposes. Setting the done flag will ensure that
903                          * any Dial operations that happen later won't record CC interfaces.
904                          */
905                         ast_ignore_cc(o->chan);
906                         ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
907                 } else
908                         ast_log(LOG_NOTICE,
909                                 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
910                                 tech, stuff, cause);
911         }
912         if (!c) {
913                 ast_channel_publish_dial(in, original, stuff, "BUSY");
914                 ast_clear_flag64(o, DIAL_STILLGOING);
915                 handle_cause(cause, num);
916                 ast_hangup(original);
917         } else {
918                 ast_channel_lock_both(c, original);
919                 ast_party_redirecting_copy(ast_channel_redirecting(c),
920                         ast_channel_redirecting(original));
921                 ast_channel_unlock(c);
922                 ast_channel_unlock(original);
923
924                 ast_channel_lock_both(c, in);
925
926                 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
927                         ast_rtp_instance_early_bridge_make_compatible(c, in);
928                 }
929
930                 if (!ast_channel_redirecting(c)->from.number.valid
931                         || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
932                         /*
933                          * The call was not previously redirected so it is
934                          * now redirected from this number.
935                          */
936                         ast_party_number_free(&ast_channel_redirecting(c)->from.number);
937                         ast_party_number_init(&ast_channel_redirecting(c)->from.number);
938                         ast_channel_redirecting(c)->from.number.valid = 1;
939                         ast_channel_redirecting(c)->from.number.str =
940                                 ast_strdup(S_OR(ast_channel_macroexten(in), ast_channel_exten(in)));
941                 }
942
943                 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
944
945                 /* Determine CallerID to store in outgoing channel. */
946                 ast_party_caller_set_init(&caller, ast_channel_caller(c));
947                 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
948                         caller.id = *stored_clid;
949                         ast_channel_set_caller_event(c, &caller, NULL);
950                         ast_set_flag64(o, DIAL_CALLERID_ABSENT);
951                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
952                         ast_channel_caller(c)->id.number.str, NULL))) {
953                         /*
954                          * The new channel has no preset CallerID number by the channel
955                          * driver.  Use the dialplan extension and hint name.
956                          */
957                         caller.id = *stored_clid;
958                         ast_channel_set_caller_event(c, &caller, NULL);
959                         ast_set_flag64(o, DIAL_CALLERID_ABSENT);
960                 } else {
961                         ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
962                 }
963
964                 /* Determine CallerID for outgoing channel to send. */
965                 if (ast_test_flag64(o, OPT_FORCECLID)) {
966                         struct ast_party_connected_line connected;
967
968                         ast_party_connected_line_init(&connected);
969                         connected.id = *forced_clid;
970                         ast_party_connected_line_copy(ast_channel_connected(c), &connected);
971                 } else {
972                         ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
973                 }
974
975                 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
976
977                 ast_channel_appl_set(c, "AppDial");
978                 ast_channel_data_set(c, "(Outgoing Line)");
979                 ast_channel_publish_snapshot(c);
980
981                 ast_channel_unlock(in);
982                 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
983                         struct ast_party_redirecting redirecting;
984
985                         /*
986                          * Redirecting updates to the caller make sense only on single
987                          * calls.
988                          *
989                          * We must unlock c before calling
990                          * ast_channel_redirecting_macro, because we put c into
991                          * autoservice there.  That is pretty much a guaranteed
992                          * deadlock.  This is why the handling of c's lock may seem a
993                          * bit unusual here.
994                          */
995                         ast_party_redirecting_init(&redirecting);
996                         ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
997                         ast_channel_unlock(c);
998                         if (ast_channel_redirecting_sub(c, in, &redirecting, 0) &&
999                                 ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
1000                                 ast_channel_update_redirecting(in, &redirecting, NULL);
1001                         }
1002                         ast_party_redirecting_free(&redirecting);
1003                 } else {
1004                         ast_channel_unlock(c);
1005                 }
1006
1007                 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1008                         *to = -1;
1009                 }
1010
1011                 if (ast_call(c, stuff, 0)) {
1012                         ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1013                                 tech, stuff);
1014                         ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1015                         ast_clear_flag64(o, DIAL_STILLGOING);
1016                         ast_hangup(original);
1017                         ast_hangup(c);
1018                         c = o->chan = NULL;
1019                         num->nochan++;
1020                 } else {
1021                         ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1022                                 ast_channel_call_forward(original));
1023
1024                         ast_channel_publish_dial(in, c, stuff, NULL);
1025
1026                         /* Hangup the original channel now, in case we needed it */
1027                         ast_hangup(original);
1028                 }
1029                 if (single && !caller_entertained) {
1030                         ast_indicate(in, -1);
1031                 }
1032         }
1033 }
1034
1035 /* argument used for some functions. */
1036 struct privacy_args {
1037         int sentringing;
1038         int privdb_val;
1039         char privcid[256];
1040         char privintro[1024];
1041         char status[256];
1042 };
1043
1044 static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
1045 {
1046         struct chanlist *outgoing;
1047         AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1048                 if (!outgoing->chan || outgoing->chan == exception) {
1049                         continue;
1050                 }
1051                 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1052         }
1053 }
1054
1055 /*!
1056  * \internal
1057  * \brief Update connected line on chan from peer.
1058  * \since 13.6.0
1059  *
1060  * \param chan Channel to get connected line updated.
1061  * \param peer Channel providing connected line information.
1062  * \param is_caller Non-zero if chan is the calling channel.
1063  *
1064  * \return Nothing
1065  */
1066 static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
1067 {
1068         struct ast_party_connected_line connected_caller;
1069
1070         ast_party_connected_line_init(&connected_caller);
1071
1072         ast_channel_lock(peer);
1073         ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1074         ast_channel_unlock(peer);
1075         connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1076         if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)
1077                 && ast_channel_connected_line_macro(peer, chan, &connected_caller, is_caller, 0)) {
1078                 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1079         }
1080         ast_party_connected_line_free(&connected_caller);
1081 }
1082
1083 static struct ast_channel *wait_for_answer(struct ast_channel *in,
1084         struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
1085         char *opt_args[],
1086         struct privacy_args *pa,
1087         const struct cause_args *num_in, int *result, char *dtmf_progress,
1088         const int ignore_cc,
1089         struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
1090 {
1091         struct cause_args num = *num_in;
1092         int prestart = num.busy + num.congestion + num.nochan;
1093         int orig = *to;
1094         struct ast_channel *peer = NULL;
1095 #ifdef HAVE_EPOLL
1096         struct chanlist *epollo;
1097 #endif
1098         struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1099         /* single is set if only one destination is enabled */
1100         int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1101         int caller_entertained = outgoing
1102                 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1103         struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1104         int cc_recall_core_id;
1105         int is_cc_recall;
1106         int cc_frame_received = 0;
1107         int num_ringing = 0;
1108         struct timeval start = ast_tvnow();
1109
1110         if (single) {
1111                 /* Turn off hold music, etc */
1112                 if (!caller_entertained) {
1113                         ast_deactivate_generator(in);
1114                         /* If we are calling a single channel, and not providing ringback or music, */
1115                         /* then, make them compatible for in-band tone purpose */
1116                         if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1117                                 /* If these channels can not be made compatible,
1118                                  * there is no point in continuing.  The bridge
1119                                  * will just fail if it gets that far.
1120                                  */
1121                                 *to = -1;
1122                                 strcpy(pa->status, "CONGESTION");
1123                                 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1124                                 return NULL;
1125                         }
1126                 }
1127
1128                 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1129                         && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1130                         update_connected_line_from_peer(in, outgoing->chan, 1);
1131                 }
1132         }
1133
1134         is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1135
1136 #ifdef HAVE_EPOLL
1137         AST_LIST_TRAVERSE(out_chans, epollo, node) {
1138                 ast_poll_channel_add(in, epollo->chan);
1139         }
1140 #endif
1141
1142         while ((*to = ast_remaining_ms(start, orig)) && !peer) {
1143                 struct chanlist *o;
1144                 int pos = 0; /* how many channels do we handle */
1145                 int numlines = prestart;
1146                 struct ast_channel *winner;
1147                 struct ast_channel *watchers[AST_MAX_WATCHERS];
1148
1149                 watchers[pos++] = in;
1150                 AST_LIST_TRAVERSE(out_chans, o, node) {
1151                         /* Keep track of important channels */
1152                         if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1153                                 watchers[pos++] = o->chan;
1154                         numlines++;
1155                 }
1156                 if (pos == 1) { /* only the input channel is available */
1157                         if (numlines == (num.busy + num.congestion + num.nochan)) {
1158                                 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1159                                 if (num.busy)
1160                                         strcpy(pa->status, "BUSY");
1161                                 else if (num.congestion)
1162                                         strcpy(pa->status, "CONGESTION");
1163                                 else if (num.nochan)
1164                                         strcpy(pa->status, "CHANUNAVAIL");
1165                         } else {
1166                                 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1167                         }
1168                         *to = 0;
1169                         if (is_cc_recall) {
1170                                 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1171                         }
1172                         return NULL;
1173                 }
1174                 winner = ast_waitfor_n(watchers, pos, to);
1175                 AST_LIST_TRAVERSE(out_chans, o, node) {
1176                         struct ast_frame *f;
1177                         struct ast_channel *c = o->chan;
1178
1179                         if (c == NULL)
1180                                 continue;
1181                         if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1182                                 if (!peer) {
1183                                         ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1184                                         if (o->orig_chan_name
1185                                                 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1186                                                 /*
1187                                                  * The channel name changed so we must generate COLP update.
1188                                                  * Likely because a call pickup channel masqueraded in.
1189                                                  */
1190                                                 update_connected_line_from_peer(in, c, 1);
1191                                         } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1192                                                 if (o->pending_connected_update) {
1193                                                         if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1194                                                                 ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1195                                                                 ast_channel_update_connected_line(in, &o->connected, NULL);
1196                                                         }
1197                                                 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1198                                                         update_connected_line_from_peer(in, c, 1);
1199                                                 }
1200                                         }
1201                                         if (o->aoc_s_rate_list) {
1202                                                 size_t encoded_size;
1203                                                 struct ast_aoc_encoded *encoded;
1204                                                 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1205                                                         ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1206                                                         ast_aoc_destroy_encoded(encoded);
1207                                                 }
1208                                         }
1209                                         peer = c;
1210                                         ast_copy_flags64(peerflags, o,
1211                                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1212                                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1213                                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1214                                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1215                                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1216                                                 DIAL_NOFORWARDHTML);
1217                                         ast_channel_dialcontext_set(c, "");
1218                                         ast_channel_exten_set(c, "");
1219                                 }
1220                                 continue;
1221                         }
1222                         if (c != winner)
1223                                 continue;
1224                         /* here, o->chan == c == winner */
1225                         if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1226                                 pa->sentringing = 0;
1227                                 if (!ignore_cc && (f = ast_read(c))) {
1228                                         if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1229                                                 /* This channel is forwarding the call, and is capable of CC, so
1230                                                  * be sure to add the new device interface to the list
1231                                                  */
1232                                                 ast_handle_cc_control_frame(in, c, f->data.ptr);
1233                                         }
1234                                         ast_frfree(f);
1235                                 }
1236
1237                                 if (o->pending_connected_update) {
1238                                         /*
1239                                          * Re-seed the chanlist's connected line information with
1240                                          * previously acquired connected line info from the incoming
1241                                          * channel.  The previously acquired connected line info could
1242                                          * have been set through the CONNECTED_LINE dialplan function.
1243                                          */
1244                                         o->pending_connected_update = 0;
1245                                         ast_channel_lock(in);
1246                                         ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1247                                         ast_channel_unlock(in);
1248                                 }
1249
1250                                 do_forward(o, &num, peerflags, single, caller_entertained, &orig,
1251                                         forced_clid, stored_clid);
1252
1253                                 if (o->chan) {
1254                                         ast_free(o->orig_chan_name);
1255                                         o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1256                                         if (single
1257                                                 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1258                                                 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1259                                                 update_connected_line_from_peer(in, o->chan, 1);
1260                                         }
1261                                 }
1262                                 continue;
1263                         }
1264                         f = ast_read(winner);
1265                         if (!f) {
1266                                 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1267 #ifdef HAVE_EPOLL
1268                                 ast_poll_channel_del(in, c);
1269 #endif
1270                                 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1271                                 ast_hangup(c);
1272                                 c = o->chan = NULL;
1273                                 ast_clear_flag64(o, DIAL_STILLGOING);
1274                                 handle_cause(ast_channel_hangupcause(in), &num);
1275                                 continue;
1276                         }
1277                         switch (f->frametype) {
1278                         case AST_FRAME_CONTROL:
1279                                 switch (f->subclass.integer) {
1280                                 case AST_CONTROL_ANSWER:
1281                                         /* This is our guy if someone answered. */
1282                                         if (!peer) {
1283                                                 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1284                                                 if (o->orig_chan_name
1285                                                         && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1286                                                         /*
1287                                                          * The channel name changed so we must generate COLP update.
1288                                                          * Likely because a call pickup channel masqueraded in.
1289                                                          */
1290                                                         update_connected_line_from_peer(in, c, 1);
1291                                                 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1292                                                         if (o->pending_connected_update) {
1293                                                                 if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1294                                                                         ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1295                                                                         ast_channel_update_connected_line(in, &o->connected, NULL);
1296                                                                 }
1297                                                         } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1298                                                                 update_connected_line_from_peer(in, c, 1);
1299                                                         }
1300                                                 }
1301                                                 if (o->aoc_s_rate_list) {
1302                                                         size_t encoded_size;
1303                                                         struct ast_aoc_encoded *encoded;
1304                                                         if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1305                                                                 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1306                                                                 ast_aoc_destroy_encoded(encoded);
1307                                                         }
1308                                                 }
1309                                                 peer = c;
1310                                                 /* Inform everyone else that they've been canceled.
1311                                                  * The dial end event for the peer will be sent out after
1312                                                  * other Dial options have been handled.
1313                                                  */
1314                                                 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1315                                                 ast_copy_flags64(peerflags, o,
1316                                                         OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1317                                                         OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1318                                                         OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1319                                                         OPT_CALLEE_PARK | OPT_CALLER_PARK |
1320                                                         OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1321                                                         DIAL_NOFORWARDHTML);
1322                                                 ast_channel_dialcontext_set(c, "");
1323                                                 ast_channel_exten_set(c, "");
1324                                                 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1325                                                         /* Setup early bridge if appropriate */
1326                                                         ast_channel_early_bridge(in, peer);
1327                                                 }
1328                                         }
1329                                         /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1330                                         ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1331                                         ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1332                                         break;
1333                                 case AST_CONTROL_BUSY:
1334                                         ast_verb(3, "%s is busy\n", ast_channel_name(c));
1335                                         ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1336                                         ast_channel_publish_dial(in, c, NULL, "BUSY");
1337                                         ast_hangup(c);
1338                                         c = o->chan = NULL;
1339                                         ast_clear_flag64(o, DIAL_STILLGOING);
1340                                         handle_cause(AST_CAUSE_BUSY, &num);
1341                                         break;
1342                                 case AST_CONTROL_CONGESTION:
1343                                         ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1344                                         ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1345                                         ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1346                                         ast_hangup(c);
1347                                         c = o->chan = NULL;
1348                                         ast_clear_flag64(o, DIAL_STILLGOING);
1349                                         handle_cause(AST_CAUSE_CONGESTION, &num);
1350                                         break;
1351                                 case AST_CONTROL_RINGING:
1352                                         /* This is a tricky area to get right when using a native
1353                                          * CC agent. The reason is that we do the best we can to send only a
1354                                          * single ringing notification to the caller.
1355                                          *
1356                                          * Call completion complicates the logic used here. CCNR is typically
1357                                          * offered during a ringing message. Let's say that party A calls
1358                                          * parties B, C, and D. B and C do not support CC requests, but D
1359                                          * does. If we were to receive a ringing notification from B before
1360                                          * the others, then we would end up sending a ringing message to
1361                                          * A with no CCNR offer present.
1362                                          *
1363                                          * The approach that we have taken is that if we receive a ringing
1364                                          * response from a party and no CCNR offer is present, we need to
1365                                          * wait. Specifically, we need to wait until either a) a called party
1366                                          * offers CCNR in its ringing response or b) all called parties have
1367                                          * responded in some way to our call and none offers CCNR.
1368                                          *
1369                                          * The drawback to this is that if one of the parties has a delayed
1370                                          * response or, god forbid, one just plain doesn't respond to our
1371                                          * outgoing call, then this will result in a significant delay between
1372                                          * when the caller places the call and hears ringback.
1373                                          *
1374                                          * Note also that if CC is disabled for this call, then it is perfectly
1375                                          * fine for ringing frames to get sent through.
1376                                          */
1377                                         ++num_ringing;
1378                                         if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1379                                                 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1380                                                 /* Setup early media if appropriate */
1381                                                 if (single && !caller_entertained
1382                                                         && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1383                                                         ast_channel_early_bridge(in, c);
1384                                                 }
1385                                                 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1386                                                         ast_indicate(in, AST_CONTROL_RINGING);
1387                                                         pa->sentringing++;
1388                                                 }
1389                                         }
1390                                         ast_channel_publish_dial(in, c, NULL, "RINGING");
1391                                         break;
1392                                 case AST_CONTROL_PROGRESS:
1393                                         ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1394                                         /* Setup early media if appropriate */
1395                                         if (single && !caller_entertained
1396                                                 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1397                                                 ast_channel_early_bridge(in, c);
1398                                         }
1399                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1400                                                 if (single || (!single && !pa->sentringing)) {
1401                                                         ast_indicate(in, AST_CONTROL_PROGRESS);
1402                                                 }
1403                                         }
1404                                         if (!ast_strlen_zero(dtmf_progress)) {
1405                                                 ast_verb(3,
1406                                                         "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
1407                                                         dtmf_progress);
1408                                                 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1409                                         }
1410                                         ast_channel_publish_dial(in, c, NULL, "PROGRESS");
1411                                         break;
1412                                 case AST_CONTROL_VIDUPDATE:
1413                                 case AST_CONTROL_SRCUPDATE:
1414                                 case AST_CONTROL_SRCCHANGE:
1415                                         if (!single || caller_entertained) {
1416                                                 break;
1417                                         }
1418                                         ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1419                                                 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1420                                         ast_indicate(in, f->subclass.integer);
1421                                         break;
1422                                 case AST_CONTROL_CONNECTED_LINE:
1423                                         if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1424                                                 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1425                                                 break;
1426                                         }
1427                                         if (!single) {
1428                                                 struct ast_party_connected_line connected;
1429
1430                                                 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1431                                                         ast_channel_name(c), ast_channel_name(in));
1432                                                 ast_party_connected_line_set_init(&connected, &o->connected);
1433                                                 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1434                                                 ast_party_connected_line_set(&o->connected, &connected, NULL);
1435                                                 ast_party_connected_line_free(&connected);
1436                                                 o->pending_connected_update = 1;
1437                                                 break;
1438                                         }
1439                                         if (ast_channel_connected_line_sub(c, in, f, 1) &&
1440                                                 ast_channel_connected_line_macro(c, in, f, 1, 1)) {
1441                                                 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1442                                         }
1443                                         break;
1444                                 case AST_CONTROL_AOC:
1445                                         {
1446                                                 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1447                                                 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1448                                                         ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1449                                                         o->aoc_s_rate_list = decoded;
1450                                                 } else {
1451                                                         ast_aoc_destroy_decoded(decoded);
1452                                                 }
1453                                         }
1454                                         break;
1455                                 case AST_CONTROL_REDIRECTING:
1456                                         if (!single) {
1457                                                 /*
1458                                                  * Redirecting updates to the caller make sense only on single
1459                                                  * calls.
1460                                                  */
1461                                                 break;
1462                                         }
1463                                         if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1464                                                 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1465                                                 break;
1466                                         }
1467                                         ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1468                                                 ast_channel_name(c), ast_channel_name(in));
1469                                         if (ast_channel_redirecting_sub(c, in, f, 1) &&
1470                                                 ast_channel_redirecting_macro(c, in, f, 1, 1)) {
1471                                                 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1472                                         }
1473                                         pa->sentringing = 0;
1474                                         break;
1475                                 case AST_CONTROL_PROCEEDING:
1476                                         ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1477                                         if (single && !caller_entertained
1478                                                 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1479                                                 ast_channel_early_bridge(in, c);
1480                                         }
1481                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1482                                                 ast_indicate(in, AST_CONTROL_PROCEEDING);
1483                                         ast_channel_publish_dial(in, c, NULL, "PROCEEDING");
1484                                         break;
1485                                 case AST_CONTROL_HOLD:
1486                                         /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1487                                         ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1488                                         ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1489                                         break;
1490                                 case AST_CONTROL_UNHOLD:
1491                                         /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1492                                         ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1493                                         ast_indicate(in, AST_CONTROL_UNHOLD);
1494                                         break;
1495                                 case AST_CONTROL_OFFHOOK:
1496                                 case AST_CONTROL_FLASH:
1497                                         /* Ignore going off hook and flash */
1498                                         break;
1499                                 case AST_CONTROL_CC:
1500                                         if (!ignore_cc) {
1501                                                 ast_handle_cc_control_frame(in, c, f->data.ptr);
1502                                                 cc_frame_received = 1;
1503                                         }
1504                                         break;
1505                                 case AST_CONTROL_PVT_CAUSE_CODE:
1506                                         ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1507                                         break;
1508                                 case -1:
1509                                         if (single && !caller_entertained) {
1510                                                 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1511                                                 ast_indicate(in, -1);
1512                                                 pa->sentringing = 0;
1513                                         }
1514                                         break;
1515                                 default:
1516                                         ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1517                                         break;
1518                                 }
1519                                 break;
1520                         case AST_FRAME_VOICE:
1521                         case AST_FRAME_IMAGE:
1522                                 if (caller_entertained) {
1523                                         break;
1524                                 }
1525                                 /* Fall through */
1526                         case AST_FRAME_TEXT:
1527                                 if (single && ast_write(in, f)) {
1528                                         ast_log(LOG_WARNING, "Unable to write frametype: %u\n",
1529                                                 f->frametype);
1530                                 }
1531                                 break;
1532                         case AST_FRAME_HTML:
1533                                 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1534                                         && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1535                                         ast_log(LOG_WARNING, "Unable to send URL\n");
1536                                 }
1537                                 break;
1538                         default:
1539                                 break;
1540                         }
1541                         ast_frfree(f);
1542                 } /* end for */
1543                 if (winner == in) {
1544                         struct ast_frame *f = ast_read(in);
1545 #if 0
1546                         if (f && (f->frametype != AST_FRAME_VOICE))
1547                                 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1548                         else if (!f || (f->frametype != AST_FRAME_VOICE))
1549                                 printf("Hangup received on %s\n", in->name);
1550 #endif
1551                         if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1552                                 /* Got hung up */
1553                                 *to = -1;
1554                                 strcpy(pa->status, "CANCEL");
1555                                 publish_dial_end_event(in, out_chans, NULL, pa->status);
1556                                 if (f) {
1557                                         if (f->data.uint32) {
1558                                                 ast_channel_hangupcause_set(in, f->data.uint32);
1559                                         }
1560                                         ast_frfree(f);
1561                                 }
1562                                 if (is_cc_recall) {
1563                                         ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1564                                 }
1565                                 return NULL;
1566                         }
1567
1568                         /* now f is guaranteed non-NULL */
1569                         if (f->frametype == AST_FRAME_DTMF) {
1570                                 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1571                                         const char *context;
1572                                         ast_channel_lock(in);
1573                                         context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1574                                         if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1575                                                 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1576                                                 *to = 0;
1577                                                 *result = f->subclass.integer;
1578                                                 strcpy(pa->status, "CANCEL");
1579                                                 publish_dial_end_event(in, out_chans, NULL, pa->status);
1580                                                 ast_frfree(f);
1581                                                 ast_channel_unlock(in);
1582                                                 if (is_cc_recall) {
1583                                                         ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1584                                                 }
1585                                                 return NULL;
1586                                         }
1587                                         ast_channel_unlock(in);
1588                                 }
1589
1590                                 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1591                                         detect_disconnect(in, f->subclass.integer, &featurecode)) {
1592                                         ast_verb(3, "User requested call disconnect.\n");
1593                                         *to = 0;
1594                                         strcpy(pa->status, "CANCEL");
1595                                         publish_dial_end_event(in, out_chans, NULL, pa->status);
1596                                         ast_frfree(f);
1597                                         if (is_cc_recall) {
1598                                                 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1599                                         }
1600                                         return NULL;
1601                                 }
1602                         }
1603
1604                         /* Send the frame from the in channel to all outgoing channels. */
1605                         AST_LIST_TRAVERSE(out_chans, o, node) {
1606                                 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1607                                         /* This outgoing channel has died so don't send the frame to it. */
1608                                         continue;
1609                                 }
1610                                 switch (f->frametype) {
1611                                 case AST_FRAME_HTML:
1612                                         /* Forward HTML stuff */
1613                                         if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1614                                                 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1615                                                 ast_log(LOG_WARNING, "Unable to send URL\n");
1616                                         }
1617                                         break;
1618                                 case AST_FRAME_VOICE:
1619                                 case AST_FRAME_IMAGE:
1620                                         if (!single || caller_entertained) {
1621                                                 /*
1622                                                  * We are calling multiple parties or caller is being
1623                                                  * entertained and has thus not been made compatible.
1624                                                  * No need to check any other called parties.
1625                                                  */
1626                                                 goto skip_frame;
1627                                         }
1628                                         /* Fall through */
1629                                 case AST_FRAME_TEXT:
1630                                 case AST_FRAME_DTMF_BEGIN:
1631                                 case AST_FRAME_DTMF_END:
1632                                         if (ast_write(o->chan, f)) {
1633                                                 ast_log(LOG_WARNING, "Unable to forward frametype: %u\n",
1634                                                         f->frametype);
1635                                         }
1636                                         break;
1637                                 case AST_FRAME_CONTROL:
1638                                         switch (f->subclass.integer) {
1639                                         case AST_CONTROL_HOLD:
1640                                                 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1641                                                 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1642                                                 break;
1643                                         case AST_CONTROL_UNHOLD:
1644                                                 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1645                                                 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1646                                                 break;
1647                                         case AST_CONTROL_VIDUPDATE:
1648                                         case AST_CONTROL_SRCUPDATE:
1649                                         case AST_CONTROL_SRCCHANGE:
1650                                                 if (!single || caller_entertained) {
1651                                                         /*
1652                                                          * We are calling multiple parties or caller is being
1653                                                          * entertained and has thus not been made compatible.
1654                                                          * No need to check any other called parties.
1655                                                          */
1656                                                         goto skip_frame;
1657                                                 }
1658                                                 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1659                                                         ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1660                                                 ast_indicate(o->chan, f->subclass.integer);
1661                                                 break;
1662                                         case AST_CONTROL_CONNECTED_LINE:
1663                                                 if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
1664                                                         ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
1665                                                         ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1666                                                 }
1667                                                 break;
1668                                         case AST_CONTROL_REDIRECTING:
1669                                                 if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
1670                                                         ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
1671                                                         ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1672                                                 }
1673                                                 break;
1674                                         default:
1675                                                 /* We are not going to do anything with this frame. */
1676                                                 goto skip_frame;
1677                                         }
1678                                         break;
1679                                 default:
1680                                         /* We are not going to do anything with this frame. */
1681                                         goto skip_frame;
1682                                 }
1683                         }
1684 skip_frame:;
1685                         ast_frfree(f);
1686                 }
1687         }
1688
1689         if (!*to || ast_check_hangup(in)) {
1690                 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1691                 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1692         }
1693
1694 #ifdef HAVE_EPOLL
1695         AST_LIST_TRAVERSE(out_chans, epollo, node) {
1696                 if (epollo->chan)
1697                         ast_poll_channel_del(in, epollo->chan);
1698         }
1699 #endif
1700
1701         if (is_cc_recall) {
1702                 ast_cc_completed(in, "Recall completed!");
1703         }
1704         return peer;
1705 }
1706
1707 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1708 {
1709         char disconnect_code[AST_FEATURE_MAX_LEN];
1710         int res;
1711
1712         ast_str_append(featurecode, 1, "%c", code);
1713
1714         res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1715         if (res) {
1716                 ast_str_reset(*featurecode);
1717                 return 0;
1718         }
1719
1720         if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1721                 /* Could be a partial match, anyway */
1722                 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1723                         ast_str_reset(*featurecode);
1724                 }
1725                 return 0;
1726         }
1727
1728         if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
1729                 ast_str_reset(*featurecode);
1730                 return 0;
1731         }
1732
1733         return 1;
1734 }
1735
1736 /* returns true if there is a valid privacy reply */
1737 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1738 {
1739         if (res < '1')
1740                 return 0;
1741         if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1742                 return 1;
1743         if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1744                 return 1;
1745         return 0;
1746 }
1747
1748 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1749         struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1750 {
1751
1752         int res2;
1753         int loopcount = 0;
1754
1755         /* Get the user's intro, store it in priv-callerintros/$CID,
1756            unless it is already there-- this should be done before the
1757            call is actually dialed  */
1758
1759         /* all ring indications and moh for the caller has been halted as soon as the
1760            target extension was picked up. We are going to have to kill some
1761            time and make the caller believe the peer hasn't picked up yet */
1762
1763         if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1764                 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
1765                 ast_indicate(chan, -1);
1766                 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
1767                 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1768                 ast_channel_musicclass_set(chan, original_moh);
1769         } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
1770                 ast_indicate(chan, AST_CONTROL_RINGING);
1771                 pa->sentringing++;
1772         }
1773
1774         /* Start autoservice on the other chan ?? */
1775         res2 = ast_autoservice_start(chan);
1776         /* Now Stream the File */
1777         for (loopcount = 0; loopcount < 3; loopcount++) {
1778                 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1779                         break;
1780                 if (!res2) /* on timeout, play the message again */
1781                         res2 = ast_play_and_wait(peer, "priv-callpending");
1782                 if (!valid_priv_reply(opts, res2))
1783                         res2 = 0;
1784                 /* priv-callpending script:
1785                    "I have a caller waiting, who introduces themselves as:"
1786                 */
1787                 if (!res2)
1788                         res2 = ast_play_and_wait(peer, pa->privintro);
1789                 if (!valid_priv_reply(opts, res2))
1790                         res2 = 0;
1791                 /* now get input from the called party, as to their choice */
1792                 if (!res2) {
1793                         /* XXX can we have both, or they are mutually exclusive ? */
1794                         if (ast_test_flag64(opts, OPT_PRIVACY))
1795                                 res2 = ast_play_and_wait(peer, "priv-callee-options");
1796                         if (ast_test_flag64(opts, OPT_SCREENING))
1797                                 res2 = ast_play_and_wait(peer, "screen-callee-options");
1798                 }
1799
1800                 /*! \page DialPrivacy Dial Privacy scripts
1801                  * \par priv-callee-options script:
1802                  * \li Dial 1 if you wish this caller to reach you directly in the future,
1803                  *      and immediately connect to their incoming call.
1804                  * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
1805                  * \li Dial 3 to send this caller to the torture menus, now and forevermore.
1806                  * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1807                  * \li Dial 5 to allow this caller to come straight thru to you in the future,
1808                  *      but right now, just this once, send them to voicemail.
1809                  *
1810                  * \par screen-callee-options script:
1811                  * \li Dial 1 if you wish to immediately connect to the incoming call
1812                  * \li Dial 2 if you wish to send this caller to voicemail.
1813                  * \li Dial 3 to send this caller to the torture menus.
1814                  * \li Dial 4 to send this caller to a simple "go away" menu.
1815                  */
1816                 if (valid_priv_reply(opts, res2))
1817                         break;
1818                 /* invalid option */
1819                 res2 = ast_play_and_wait(peer, "vm-sorry");
1820         }
1821
1822         if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1823                 ast_moh_stop(chan);
1824         } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
1825                 ast_indicate(chan, -1);
1826                 pa->sentringing = 0;
1827         }
1828         ast_autoservice_stop(chan);
1829         if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1830                 /* map keypresses to various things, the index is res2 - '1' */
1831                 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1832                 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1833                 int i = res2 - '1';
1834                 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1835                         opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1836                 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1837         }
1838         switch (res2) {
1839         case '1':
1840                 break;
1841         case '2':
1842                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1843                 break;
1844         case '3':
1845                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1846                 break;
1847         case '4':
1848                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1849                 break;
1850         case '5':
1851                 /* XXX should we set status to DENY ? */
1852                 if (ast_test_flag64(opts, OPT_PRIVACY))
1853                         break;
1854                 /* if not privacy, then 5 is the same as "default" case */
1855         default: /* bad input or -1 if failure to start autoservice */
1856                 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do?  */
1857                 /* well, there seems basically two choices. Just patch the caller thru immediately,
1858                           or,... put 'em thru to voicemail. */
1859                 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1860                 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1861                 /* XXX should we set status to DENY ? */
1862                 /* XXX what about the privacy flags ? */
1863                 break;
1864         }
1865
1866         if (res2 == '1') { /* the only case where we actually connect */
1867                 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1868                    just clog things up, and it's not useful information, not being tied to a CID */
1869                 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1870                         ast_filedelete(pa->privintro, NULL);
1871                         if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1872                                 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1873                         else
1874                                 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1875                 }
1876                 return 0; /* the good exit path */
1877         } else {
1878                 /* hang up on the callee -- he didn't want to talk anyway! */
1879                 ast_autoservice_chan_hangup_peer(chan, peer);
1880                 return -1;
1881         }
1882 }
1883
1884 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1885 static int setup_privacy_args(struct privacy_args *pa,
1886         struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1887 {
1888         char callerid[60];
1889         int res;
1890         char *l;
1891
1892         if (ast_channel_caller(chan)->id.number.valid
1893                 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
1894                 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
1895                 ast_shrink_phone_number(l);
1896                 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1897                         ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1898                         pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1899                 } else {
1900                         ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1901                         pa->privdb_val = AST_PRIVACY_UNKNOWN;
1902                 }
1903         } else {
1904                 char *tnam, *tn2;
1905
1906                 tnam = ast_strdupa(ast_channel_name(chan));
1907                 /* clean the channel name so slashes don't try to end up in disk file name */
1908                 for (tn2 = tnam; *tn2; tn2++) {
1909                         if (*tn2 == '/')  /* any other chars to be afraid of? */
1910                                 *tn2 = '=';
1911                 }
1912                 ast_verb(3, "Privacy-- callerid is empty\n");
1913
1914                 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
1915                 l = callerid;
1916                 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1917         }
1918
1919         ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1920
1921         if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1922                 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1923                 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1924                 pa->privdb_val = AST_PRIVACY_ALLOW;
1925         } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1926                 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1927         }
1928
1929         if (pa->privdb_val == AST_PRIVACY_DENY) {
1930                 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1931                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1932                 return 0;
1933         } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1934                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1935                 return 0; /* Is this right? */
1936         } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1937                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1938                 return 0; /* is this right??? */
1939         } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1940                 /* Get the user's intro, store it in priv-callerintros/$CID,
1941                    unless it is already there-- this should be done before the
1942                    call is actually dialed  */
1943
1944                 /* make sure the priv-callerintros dir actually exists */
1945                 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1946                 if ((res = ast_mkdir(pa->privintro, 0755))) {
1947                         ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1948                         return -1;
1949                 }
1950
1951                 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1952                 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1953                         /* the DELUX version of this code would allow this caller the
1954                            option to hear and retape their previously recorded intro.
1955                         */
1956                 } else {
1957                         int duration; /* for feedback from play_and_wait */
1958                         /* the file doesn't exist yet. Let the caller submit his
1959                            vocal intro for posterity */
1960                         /* priv-recordintro script:
1961
1962                            "At the tone, please say your name:"
1963
1964                         */
1965                         int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1966                         ast_answer(chan);
1967                         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 */
1968                                                                         /* don't think we'll need a lock removed, we took care of
1969                                                                            conflicts by naming the pa.privintro file */
1970                         if (res == -1) {
1971                                 /* Delete the file regardless since they hung up during recording */
1972                                 ast_filedelete(pa->privintro, NULL);
1973                                 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1974                                         ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1975                                 else
1976                                         ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1977                                 return -1;
1978                         }
1979                         if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
1980                                 ast_waitstream(chan, "");
1981                 }
1982         }
1983         return 1; /* success */
1984 }
1985
1986 static void end_bridge_callback(void *data)
1987 {
1988         char buf[80];
1989         time_t end;
1990         struct ast_channel *chan = data;
1991
1992         time(&end);
1993
1994         ast_channel_lock(chan);
1995         ast_channel_stage_snapshot(chan);
1996         snprintf(buf, sizeof(buf), "%d", ast_channel_get_up_time(chan));
1997         pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1998         snprintf(buf, sizeof(buf), "%d", ast_channel_get_duration(chan));
1999         pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
2000         ast_channel_stage_snapshot_done(chan);
2001         ast_channel_unlock(chan);
2002 }
2003
2004 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2005         bconfig->end_bridge_callback_data = originator;
2006 }
2007
2008 static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2009 {
2010         struct ast_tone_zone_sound *ts = NULL;
2011         int res;
2012         const char *str = data;
2013
2014         if (ast_strlen_zero(str)) {
2015                 ast_debug(1,"Nothing to play\n");
2016                 return -1;
2017         }
2018
2019         ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2020
2021         if (ts && ts->data[0]) {
2022                 res = ast_playtones_start(chan, 0, ts->data, 0);
2023         } else {
2024                 res = -1;
2025         }
2026
2027         if (ts) {
2028                 ts = ast_tone_zone_sound_unref(ts);
2029         }
2030
2031         if (res) {
2032                 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2033         }
2034
2035         return res;
2036 }
2037
2038 /*!
2039  * \internal
2040  * \brief Setup the after bridge goto location on the peer.
2041  * \since 12.0.0
2042  *
2043  * \param chan Calling channel for bridge.
2044  * \param peer Peer channel for bridge.
2045  * \param opts Dialing option flags.
2046  * \param opt_args Dialing option argument strings.
2047  *
2048  * \return Nothing
2049  */
2050 static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
2051 {
2052         const char *context;
2053         const char *extension;
2054         int priority;
2055
2056         if (ast_test_flag64(opts, OPT_PEER_H)) {
2057                 ast_channel_lock(chan);
2058                 context = ast_strdupa(ast_channel_context(chan));
2059                 ast_channel_unlock(chan);
2060                 ast_bridge_set_after_h(peer, context);
2061         } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2062                 ast_channel_lock(chan);
2063                 context = ast_strdupa(ast_channel_context(chan));
2064                 extension = ast_strdupa(ast_channel_exten(chan));
2065                 priority = ast_channel_priority(chan);
2066                 ast_channel_unlock(chan);
2067                 ast_bridge_set_after_go_on(peer, context, extension, priority,
2068                         opt_args[OPT_ARG_CALLEE_GO_ON]);
2069         }
2070 }
2071
2072 static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2073 {
2074         int res = -1; /* default: error */
2075         char *rest, *cur; /* scan the list of destinations */
2076         struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2077         struct chanlist *outgoing;
2078         struct chanlist *tmp;
2079         struct ast_channel *peer;
2080         int to; /* timeout */
2081         struct cause_args num = { chan, 0, 0, 0 };
2082         int cause;
2083
2084         struct ast_bridge_config config = { { 0, } };
2085         struct timeval calldurationlimit = { 0, };
2086         char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
2087         struct privacy_args pa = {
2088                 .sentringing = 0,
2089                 .privdb_val = 0,
2090                 .status = "INVALIDARGS",
2091         };
2092         int sentringing = 0, moh = 0;
2093         const char *outbound_group = NULL;
2094         int result = 0;
2095         char *parse;
2096         int opermode = 0;
2097         int delprivintro = 0;
2098         AST_DECLARE_APP_ARGS(args,
2099                 AST_APP_ARG(peers);
2100                 AST_APP_ARG(timeout);
2101                 AST_APP_ARG(options);
2102                 AST_APP_ARG(url);
2103         );
2104         struct ast_flags64 opts = { 0, };
2105         char *opt_args[OPT_ARG_ARRAY_SIZE];
2106         int fulldial = 0, num_dialed = 0;
2107         int ignore_cc = 0;
2108         char device_name[AST_CHANNEL_NAME];
2109         char forced_clid_name[AST_MAX_EXTENSION];
2110         char stored_clid_name[AST_MAX_EXTENSION];
2111         int force_forwards_only;        /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2112         /*!
2113          * \brief Forced CallerID party information to send.
2114          * \note This will not have any malloced strings so do not free it.
2115          */
2116         struct ast_party_id forced_clid;
2117         /*!
2118          * \brief Stored CallerID information if needed.
2119          *
2120          * \note If OPT_ORIGINAL_CLID set then this is the o option
2121          * CallerID.  Otherwise it is the dialplan extension and hint
2122          * name.
2123          *
2124          * \note This will not have any malloced strings so do not free it.
2125          */
2126         struct ast_party_id stored_clid;
2127         /*!
2128          * \brief CallerID party information to store.
2129          * \note This will not have any malloced strings so do not free it.
2130          */
2131         struct ast_party_caller caller;
2132         int max_forwards;
2133
2134         /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2135         ast_channel_lock(chan);
2136         ast_channel_stage_snapshot(chan);
2137         pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2138         pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2139         pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2140         pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2141         pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2142         ast_channel_stage_snapshot_done(chan);
2143         max_forwards = ast_max_forwards_get(chan);
2144         ast_channel_unlock(chan);
2145
2146         if (max_forwards <= 0) {
2147                 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2148                                 ast_channel_name(chan));
2149                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2150                 return -1;
2151         }
2152
2153         if (ast_strlen_zero(data)) {
2154                 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2155                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2156                 return -1;
2157         }
2158
2159         if (ast_check_hangup_locked(chan)) {
2160                 /*
2161                  * Caller hung up before we could dial.  If dial is executed
2162                  * within an AGI then the AGI has likely eaten all queued
2163                  * frames before executing the dial in DeadAGI mode.  With
2164                  * the caller hung up and no pending frames from the caller's
2165                  * read queue, dial would not know that the call has hung up
2166                  * until a called channel answers.  It is rather annoying to
2167                  * whoever just answered the non-existent call.
2168                  *
2169                  * Dial should not continue execution in DeadAGI mode, hangup
2170                  * handlers, or the h exten.
2171                  */
2172                 ast_verb(3, "Caller hung up before dial.\n");
2173                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2174                 return -1;
2175         }
2176
2177         parse = ast_strdupa(data);
2178
2179         AST_STANDARD_APP_ARGS(args, parse);
2180
2181         if (!ast_strlen_zero(args.options) &&
2182                 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2183                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2184                 goto done;
2185         }
2186
2187         if (ast_strlen_zero(args.peers)) {
2188                 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2189                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2190                 goto done;
2191         }
2192
2193         if (ast_cc_call_init(chan, &ignore_cc)) {
2194                 goto done;
2195         }
2196
2197         if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2198                 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2199
2200                 if (delprivintro < 0 || delprivintro > 1) {
2201                         ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2202                         delprivintro = 0;
2203                 }
2204         }
2205
2206         if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2207                 opt_args[OPT_ARG_RINGBACK] = NULL;
2208         }
2209
2210         if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2211                 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2212                 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2213         }
2214
2215         if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2216                 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2217                 if (!calldurationlimit.tv_sec) {
2218                         ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
2219                         pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2220                         goto done;
2221                 }
2222                 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2223         }
2224
2225         if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2226                 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
2227                 dtmfcalled = strsep(&dtmf_progress, ":");
2228                 dtmfcalling = strsep(&dtmf_progress, ":");
2229         }
2230
2231         if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2232                 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2233                         goto done;
2234         }
2235
2236         /* Setup the forced CallerID information to send if used. */
2237         ast_party_id_init(&forced_clid);
2238         force_forwards_only = 0;
2239         if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2240                 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2241                         ast_channel_lock(chan);
2242                         forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2243                         ast_channel_unlock(chan);
2244                         forced_clid_name[0] = '\0';
2245                         forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2246                                 sizeof(forced_clid_name), chan);
2247                         force_forwards_only = 1;
2248                 } else {
2249                         /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2250                         ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2251                                 &forced_clid.number.str);
2252                 }
2253                 if (!ast_strlen_zero(forced_clid.name.str)) {
2254                         forced_clid.name.valid = 1;
2255                 }
2256                 if (!ast_strlen_zero(forced_clid.number.str)) {
2257                         forced_clid.number.valid = 1;
2258                 }
2259         }
2260         if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2261                 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2262                 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2263         }
2264         forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2265         if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2266                 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2267                 int pres;
2268
2269                 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2270                 if (0 <= pres) {
2271                         forced_clid.number.presentation = pres;
2272                 }
2273         }
2274
2275         /* Setup the stored CallerID information if needed. */
2276         ast_party_id_init(&stored_clid);
2277         if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2278                 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2279                         ast_channel_lock(chan);
2280                         ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2281                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2282                                 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2283                         }
2284                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2285                                 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2286                         }
2287                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2288                                 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2289                         }
2290                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2291                                 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2292                         }
2293                         ast_channel_unlock(chan);
2294                 } else {
2295                         /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2296                         ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2297                                 &stored_clid.number.str);
2298                         if (!ast_strlen_zero(stored_clid.name.str)) {
2299                                 stored_clid.name.valid = 1;
2300                         }
2301                         if (!ast_strlen_zero(stored_clid.number.str)) {
2302                                 stored_clid.number.valid = 1;
2303                         }
2304                 }
2305         } else {
2306                 /*
2307                  * In case the new channel has no preset CallerID number by the
2308                  * channel driver, setup the dialplan extension and hint name.
2309                  */
2310                 stored_clid_name[0] = '\0';
2311                 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2312                         sizeof(stored_clid_name), chan);
2313                 if (ast_strlen_zero(stored_clid.name.str)) {
2314                         stored_clid.name.str = NULL;
2315                 } else {
2316                         stored_clid.name.valid = 1;
2317                 }
2318                 ast_channel_lock(chan);
2319                 stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2320                 stored_clid.number.valid = 1;
2321                 ast_channel_unlock(chan);
2322         }
2323
2324         if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2325                 ast_cdr_reset(ast_channel_name(chan), 0);
2326         }
2327         if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2328                 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2329
2330         if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2331                 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2332                 if (res <= 0)
2333                         goto out;
2334                 res = -1; /* reset default */
2335         }
2336
2337         if (continue_exec)
2338                 *continue_exec = 0;
2339
2340         /* If a channel group has been specified, get it for use when we create peer channels */
2341
2342         ast_channel_lock(chan);
2343         if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2344                 outbound_group = ast_strdupa(outbound_group);
2345                 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2346         } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2347                 outbound_group = ast_strdupa(outbound_group);
2348         }
2349         ast_channel_unlock(chan);
2350
2351         /* Set per dial instance flags.  These flags are also passed back to RetryDial. */
2352         ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2353                 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2354                 | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2355
2356         /* PREDIAL: Run gosub on the caller's channel */
2357         if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2358                 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2359                 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2360                 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2361         }
2362
2363         /* loop through the list of dial destinations */
2364         rest = args.peers;
2365         while ((cur = strsep(&rest, "&")) ) {
2366                 struct ast_channel *tc; /* channel for this destination */
2367                 /* Get a technology/resource pair */
2368                 char *number = cur;
2369                 char *tech = strsep(&number, "/");
2370                 size_t tech_len;
2371                 size_t number_len;
2372                 struct ast_format_cap *nativeformats;
2373
2374                 num_dialed++;
2375                 if (ast_strlen_zero(number)) {
2376                         ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2377                         goto out;
2378                 }
2379
2380                 tech_len = strlen(tech) + 1;
2381                 number_len = strlen(number) + 1;
2382                 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2383                 if (!tmp) {
2384                         goto out;
2385                 }
2386
2387                 /* Save tech, number, and interface. */
2388                 cur = tmp->stuff;
2389                 strcpy(cur, tech);
2390                 tmp->tech = cur;
2391                 cur += tech_len;
2392                 strcpy(cur, tech);
2393                 cur[tech_len - 1] = '/';
2394                 tmp->interface = cur;
2395                 cur += tech_len;
2396                 strcpy(cur, number);
2397                 tmp->number = cur;
2398
2399                 if (opts.flags) {
2400                         /* Set per outgoing call leg options. */
2401                         ast_copy_flags64(tmp, &opts,
2402                                 OPT_CANCEL_ELSEWHERE |
2403                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2404                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2405                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2406                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2407                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2408                                 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2409                                 OPT_RING_WITH_EARLY_MEDIA);
2410                         ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2411                 }
2412
2413                 /* Request the peer */
2414
2415                 ast_channel_lock(chan);
2416                 /*
2417                  * Seed the chanlist's connected line information with previously
2418                  * acquired connected line info from the incoming channel.  The
2419                  * previously acquired connected line info could have been set
2420                  * through the CONNECTED_LINE dialplan function.
2421                  */
2422                 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2423
2424                 nativeformats = ao2_bump(ast_channel_nativeformats(chan));
2425
2426                 ast_channel_unlock(chan);
2427
2428                 tc = ast_request(tmp->tech, nativeformats, NULL, chan, tmp->number, &cause);
2429
2430                 ao2_cleanup(nativeformats);
2431
2432                 if (!tc) {
2433                         /* If we can't, just go on to the next call */
2434                         ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
2435                                 tmp->tech, cause, ast_cause2str(cause));
2436                         handle_cause(cause, &num);
2437                         if (!rest) {
2438                                 /* we are on the last destination */
2439                                 ast_channel_hangupcause_set(chan, cause);
2440                         }
2441                         if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2442                                 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2443                                         ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2444                                 }
2445                         }
2446                         chanlist_free(tmp);
2447                         continue;
2448                 }
2449
2450                 ast_channel_lock(tc);
2451                 ast_channel_stage_snapshot(tc);
2452                 ast_channel_unlock(tc);
2453
2454                 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2455                 if (!ignore_cc) {
2456                         ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2457                 }
2458
2459                 ast_channel_lock_both(tc, chan);
2460                 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2461
2462                 /* Setup outgoing SDP to match incoming one */
2463                 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2464                         /* We are on the only destination. */
2465                         ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2466                 }
2467
2468                 /* Inherit specially named variables from parent channel */
2469                 ast_channel_inherit_variables(chan, tc);
2470                 ast_channel_datastore_inherit(chan, tc);
2471                 ast_max_forwards_decrement(tc);
2472
2473                 ast_channel_appl_set(tc, "AppDial");
2474                 ast_channel_data_set(tc, "(Outgoing Line)");
2475                 ast_channel_publish_snapshot(tc);
2476
2477                 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2478
2479                 /* Determine CallerID to store in outgoing channel. */
2480                 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2481                 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2482                         caller.id = stored_clid;
2483                         ast_channel_set_caller_event(tc, &caller, NULL);
2484                         ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2485                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2486                         ast_channel_caller(tc)->id.number.str, NULL))) {
2487                         /*
2488                          * The new channel has no preset CallerID number by the channel
2489                          * driver.  Use the dialplan extension and hint name.
2490                          */
2491                         caller.id = stored_clid;
2492                         if (!caller.id.name.valid
2493                                 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2494                                         ast_channel_connected(chan)->id.name.str, NULL))) {
2495                                 /*
2496                                  * No hint name available.  We have a connected name supplied by
2497                                  * the dialplan we can use instead.
2498                                  */
2499                                 caller.id.name.valid = 1;
2500                                 caller.id.name = ast_channel_connected(chan)->id.name;
2501                         }
2502                         ast_channel_set_caller_event(tc, &caller, NULL);
2503                         ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2504                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2505                         NULL))) {
2506                         /* The new channel has no preset CallerID name by the channel driver. */
2507                         if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2508                                 ast_channel_connected(chan)->id.name.str, NULL))) {
2509                                 /*
2510                                  * We have a connected name supplied by the dialplan we can
2511                                  * use instead.
2512                                  */
2513                                 caller.id.name.valid = 1;
2514                                 caller.id.name = ast_channel_connected(chan)->id.name;
2515                                 ast_channel_set_caller_event(tc, &caller, NULL);
2516                         }
2517                 }
2518
2519                 /* Determine CallerID for outgoing channel to send. */
2520                 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2521                         struct ast_party_connected_line connected;
2522
2523                         ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2524                         connected.id = forced_clid;
2525                         ast_channel_set_connected_line(tc, &connected, NULL);
2526                 } else {
2527                         ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2528                 }
2529
2530                 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2531
2532                 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2533
2534                 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2535                 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2536                         ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));