Merge "voicemail: Move app_voicemail / res_mwi_external conflict to runtime"
[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 <sys/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                 c = o->chan = NULL;
876                 cause = AST_CAUSE_BUSY;
877         } else {
878                 struct ast_format_cap *nativeformats;
879
880                 ast_channel_lock(in);
881                 nativeformats = ao2_bump(ast_channel_nativeformats(in));
882                 ast_channel_unlock(in);
883
884                 /* Setup parameters */
885                 c = o->chan = ast_request(tech, nativeformats, NULL, in, stuff, &cause);
886
887                 ao2_cleanup(nativeformats);
888
889                 if (c) {
890                         if (single && !caller_entertained) {
891                                 ast_channel_make_compatible(in, o->chan);
892                         }
893                         ast_channel_lock_both(in, o->chan);
894                         ast_channel_inherit_variables(in, o->chan);
895                         ast_channel_datastore_inherit(in, o->chan);
896                         ast_max_forwards_decrement(o->chan);
897                         ast_channel_unlock(in);
898                         ast_channel_unlock(o->chan);
899                         /* When a call is forwarded, we don't want to track new interfaces
900                          * dialed for CC purposes. Setting the done flag will ensure that
901                          * any Dial operations that happen later won't record CC interfaces.
902                          */
903                         ast_ignore_cc(o->chan);
904                         ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
905                 } else
906                         ast_log(LOG_NOTICE,
907                                 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
908                                 tech, stuff, cause);
909         }
910         if (!c) {
911                 ast_channel_publish_dial(in, original, stuff, "BUSY");
912                 ast_clear_flag64(o, DIAL_STILLGOING);
913                 handle_cause(cause, num);
914                 ast_hangup(original);
915         } else {
916                 ast_channel_lock_both(c, original);
917                 ast_party_redirecting_copy(ast_channel_redirecting(c),
918                         ast_channel_redirecting(original));
919                 ast_channel_unlock(c);
920                 ast_channel_unlock(original);
921
922                 ast_channel_lock_both(c, in);
923
924                 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
925                         ast_rtp_instance_early_bridge_make_compatible(c, in);
926                 }
927
928                 if (!ast_channel_redirecting(c)->from.number.valid
929                         || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
930                         /*
931                          * The call was not previously redirected so it is
932                          * now redirected from this number.
933                          */
934                         ast_party_number_free(&ast_channel_redirecting(c)->from.number);
935                         ast_party_number_init(&ast_channel_redirecting(c)->from.number);
936                         ast_channel_redirecting(c)->from.number.valid = 1;
937                         ast_channel_redirecting(c)->from.number.str =
938                                 ast_strdup(S_OR(ast_channel_macroexten(in), ast_channel_exten(in)));
939                 }
940
941                 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
942
943                 /* Determine CallerID to store in outgoing channel. */
944                 ast_party_caller_set_init(&caller, ast_channel_caller(c));
945                 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
946                         caller.id = *stored_clid;
947                         ast_channel_set_caller_event(c, &caller, NULL);
948                         ast_set_flag64(o, DIAL_CALLERID_ABSENT);
949                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
950                         ast_channel_caller(c)->id.number.str, NULL))) {
951                         /*
952                          * The new channel has no preset CallerID number by the channel
953                          * driver.  Use the dialplan extension and hint name.
954                          */
955                         caller.id = *stored_clid;
956                         ast_channel_set_caller_event(c, &caller, NULL);
957                         ast_set_flag64(o, DIAL_CALLERID_ABSENT);
958                 } else {
959                         ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
960                 }
961
962                 /* Determine CallerID for outgoing channel to send. */
963                 if (ast_test_flag64(o, OPT_FORCECLID)) {
964                         struct ast_party_connected_line connected;
965
966                         ast_party_connected_line_init(&connected);
967                         connected.id = *forced_clid;
968                         ast_party_connected_line_copy(ast_channel_connected(c), &connected);
969                 } else {
970                         ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
971                 }
972
973                 ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
974
975                 ast_channel_appl_set(c, "AppDial");
976                 ast_channel_data_set(c, "(Outgoing Line)");
977                 ast_channel_publish_snapshot(c);
978
979                 ast_channel_unlock(in);
980                 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
981                         struct ast_party_redirecting redirecting;
982
983                         /*
984                          * Redirecting updates to the caller make sense only on single
985                          * calls.
986                          *
987                          * We must unlock c before calling
988                          * ast_channel_redirecting_macro, because we put c into
989                          * autoservice there.  That is pretty much a guaranteed
990                          * deadlock.  This is why the handling of c's lock may seem a
991                          * bit unusual here.
992                          */
993                         ast_party_redirecting_init(&redirecting);
994                         ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
995                         ast_channel_unlock(c);
996                         if (ast_channel_redirecting_sub(c, in, &redirecting, 0) &&
997                                 ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
998                                 ast_channel_update_redirecting(in, &redirecting, NULL);
999                         }
1000                         ast_party_redirecting_free(&redirecting);
1001                 } else {
1002                         ast_channel_unlock(c);
1003                 }
1004
1005                 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1006                         *to = -1;
1007                 }
1008
1009                 if (ast_call(c, stuff, 0)) {
1010                         ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1011                                 tech, stuff);
1012                         ast_channel_publish_dial(in, original, stuff, "CONGESTION");
1013                         ast_clear_flag64(o, DIAL_STILLGOING);
1014                         ast_hangup(original);
1015                         ast_hangup(c);
1016                         c = o->chan = NULL;
1017                         num->nochan++;
1018                 } else {
1019                         ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
1020                                 ast_channel_call_forward(original));
1021
1022                         ast_channel_publish_dial(in, c, stuff, NULL);
1023
1024                         /* Hangup the original channel now, in case we needed it */
1025                         ast_hangup(original);
1026                 }
1027                 if (single && !caller_entertained) {
1028                         ast_indicate(in, -1);
1029                 }
1030         }
1031 }
1032
1033 /* argument used for some functions. */
1034 struct privacy_args {
1035         int sentringing;
1036         int privdb_val;
1037         char privcid[256];
1038         char privintro[1024];
1039         char status[256];
1040 };
1041
1042 static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
1043 {
1044         struct chanlist *outgoing;
1045         AST_LIST_TRAVERSE(out_chans, outgoing, node) {
1046                 if (!outgoing->chan || outgoing->chan == exception) {
1047                         continue;
1048                 }
1049                 ast_channel_publish_dial(in, outgoing->chan, NULL, status);
1050         }
1051 }
1052
1053 /*!
1054  * \internal
1055  * \brief Update connected line on chan from peer.
1056  * \since 13.6.0
1057  *
1058  * \param chan Channel to get connected line updated.
1059  * \param peer Channel providing connected line information.
1060  * \param is_caller Non-zero if chan is the calling channel.
1061  *
1062  * \return Nothing
1063  */
1064 static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
1065 {
1066         struct ast_party_connected_line connected_caller;
1067
1068         ast_party_connected_line_init(&connected_caller);
1069
1070         ast_channel_lock(peer);
1071         ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
1072         ast_channel_unlock(peer);
1073         connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1074         if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)
1075                 && ast_channel_connected_line_macro(peer, chan, &connected_caller, is_caller, 0)) {
1076                 ast_channel_update_connected_line(chan, &connected_caller, NULL);
1077         }
1078         ast_party_connected_line_free(&connected_caller);
1079 }
1080
1081 static struct ast_channel *wait_for_answer(struct ast_channel *in,
1082         struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
1083         char *opt_args[],
1084         struct privacy_args *pa,
1085         const struct cause_args *num_in, int *result, char *dtmf_progress,
1086         const int ignore_cc,
1087         struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
1088 {
1089         struct cause_args num = *num_in;
1090         int prestart = num.busy + num.congestion + num.nochan;
1091         int orig = *to;
1092         struct ast_channel *peer = NULL;
1093 #ifdef HAVE_EPOLL
1094         struct chanlist *epollo;
1095 #endif
1096         struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1097         /* single is set if only one destination is enabled */
1098         int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1099         int caller_entertained = outgoing
1100                 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1101         struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
1102         int cc_recall_core_id;
1103         int is_cc_recall;
1104         int cc_frame_received = 0;
1105         int num_ringing = 0;
1106         struct timeval start = ast_tvnow();
1107
1108         if (single) {
1109                 /* Turn off hold music, etc */
1110                 if (!caller_entertained) {
1111                         ast_deactivate_generator(in);
1112                         /* If we are calling a single channel, and not providing ringback or music, */
1113                         /* then, make them compatible for in-band tone purpose */
1114                         if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
1115                                 /* If these channels can not be made compatible,
1116                                  * there is no point in continuing.  The bridge
1117                                  * will just fail if it gets that far.
1118                                  */
1119                                 *to = -1;
1120                                 strcpy(pa->status, "CONGESTION");
1121                                 ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
1122                                 return NULL;
1123                         }
1124                 }
1125
1126                 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1127                         && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1128                         update_connected_line_from_peer(in, outgoing->chan, 1);
1129                 }
1130         }
1131
1132         is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1133
1134 #ifdef HAVE_EPOLL
1135         AST_LIST_TRAVERSE(out_chans, epollo, node) {
1136                 ast_poll_channel_add(in, epollo->chan);
1137         }
1138 #endif
1139
1140         while ((*to = ast_remaining_ms(start, orig)) && !peer) {
1141                 struct chanlist *o;
1142                 int pos = 0; /* how many channels do we handle */
1143                 int numlines = prestart;
1144                 struct ast_channel *winner;
1145                 struct ast_channel *watchers[AST_MAX_WATCHERS];
1146
1147                 watchers[pos++] = in;
1148                 AST_LIST_TRAVERSE(out_chans, o, node) {
1149                         /* Keep track of important channels */
1150                         if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1151                                 watchers[pos++] = o->chan;
1152                         numlines++;
1153                 }
1154                 if (pos == 1) { /* only the input channel is available */
1155                         if (numlines == (num.busy + num.congestion + num.nochan)) {
1156                                 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1157                                 if (num.busy)
1158                                         strcpy(pa->status, "BUSY");
1159                                 else if (num.congestion)
1160                                         strcpy(pa->status, "CONGESTION");
1161                                 else if (num.nochan)
1162                                         strcpy(pa->status, "CHANUNAVAIL");
1163                         } else {
1164                                 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1165                         }
1166                         *to = 0;
1167                         if (is_cc_recall) {
1168                                 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1169                         }
1170                         return NULL;
1171                 }
1172                 winner = ast_waitfor_n(watchers, pos, to);
1173                 AST_LIST_TRAVERSE(out_chans, o, node) {
1174                         struct ast_frame *f;
1175                         struct ast_channel *c = o->chan;
1176
1177                         if (c == NULL)
1178                                 continue;
1179                         if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1180                                 if (!peer) {
1181                                         ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1182                                         if (o->orig_chan_name
1183                                                 && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1184                                                 /*
1185                                                  * The channel name changed so we must generate COLP update.
1186                                                  * Likely because a call pickup channel masqueraded in.
1187                                                  */
1188                                                 update_connected_line_from_peer(in, c, 1);
1189                                         } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1190                                                 if (o->pending_connected_update) {
1191                                                         if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1192                                                                 ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1193                                                                 ast_channel_update_connected_line(in, &o->connected, NULL);
1194                                                         }
1195                                                 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1196                                                         update_connected_line_from_peer(in, c, 1);
1197                                                 }
1198                                         }
1199                                         if (o->aoc_s_rate_list) {
1200                                                 size_t encoded_size;
1201                                                 struct ast_aoc_encoded *encoded;
1202                                                 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1203                                                         ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1204                                                         ast_aoc_destroy_encoded(encoded);
1205                                                 }
1206                                         }
1207                                         peer = c;
1208                                         ast_copy_flags64(peerflags, o,
1209                                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1210                                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1211                                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1212                                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1213                                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1214                                                 DIAL_NOFORWARDHTML);
1215                                         ast_channel_dialcontext_set(c, "");
1216                                         ast_channel_exten_set(c, "");
1217                                 }
1218                                 continue;
1219                         }
1220                         if (c != winner)
1221                                 continue;
1222                         /* here, o->chan == c == winner */
1223                         if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1224                                 pa->sentringing = 0;
1225                                 if (!ignore_cc && (f = ast_read(c))) {
1226                                         if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1227                                                 /* This channel is forwarding the call, and is capable of CC, so
1228                                                  * be sure to add the new device interface to the list
1229                                                  */
1230                                                 ast_handle_cc_control_frame(in, c, f->data.ptr);
1231                                         }
1232                                         ast_frfree(f);
1233                                 }
1234
1235                                 if (o->pending_connected_update) {
1236                                         /*
1237                                          * Re-seed the chanlist's connected line information with
1238                                          * previously acquired connected line info from the incoming
1239                                          * channel.  The previously acquired connected line info could
1240                                          * have been set through the CONNECTED_LINE dialplan function.
1241                                          */
1242                                         o->pending_connected_update = 0;
1243                                         ast_channel_lock(in);
1244                                         ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1245                                         ast_channel_unlock(in);
1246                                 }
1247
1248                                 do_forward(o, &num, peerflags, single, caller_entertained, &orig,
1249                                         forced_clid, stored_clid);
1250
1251                                 if (o->chan) {
1252                                         ast_free(o->orig_chan_name);
1253                                         o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
1254                                         if (single
1255                                                 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1256                                                 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1257                                                 update_connected_line_from_peer(in, o->chan, 1);
1258                                         }
1259                                 }
1260                                 continue;
1261                         }
1262                         f = ast_read(winner);
1263                         if (!f) {
1264                                 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1265 #ifdef HAVE_EPOLL
1266                                 ast_poll_channel_del(in, c);
1267 #endif
1268                                 ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
1269                                 ast_hangup(c);
1270                                 c = o->chan = NULL;
1271                                 ast_clear_flag64(o, DIAL_STILLGOING);
1272                                 handle_cause(ast_channel_hangupcause(in), &num);
1273                                 continue;
1274                         }
1275                         switch (f->frametype) {
1276                         case AST_FRAME_CONTROL:
1277                                 switch (f->subclass.integer) {
1278                                 case AST_CONTROL_ANSWER:
1279                                         /* This is our guy if someone answered. */
1280                                         if (!peer) {
1281                                                 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1282                                                 if (o->orig_chan_name
1283                                                         && strcmp(o->orig_chan_name, ast_channel_name(c))) {
1284                                                         /*
1285                                                          * The channel name changed so we must generate COLP update.
1286                                                          * Likely because a call pickup channel masqueraded in.
1287                                                          */
1288                                                         update_connected_line_from_peer(in, c, 1);
1289                                                 } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1290                                                         if (o->pending_connected_update) {
1291                                                                 if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1292                                                                         ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1293                                                                         ast_channel_update_connected_line(in, &o->connected, NULL);
1294                                                                 }
1295                                                         } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1296                                                                 update_connected_line_from_peer(in, c, 1);
1297                                                         }
1298                                                 }
1299                                                 if (o->aoc_s_rate_list) {
1300                                                         size_t encoded_size;
1301                                                         struct ast_aoc_encoded *encoded;
1302                                                         if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1303                                                                 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1304                                                                 ast_aoc_destroy_encoded(encoded);
1305                                                         }
1306                                                 }
1307                                                 peer = c;
1308                                                 /* Inform everyone else that they've been canceled.
1309                                                  * The dial end event for the peer will be sent out after
1310                                                  * other Dial options have been handled.
1311                                                  */
1312                                                 publish_dial_end_event(in, out_chans, peer, "CANCEL");
1313                                                 ast_copy_flags64(peerflags, o,
1314                                                         OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1315                                                         OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1316                                                         OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1317                                                         OPT_CALLEE_PARK | OPT_CALLER_PARK |
1318                                                         OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1319                                                         DIAL_NOFORWARDHTML);
1320                                                 ast_channel_dialcontext_set(c, "");
1321                                                 ast_channel_exten_set(c, "");
1322                                                 if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
1323                                                         /* Setup early bridge if appropriate */
1324                                                         ast_channel_early_bridge(in, peer);
1325                                                 }
1326                                         }
1327                                         /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1328                                         ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1329                                         ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1330                                         break;
1331                                 case AST_CONTROL_BUSY:
1332                                         ast_verb(3, "%s is busy\n", ast_channel_name(c));
1333                                         ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1334                                         ast_channel_publish_dial(in, c, NULL, "BUSY");
1335                                         ast_hangup(c);
1336                                         c = o->chan = NULL;
1337                                         ast_clear_flag64(o, DIAL_STILLGOING);
1338                                         handle_cause(AST_CAUSE_BUSY, &num);
1339                                         break;
1340                                 case AST_CONTROL_CONGESTION:
1341                                         ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1342                                         ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1343                                         ast_channel_publish_dial(in, c, NULL, "CONGESTION");
1344                                         ast_hangup(c);
1345                                         c = o->chan = NULL;
1346                                         ast_clear_flag64(o, DIAL_STILLGOING);
1347                                         handle_cause(AST_CAUSE_CONGESTION, &num);
1348                                         break;
1349                                 case AST_CONTROL_RINGING:
1350                                         /* This is a tricky area to get right when using a native
1351                                          * CC agent. The reason is that we do the best we can to send only a
1352                                          * single ringing notification to the caller.
1353                                          *
1354                                          * Call completion complicates the logic used here. CCNR is typically
1355                                          * offered during a ringing message. Let's say that party A calls
1356                                          * parties B, C, and D. B and C do not support CC requests, but D
1357                                          * does. If we were to receive a ringing notification from B before
1358                                          * the others, then we would end up sending a ringing message to
1359                                          * A with no CCNR offer present.
1360                                          *
1361                                          * The approach that we have taken is that if we receive a ringing
1362                                          * response from a party and no CCNR offer is present, we need to
1363                                          * wait. Specifically, we need to wait until either a) a called party
1364                                          * offers CCNR in its ringing response or b) all called parties have
1365                                          * responded in some way to our call and none offers CCNR.
1366                                          *
1367                                          * The drawback to this is that if one of the parties has a delayed
1368                                          * response or, god forbid, one just plain doesn't respond to our
1369                                          * outgoing call, then this will result in a significant delay between
1370                                          * when the caller places the call and hears ringback.
1371                                          *
1372                                          * Note also that if CC is disabled for this call, then it is perfectly
1373                                          * fine for ringing frames to get sent through.
1374                                          */
1375                                         ++num_ringing;
1376                                         if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1377                                                 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1378                                                 /* Setup early media if appropriate */
1379                                                 if (single && !caller_entertained
1380                                                         && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1381                                                         ast_channel_early_bridge(in, c);
1382                                                 }
1383                                                 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1384                                                         ast_indicate(in, AST_CONTROL_RINGING);
1385                                                         pa->sentringing++;
1386                                                 }
1387                                         }
1388                                         break;
1389                                 case AST_CONTROL_PROGRESS:
1390                                         ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1391                                         /* Setup early media if appropriate */
1392                                         if (single && !caller_entertained
1393                                                 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1394                                                 ast_channel_early_bridge(in, c);
1395                                         }
1396                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1397                                                 if (single || (!single && !pa->sentringing)) {
1398                                                         ast_indicate(in, AST_CONTROL_PROGRESS);
1399                                                 }
1400                                         }
1401                                         if (!ast_strlen_zero(dtmf_progress)) {
1402                                                 ast_verb(3,
1403                                                         "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
1404                                                         dtmf_progress);
1405                                                 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1406                                         }
1407                                         break;
1408                                 case AST_CONTROL_VIDUPDATE:
1409                                 case AST_CONTROL_SRCUPDATE:
1410                                 case AST_CONTROL_SRCCHANGE:
1411                                         if (!single || caller_entertained) {
1412                                                 break;
1413                                         }
1414                                         ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1415                                                 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1416                                         ast_indicate(in, f->subclass.integer);
1417                                         break;
1418                                 case AST_CONTROL_CONNECTED_LINE:
1419                                         if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1420                                                 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1421                                                 break;
1422                                         }
1423                                         if (!single) {
1424                                                 struct ast_party_connected_line connected;
1425
1426                                                 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1427                                                         ast_channel_name(c), ast_channel_name(in));
1428                                                 ast_party_connected_line_set_init(&connected, &o->connected);
1429                                                 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1430                                                 ast_party_connected_line_set(&o->connected, &connected, NULL);
1431                                                 ast_party_connected_line_free(&connected);
1432                                                 o->pending_connected_update = 1;
1433                                                 break;
1434                                         }
1435                                         if (ast_channel_connected_line_sub(c, in, f, 1) &&
1436                                                 ast_channel_connected_line_macro(c, in, f, 1, 1)) {
1437                                                 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1438                                         }
1439                                         break;
1440                                 case AST_CONTROL_AOC:
1441                                         {
1442                                                 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1443                                                 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1444                                                         ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1445                                                         o->aoc_s_rate_list = decoded;
1446                                                 } else {
1447                                                         ast_aoc_destroy_decoded(decoded);
1448                                                 }
1449                                         }
1450                                         break;
1451                                 case AST_CONTROL_REDIRECTING:
1452                                         if (!single) {
1453                                                 /*
1454                                                  * Redirecting updates to the caller make sense only on single
1455                                                  * calls.
1456                                                  */
1457                                                 break;
1458                                         }
1459                                         if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1460                                                 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1461                                                 break;
1462                                         }
1463                                         ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1464                                                 ast_channel_name(c), ast_channel_name(in));
1465                                         if (ast_channel_redirecting_sub(c, in, f, 1) &&
1466                                                 ast_channel_redirecting_macro(c, in, f, 1, 1)) {
1467                                                 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1468                                         }
1469                                         pa->sentringing = 0;
1470                                         break;
1471                                 case AST_CONTROL_PROCEEDING:
1472                                         ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1473                                         if (single && !caller_entertained
1474                                                 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1475                                                 ast_channel_early_bridge(in, c);
1476                                         }
1477                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1478                                                 ast_indicate(in, AST_CONTROL_PROCEEDING);
1479                                         break;
1480                                 case AST_CONTROL_HOLD:
1481                                         /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1482                                         ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1483                                         ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1484                                         break;
1485                                 case AST_CONTROL_UNHOLD:
1486                                         /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1487                                         ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1488                                         ast_indicate(in, AST_CONTROL_UNHOLD);
1489                                         break;
1490                                 case AST_CONTROL_OFFHOOK:
1491                                 case AST_CONTROL_FLASH:
1492                                         /* Ignore going off hook and flash */
1493                                         break;
1494                                 case AST_CONTROL_CC:
1495                                         if (!ignore_cc) {
1496                                                 ast_handle_cc_control_frame(in, c, f->data.ptr);
1497                                                 cc_frame_received = 1;
1498                                         }
1499                                         break;
1500                                 case AST_CONTROL_PVT_CAUSE_CODE:
1501                                         ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1502                                         break;
1503                                 case -1:
1504                                         if (single && !caller_entertained) {
1505                                                 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1506                                                 ast_indicate(in, -1);
1507                                                 pa->sentringing = 0;
1508                                         }
1509                                         break;
1510                                 default:
1511                                         ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1512                                         break;
1513                                 }
1514                                 break;
1515                         case AST_FRAME_VOICE:
1516                         case AST_FRAME_IMAGE:
1517                                 if (caller_entertained) {
1518                                         break;
1519                                 }
1520                                 /* Fall through */
1521                         case AST_FRAME_TEXT:
1522                                 if (single && ast_write(in, f)) {
1523                                         ast_log(LOG_WARNING, "Unable to write frametype: %u\n",
1524                                                 f->frametype);
1525                                 }
1526                                 break;
1527                         case AST_FRAME_HTML:
1528                                 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1529                                         && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1530                                         ast_log(LOG_WARNING, "Unable to send URL\n");
1531                                 }
1532                                 break;
1533                         default:
1534                                 break;
1535                         }
1536                         ast_frfree(f);
1537                 } /* end for */
1538                 if (winner == in) {
1539                         struct ast_frame *f = ast_read(in);
1540 #if 0
1541                         if (f && (f->frametype != AST_FRAME_VOICE))
1542                                 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1543                         else if (!f || (f->frametype != AST_FRAME_VOICE))
1544                                 printf("Hangup received on %s\n", in->name);
1545 #endif
1546                         if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1547                                 /* Got hung up */
1548                                 *to = -1;
1549                                 strcpy(pa->status, "CANCEL");
1550                                 publish_dial_end_event(in, out_chans, NULL, pa->status);
1551                                 if (f) {
1552                                         if (f->data.uint32) {
1553                                                 ast_channel_hangupcause_set(in, f->data.uint32);
1554                                         }
1555                                         ast_frfree(f);
1556                                 }
1557                                 if (is_cc_recall) {
1558                                         ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1559                                 }
1560                                 return NULL;
1561                         }
1562
1563                         /* now f is guaranteed non-NULL */
1564                         if (f->frametype == AST_FRAME_DTMF) {
1565                                 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1566                                         const char *context;
1567                                         ast_channel_lock(in);
1568                                         context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1569                                         if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1570                                                 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1571                                                 *to = 0;
1572                                                 *result = f->subclass.integer;
1573                                                 strcpy(pa->status, "CANCEL");
1574                                                 publish_dial_end_event(in, out_chans, NULL, pa->status);
1575                                                 ast_frfree(f);
1576                                                 ast_channel_unlock(in);
1577                                                 if (is_cc_recall) {
1578                                                         ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1579                                                 }
1580                                                 return NULL;
1581                                         }
1582                                         ast_channel_unlock(in);
1583                                 }
1584
1585                                 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1586                                         detect_disconnect(in, f->subclass.integer, &featurecode)) {
1587                                         ast_verb(3, "User requested call disconnect.\n");
1588                                         *to = 0;
1589                                         strcpy(pa->status, "CANCEL");
1590                                         publish_dial_end_event(in, out_chans, NULL, pa->status);
1591                                         ast_frfree(f);
1592                                         if (is_cc_recall) {
1593                                                 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1594                                         }
1595                                         return NULL;
1596                                 }
1597                         }
1598
1599                         /* Send the frame from the in channel to all outgoing channels. */
1600                         AST_LIST_TRAVERSE(out_chans, o, node) {
1601                                 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1602                                         /* This outgoing channel has died so don't send the frame to it. */
1603                                         continue;
1604                                 }
1605                                 switch (f->frametype) {
1606                                 case AST_FRAME_HTML:
1607                                         /* Forward HTML stuff */
1608                                         if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1609                                                 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1610                                                 ast_log(LOG_WARNING, "Unable to send URL\n");
1611                                         }
1612                                         break;
1613                                 case AST_FRAME_VOICE:
1614                                 case AST_FRAME_IMAGE:
1615                                         if (!single || caller_entertained) {
1616                                                 /*
1617                                                  * We are calling multiple parties or caller is being
1618                                                  * entertained and has thus not been made compatible.
1619                                                  * No need to check any other called parties.
1620                                                  */
1621                                                 goto skip_frame;
1622                                         }
1623                                         /* Fall through */
1624                                 case AST_FRAME_TEXT:
1625                                 case AST_FRAME_DTMF_BEGIN:
1626                                 case AST_FRAME_DTMF_END:
1627                                         if (ast_write(o->chan, f)) {
1628                                                 ast_log(LOG_WARNING, "Unable to forward frametype: %u\n",
1629                                                         f->frametype);
1630                                         }
1631                                         break;
1632                                 case AST_FRAME_CONTROL:
1633                                         switch (f->subclass.integer) {
1634                                         case AST_CONTROL_HOLD:
1635                                                 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1636                                                 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1637                                                 break;
1638                                         case AST_CONTROL_UNHOLD:
1639                                                 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1640                                                 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1641                                                 break;
1642                                         case AST_CONTROL_VIDUPDATE:
1643                                         case AST_CONTROL_SRCUPDATE:
1644                                         case AST_CONTROL_SRCCHANGE:
1645                                                 if (!single || caller_entertained) {
1646                                                         /*
1647                                                          * We are calling multiple parties or caller is being
1648                                                          * entertained and has thus not been made compatible.
1649                                                          * No need to check any other called parties.
1650                                                          */
1651                                                         goto skip_frame;
1652                                                 }
1653                                                 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1654                                                         ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1655                                                 ast_indicate(o->chan, f->subclass.integer);
1656                                                 break;
1657                                         case AST_CONTROL_CONNECTED_LINE:
1658                                                 if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
1659                                                         ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
1660                                                         ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1661                                                 }
1662                                                 break;
1663                                         case AST_CONTROL_REDIRECTING:
1664                                                 if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
1665                                                         ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
1666                                                         ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1667                                                 }
1668                                                 break;
1669                                         default:
1670                                                 /* We are not going to do anything with this frame. */
1671                                                 goto skip_frame;
1672                                         }
1673                                         break;
1674                                 default:
1675                                         /* We are not going to do anything with this frame. */
1676                                         goto skip_frame;
1677                                 }
1678                         }
1679 skip_frame:;
1680                         ast_frfree(f);
1681                 }
1682         }
1683
1684         if (!*to || ast_check_hangup(in)) {
1685                 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1686                 publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
1687         }
1688
1689 #ifdef HAVE_EPOLL
1690         AST_LIST_TRAVERSE(out_chans, epollo, node) {
1691                 if (epollo->chan)
1692                         ast_poll_channel_del(in, epollo->chan);
1693         }
1694 #endif
1695
1696         if (is_cc_recall) {
1697                 ast_cc_completed(in, "Recall completed!");
1698         }
1699         return peer;
1700 }
1701
1702 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1703 {
1704         char disconnect_code[AST_FEATURE_MAX_LEN];
1705         int res;
1706
1707         ast_str_append(featurecode, 1, "%c", code);
1708
1709         res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
1710         if (res) {
1711                 ast_str_reset(*featurecode);
1712                 return 0;
1713         }
1714
1715         if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
1716                 /* Could be a partial match, anyway */
1717                 if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
1718                         ast_str_reset(*featurecode);
1719                 }
1720                 return 0;
1721         }
1722
1723         if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
1724                 ast_str_reset(*featurecode);
1725                 return 0;
1726         }
1727
1728         return 1;
1729 }
1730
1731 /* returns true if there is a valid privacy reply */
1732 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1733 {
1734         if (res < '1')
1735                 return 0;
1736         if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1737                 return 1;
1738         if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1739                 return 1;
1740         return 0;
1741 }
1742
1743 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1744         struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1745 {
1746
1747         int res2;
1748         int loopcount = 0;
1749
1750         /* Get the user's intro, store it in priv-callerintros/$CID,
1751            unless it is already there-- this should be done before the
1752            call is actually dialed  */
1753
1754         /* all ring indications and moh for the caller has been halted as soon as the
1755            target extension was picked up. We are going to have to kill some
1756            time and make the caller believe the peer hasn't picked up yet */
1757
1758         if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1759                 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
1760                 ast_indicate(chan, -1);
1761                 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
1762                 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1763                 ast_channel_musicclass_set(chan, original_moh);
1764         } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
1765                 ast_indicate(chan, AST_CONTROL_RINGING);
1766                 pa->sentringing++;
1767         }
1768
1769         /* Start autoservice on the other chan ?? */
1770         res2 = ast_autoservice_start(chan);
1771         /* Now Stream the File */
1772         for (loopcount = 0; loopcount < 3; loopcount++) {
1773                 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1774                         break;
1775                 if (!res2) /* on timeout, play the message again */
1776                         res2 = ast_play_and_wait(peer, "priv-callpending");
1777                 if (!valid_priv_reply(opts, res2))
1778                         res2 = 0;
1779                 /* priv-callpending script:
1780                    "I have a caller waiting, who introduces themselves as:"
1781                 */
1782                 if (!res2)
1783                         res2 = ast_play_and_wait(peer, pa->privintro);
1784                 if (!valid_priv_reply(opts, res2))
1785                         res2 = 0;
1786                 /* now get input from the called party, as to their choice */
1787                 if (!res2) {
1788                         /* XXX can we have both, or they are mutually exclusive ? */
1789                         if (ast_test_flag64(opts, OPT_PRIVACY))
1790                                 res2 = ast_play_and_wait(peer, "priv-callee-options");
1791                         if (ast_test_flag64(opts, OPT_SCREENING))
1792                                 res2 = ast_play_and_wait(peer, "screen-callee-options");
1793                 }
1794
1795                 /*! \page DialPrivacy Dial Privacy scripts
1796                  * \par priv-callee-options script:
1797                  * \li Dial 1 if you wish this caller to reach you directly in the future,
1798                  *      and immediately connect to their incoming call.
1799                  * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
1800                  * \li Dial 3 to send this caller to the torture menus, now and forevermore.
1801                  * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1802                  * \li Dial 5 to allow this caller to come straight thru to you in the future,
1803                  *      but right now, just this once, send them to voicemail.
1804                  *
1805                  * \par screen-callee-options script:
1806                  * \li Dial 1 if you wish to immediately connect to the incoming call
1807                  * \li Dial 2 if you wish to send this caller to voicemail.
1808                  * \li Dial 3 to send this caller to the torture menus.
1809                  * \li Dial 4 to send this caller to a simple "go away" menu.
1810                  */
1811                 if (valid_priv_reply(opts, res2))
1812                         break;
1813                 /* invalid option */
1814                 res2 = ast_play_and_wait(peer, "vm-sorry");
1815         }
1816
1817         if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1818                 ast_moh_stop(chan);
1819         } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
1820                 ast_indicate(chan, -1);
1821                 pa->sentringing = 0;
1822         }
1823         ast_autoservice_stop(chan);
1824         if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1825                 /* map keypresses to various things, the index is res2 - '1' */
1826                 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1827                 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1828                 int i = res2 - '1';
1829                 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1830                         opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1831                 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1832         }
1833         switch (res2) {
1834         case '1':
1835                 break;
1836         case '2':
1837                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1838                 break;
1839         case '3':
1840                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1841                 break;
1842         case '4':
1843                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1844                 break;
1845         case '5':
1846                 /* XXX should we set status to DENY ? */
1847                 if (ast_test_flag64(opts, OPT_PRIVACY))
1848                         break;
1849                 /* if not privacy, then 5 is the same as "default" case */
1850         default: /* bad input or -1 if failure to start autoservice */
1851                 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do?  */
1852                 /* well, there seems basically two choices. Just patch the caller thru immediately,
1853                           or,... put 'em thru to voicemail. */
1854                 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1855                 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1856                 /* XXX should we set status to DENY ? */
1857                 /* XXX what about the privacy flags ? */
1858                 break;
1859         }
1860
1861         if (res2 == '1') { /* the only case where we actually connect */
1862                 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1863                    just clog things up, and it's not useful information, not being tied to a CID */
1864                 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1865                         ast_filedelete(pa->privintro, NULL);
1866                         if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1867                                 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1868                         else
1869                                 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1870                 }
1871                 return 0; /* the good exit path */
1872         } else {
1873                 /* hang up on the callee -- he didn't want to talk anyway! */
1874                 ast_autoservice_chan_hangup_peer(chan, peer);
1875                 return -1;
1876         }
1877 }
1878
1879 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1880 static int setup_privacy_args(struct privacy_args *pa,
1881         struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1882 {
1883         char callerid[60];
1884         int res;
1885         char *l;
1886
1887         if (ast_channel_caller(chan)->id.number.valid
1888                 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
1889                 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
1890                 ast_shrink_phone_number(l);
1891                 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1892                         ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1893                         pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1894                 } else {
1895                         ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1896                         pa->privdb_val = AST_PRIVACY_UNKNOWN;
1897                 }
1898         } else {
1899                 char *tnam, *tn2;
1900
1901                 tnam = ast_strdupa(ast_channel_name(chan));
1902                 /* clean the channel name so slashes don't try to end up in disk file name */
1903                 for (tn2 = tnam; *tn2; tn2++) {
1904                         if (*tn2 == '/')  /* any other chars to be afraid of? */
1905                                 *tn2 = '=';
1906                 }
1907                 ast_verb(3, "Privacy-- callerid is empty\n");
1908
1909                 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
1910                 l = callerid;
1911                 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1912         }
1913
1914         ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1915
1916         if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1917                 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1918                 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1919                 pa->privdb_val = AST_PRIVACY_ALLOW;
1920         } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1921                 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1922         }
1923
1924         if (pa->privdb_val == AST_PRIVACY_DENY) {
1925                 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1926                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1927                 return 0;
1928         } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1929                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1930                 return 0; /* Is this right? */
1931         } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1932                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1933                 return 0; /* is this right??? */
1934         } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1935                 /* Get the user's intro, store it in priv-callerintros/$CID,
1936                    unless it is already there-- this should be done before the
1937                    call is actually dialed  */
1938
1939                 /* make sure the priv-callerintros dir actually exists */
1940                 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1941                 if ((res = ast_mkdir(pa->privintro, 0755))) {
1942                         ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1943                         return -1;
1944                 }
1945
1946                 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1947                 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1948                         /* the DELUX version of this code would allow this caller the
1949                            option to hear and retape their previously recorded intro.
1950                         */
1951                 } else {
1952                         int duration; /* for feedback from play_and_wait */
1953                         /* the file doesn't exist yet. Let the caller submit his
1954                            vocal intro for posterity */
1955                         /* priv-recordintro script:
1956
1957                            "At the tone, please say your name:"
1958
1959                         */
1960                         int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1961                         ast_answer(chan);
1962                         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 */
1963                                                                         /* don't think we'll need a lock removed, we took care of
1964                                                                            conflicts by naming the pa.privintro file */
1965                         if (res == -1) {
1966                                 /* Delete the file regardless since they hung up during recording */
1967                                 ast_filedelete(pa->privintro, NULL);
1968                                 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1969                                         ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1970                                 else
1971                                         ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1972                                 return -1;
1973                         }
1974                         if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
1975                                 ast_waitstream(chan, "");
1976                 }
1977         }
1978         return 1; /* success */
1979 }
1980
1981 static void end_bridge_callback(void *data)
1982 {
1983         char buf[80];
1984         time_t end;
1985         struct ast_channel *chan = data;
1986
1987         time(&end);
1988
1989         ast_channel_lock(chan);
1990         ast_channel_stage_snapshot(chan);
1991         snprintf(buf, sizeof(buf), "%d", ast_channel_get_up_time(chan));
1992         pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1993         snprintf(buf, sizeof(buf), "%d", ast_channel_get_duration(chan));
1994         pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1995         ast_channel_stage_snapshot_done(chan);
1996         ast_channel_unlock(chan);
1997 }
1998
1999 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2000         bconfig->end_bridge_callback_data = originator;
2001 }
2002
2003 static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2004 {
2005         struct ast_tone_zone_sound *ts = NULL;
2006         int res;
2007         const char *str = data;
2008
2009         if (ast_strlen_zero(str)) {
2010                 ast_debug(1,"Nothing to play\n");
2011                 return -1;
2012         }
2013
2014         ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2015
2016         if (ts && ts->data[0]) {
2017                 res = ast_playtones_start(chan, 0, ts->data, 0);
2018         } else {
2019                 res = -1;
2020         }
2021
2022         if (ts) {
2023                 ts = ast_tone_zone_sound_unref(ts);
2024         }
2025
2026         if (res) {
2027                 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2028         }
2029
2030         return res;
2031 }
2032
2033 /*!
2034  * \internal
2035  * \brief Setup the after bridge goto location on the peer.
2036  * \since 12.0.0
2037  *
2038  * \param chan Calling channel for bridge.
2039  * \param peer Peer channel for bridge.
2040  * \param opts Dialing option flags.
2041  * \param opt_args Dialing option argument strings.
2042  *
2043  * \return Nothing
2044  */
2045 static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
2046 {
2047         const char *context;
2048         const char *extension;
2049         int priority;
2050
2051         if (ast_test_flag64(opts, OPT_PEER_H)) {
2052                 ast_channel_lock(chan);
2053                 context = ast_strdupa(ast_channel_context(chan));
2054                 ast_channel_unlock(chan);
2055                 ast_bridge_set_after_h(peer, context);
2056         } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
2057                 ast_channel_lock(chan);
2058                 context = ast_strdupa(ast_channel_context(chan));
2059                 extension = ast_strdupa(ast_channel_exten(chan));
2060                 priority = ast_channel_priority(chan);
2061                 ast_channel_unlock(chan);
2062                 ast_bridge_set_after_go_on(peer, context, extension, priority,
2063                         opt_args[OPT_ARG_CALLEE_GO_ON]);
2064         }
2065 }
2066
2067 static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2068 {
2069         int res = -1; /* default: error */
2070         char *rest, *cur; /* scan the list of destinations */
2071         struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2072         struct chanlist *outgoing;
2073         struct chanlist *tmp;
2074         struct ast_channel *peer;
2075         int to; /* timeout */
2076         struct cause_args num = { chan, 0, 0, 0 };
2077         int cause;
2078
2079         struct ast_bridge_config config = { { 0, } };
2080         struct timeval calldurationlimit = { 0, };
2081         char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
2082         struct privacy_args pa = {
2083                 .sentringing = 0,
2084                 .privdb_val = 0,
2085                 .status = "INVALIDARGS",
2086         };
2087         int sentringing = 0, moh = 0;
2088         const char *outbound_group = NULL;
2089         int result = 0;
2090         char *parse;
2091         int opermode = 0;
2092         int delprivintro = 0;
2093         AST_DECLARE_APP_ARGS(args,
2094                 AST_APP_ARG(peers);
2095                 AST_APP_ARG(timeout);
2096                 AST_APP_ARG(options);
2097                 AST_APP_ARG(url);
2098         );
2099         struct ast_flags64 opts = { 0, };
2100         char *opt_args[OPT_ARG_ARRAY_SIZE];
2101         int fulldial = 0, num_dialed = 0;
2102         int ignore_cc = 0;
2103         char device_name[AST_CHANNEL_NAME];
2104         char forced_clid_name[AST_MAX_EXTENSION];
2105         char stored_clid_name[AST_MAX_EXTENSION];
2106         int force_forwards_only;        /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2107         /*!
2108          * \brief Forced CallerID party information to send.
2109          * \note This will not have any malloced strings so do not free it.
2110          */
2111         struct ast_party_id forced_clid;
2112         /*!
2113          * \brief Stored CallerID information if needed.
2114          *
2115          * \note If OPT_ORIGINAL_CLID set then this is the o option
2116          * CallerID.  Otherwise it is the dialplan extension and hint
2117          * name.
2118          *
2119          * \note This will not have any malloced strings so do not free it.
2120          */
2121         struct ast_party_id stored_clid;
2122         /*!
2123          * \brief CallerID party information to store.
2124          * \note This will not have any malloced strings so do not free it.
2125          */
2126         struct ast_party_caller caller;
2127         int max_forwards;
2128
2129         /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2130         ast_channel_lock(chan);
2131         ast_channel_stage_snapshot(chan);
2132         pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2133         pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2134         pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2135         pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2136         pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2137         ast_channel_stage_snapshot_done(chan);
2138         max_forwards = ast_max_forwards_get(chan);
2139         ast_channel_unlock(chan);
2140
2141         if (max_forwards <= 0) {
2142                 ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
2143                                 ast_channel_name(chan));
2144                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
2145                 return -1;
2146         }
2147
2148         if (ast_strlen_zero(data)) {
2149                 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2150                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2151                 return -1;
2152         }
2153
2154         if (ast_check_hangup_locked(chan)) {
2155                 /*
2156                  * Caller hung up before we could dial.  If dial is executed
2157                  * within an AGI then the AGI has likely eaten all queued
2158                  * frames before executing the dial in DeadAGI mode.  With
2159                  * the caller hung up and no pending frames from the caller's
2160                  * read queue, dial would not know that the call has hung up
2161                  * until a called channel answers.  It is rather annoying to
2162                  * whoever just answered the non-existent call.
2163                  *
2164                  * Dial should not continue execution in DeadAGI mode, hangup
2165                  * handlers, or the h exten.
2166                  */
2167                 ast_verb(3, "Caller hung up before dial.\n");
2168                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
2169                 return -1;
2170         }
2171
2172         parse = ast_strdupa(data);
2173
2174         AST_STANDARD_APP_ARGS(args, parse);
2175
2176         if (!ast_strlen_zero(args.options) &&
2177                 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2178                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2179                 goto done;
2180         }
2181
2182         if (ast_strlen_zero(args.peers)) {
2183                 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2184                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2185                 goto done;
2186         }
2187
2188         if (ast_cc_call_init(chan, &ignore_cc)) {
2189                 goto done;
2190         }
2191
2192         if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2193                 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2194
2195                 if (delprivintro < 0 || delprivintro > 1) {
2196                         ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2197                         delprivintro = 0;
2198                 }
2199         }
2200
2201         if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2202                 opt_args[OPT_ARG_RINGBACK] = NULL;
2203         }
2204
2205         if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2206                 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2207                 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2208         }
2209
2210         if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2211                 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2212                 if (!calldurationlimit.tv_sec) {
2213                         ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
2214                         pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2215                         goto done;
2216                 }
2217                 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2218         }
2219
2220         if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2221                 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
2222                 dtmfcalled = strsep(&dtmf_progress, ":");
2223                 dtmfcalling = strsep(&dtmf_progress, ":");
2224         }
2225
2226         if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2227                 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2228                         goto done;
2229         }
2230
2231         /* Setup the forced CallerID information to send if used. */
2232         ast_party_id_init(&forced_clid);
2233         force_forwards_only = 0;
2234         if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2235                 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2236                         ast_channel_lock(chan);
2237                         forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2238                         ast_channel_unlock(chan);
2239                         forced_clid_name[0] = '\0';
2240                         forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2241                                 sizeof(forced_clid_name), chan);
2242                         force_forwards_only = 1;
2243                 } else {
2244                         /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2245                         ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2246                                 &forced_clid.number.str);
2247                 }
2248                 if (!ast_strlen_zero(forced_clid.name.str)) {
2249                         forced_clid.name.valid = 1;
2250                 }
2251                 if (!ast_strlen_zero(forced_clid.number.str)) {
2252                         forced_clid.number.valid = 1;
2253                 }
2254         }
2255         if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2256                 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2257                 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2258         }
2259         forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2260         if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2261                 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2262                 int pres;
2263
2264                 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2265                 if (0 <= pres) {
2266                         forced_clid.number.presentation = pres;
2267                 }
2268         }
2269
2270         /* Setup the stored CallerID information if needed. */
2271         ast_party_id_init(&stored_clid);
2272         if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2273                 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2274                         ast_channel_lock(chan);
2275                         ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2276                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2277                                 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2278                         }
2279                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2280                                 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2281                         }
2282                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2283                                 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2284                         }
2285                         if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2286                                 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2287                         }
2288                         ast_channel_unlock(chan);
2289                 } else {
2290                         /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2291                         ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2292                                 &stored_clid.number.str);
2293                         if (!ast_strlen_zero(stored_clid.name.str)) {
2294                                 stored_clid.name.valid = 1;
2295                         }
2296                         if (!ast_strlen_zero(stored_clid.number.str)) {
2297                                 stored_clid.number.valid = 1;
2298                         }
2299                 }
2300         } else {
2301                 /*
2302                  * In case the new channel has no preset CallerID number by the
2303                  * channel driver, setup the dialplan extension and hint name.
2304                  */
2305                 stored_clid_name[0] = '\0';
2306                 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2307                         sizeof(stored_clid_name), chan);
2308                 if (ast_strlen_zero(stored_clid.name.str)) {
2309                         stored_clid.name.str = NULL;
2310                 } else {
2311                         stored_clid.name.valid = 1;
2312                 }
2313                 ast_channel_lock(chan);
2314                 stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2315                 stored_clid.number.valid = 1;
2316                 ast_channel_unlock(chan);
2317         }
2318
2319         if (ast_test_flag64(&opts, OPT_RESETCDR)) {
2320                 ast_cdr_reset(ast_channel_name(chan), 0);
2321         }
2322         if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2323                 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2324
2325         if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2326                 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2327                 if (res <= 0)
2328                         goto out;
2329                 res = -1; /* reset default */
2330         }
2331
2332         if (continue_exec)
2333                 *continue_exec = 0;
2334
2335         /* If a channel group has been specified, get it for use when we create peer channels */
2336
2337         ast_channel_lock(chan);
2338         if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2339                 outbound_group = ast_strdupa(outbound_group);
2340                 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2341         } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2342                 outbound_group = ast_strdupa(outbound_group);
2343         }
2344         ast_channel_unlock(chan);
2345
2346         /* Set per dial instance flags.  These flags are also passed back to RetryDial. */
2347         ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2348                 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2349                 | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2350
2351         /* PREDIAL: Run gosub on the caller's channel */
2352         if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2353                 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2354                 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2355                 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2356         }
2357
2358         /* loop through the list of dial destinations */
2359         rest = args.peers;
2360         while ((cur = strsep(&rest, "&")) ) {
2361                 struct ast_channel *tc; /* channel for this destination */
2362                 /* Get a technology/resource pair */
2363                 char *number = cur;
2364                 char *tech = strsep(&number, "/");
2365                 size_t tech_len;
2366                 size_t number_len;
2367                 struct ast_format_cap *nativeformats;
2368
2369                 num_dialed++;
2370                 if (ast_strlen_zero(number)) {
2371                         ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2372                         goto out;
2373                 }
2374
2375                 tech_len = strlen(tech) + 1;
2376                 number_len = strlen(number) + 1;
2377                 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2378                 if (!tmp) {
2379                         goto out;
2380                 }
2381
2382                 /* Save tech, number, and interface. */
2383                 cur = tmp->stuff;
2384                 strcpy(cur, tech);
2385                 tmp->tech = cur;
2386                 cur += tech_len;
2387                 strcpy(cur, tech);
2388                 cur[tech_len - 1] = '/';
2389                 tmp->interface = cur;
2390                 cur += tech_len;
2391                 strcpy(cur, number);
2392                 tmp->number = cur;
2393
2394                 if (opts.flags) {
2395                         /* Set per outgoing call leg options. */
2396                         ast_copy_flags64(tmp, &opts,
2397                                 OPT_CANCEL_ELSEWHERE |
2398                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2399                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2400                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2401                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2402                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2403                                 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
2404                                 OPT_RING_WITH_EARLY_MEDIA);
2405                         ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2406                 }
2407
2408                 /* Request the peer */
2409
2410                 ast_channel_lock(chan);
2411                 /*
2412                  * Seed the chanlist's connected line information with previously
2413                  * acquired connected line info from the incoming channel.  The
2414                  * previously acquired connected line info could have been set
2415                  * through the CONNECTED_LINE dialplan function.
2416                  */
2417                 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2418
2419                 nativeformats = ao2_bump(ast_channel_nativeformats(chan));
2420
2421                 ast_channel_unlock(chan);
2422
2423                 tc = ast_request(tmp->tech, nativeformats, NULL, chan, tmp->number, &cause);
2424
2425                 ao2_cleanup(nativeformats);
2426
2427                 if (!tc) {
2428                         /* If we can't, just go on to the next call */
2429                         ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
2430                                 tmp->tech, cause, ast_cause2str(cause));
2431                         handle_cause(cause, &num);
2432                         if (!rest) {
2433                                 /* we are on the last destination */
2434                                 ast_channel_hangupcause_set(chan, cause);
2435                         }
2436                         if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2437                                 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2438                                         ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2439                                 }
2440                         }
2441                         chanlist_free(tmp);
2442                         continue;
2443                 }
2444
2445                 ast_channel_lock(tc);
2446                 ast_channel_stage_snapshot(tc);
2447                 ast_channel_unlock(tc);
2448
2449                 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2450                 if (!ignore_cc) {
2451                         ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2452                 }
2453
2454                 ast_channel_lock_both(tc, chan);
2455                 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2456
2457                 /* Setup outgoing SDP to match incoming one */
2458                 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2459                         /* We are on the only destination. */
2460                         ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2461                 }
2462
2463                 /* Inherit specially named variables from parent channel */
2464                 ast_channel_inherit_variables(chan, tc);
2465                 ast_channel_datastore_inherit(chan, tc);
2466                 ast_max_forwards_decrement(tc);
2467
2468                 ast_channel_appl_set(tc, "AppDial");
2469                 ast_channel_data_set(tc, "(Outgoing Line)");
2470                 ast_channel_publish_snapshot(tc);
2471
2472                 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2473
2474                 /* Determine CallerID to store in outgoing channel. */
2475                 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2476                 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2477                         caller.id = stored_clid;
2478                         ast_channel_set_caller_event(tc, &caller, NULL);
2479                         ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2480                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2481                         ast_channel_caller(tc)->id.number.str, NULL))) {
2482                         /*
2483                          * The new channel has no preset CallerID number by the channel
2484                          * driver.  Use the dialplan extension and hint name.
2485                          */
2486                         caller.id = stored_clid;
2487                         if (!caller.id.name.valid
2488                                 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2489                                         ast_channel_connected(chan)->id.name.str, NULL))) {
2490                                 /*
2491                                  * No hint name available.  We have a connected name supplied by
2492                                  * the dialplan we can use instead.
2493                                  */
2494                                 caller.id.name.valid = 1;
2495                                 caller.id.name = ast_channel_connected(chan)->id.name;
2496                         }
2497                         ast_channel_set_caller_event(tc, &caller, NULL);
2498                         ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2499                 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2500                         NULL))) {
2501                         /* The new channel has no preset CallerID name by the channel driver. */
2502                         if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2503                                 ast_channel_connected(chan)->id.name.str, NULL))) {
2504                                 /*
2505                                  * We have a connected name supplied by the dialplan we can
2506                                  * use instead.
2507                                  */
2508                                 caller.id.name.valid = 1;
2509                                 caller.id.name = ast_channel_connected(chan)->id.name;
2510                                 ast_channel_set_caller_event(tc, &caller, NULL);
2511                         }
2512                 }
2513
2514                 /* Determine CallerID for outgoing channel to send. */
2515                 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
2516                         struct ast_party_connected_line connected;
2517
2518                         ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
2519                         connected.id = forced_clid;
2520                         ast_channel_set_connected_line(tc, &connected, NULL);
2521                 } else {
2522                         ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
2523                 }
2524
2525                 ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
2526
2527                 ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
2528
2529                 ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
2530                 if (ast_strlen_zero(ast_channel_musicclass(tc))) {
2531                         ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
2532                 }
2533
2534                 /* Pass ADSI CPE and transfer capability */
2535                 ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
2536                 ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
2537
2538                 /* If we have an outbound group, set this peer channel to it&nbs