2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2012, Digium, Inc.
6 * Mark Spencer <markster@digium.com>
8 * See http://www.asterisk.org for more information about
9 * the Asterisk project. Please do not directly contact
10 * any of the maintainers of this project for assistance;
11 * the project provides a web site, mailing lists and IRC
12 * channels for your use.
14 * This program is free software, distributed under the terms of
15 * the GNU General Public License Version 2. See the LICENSE file
16 * at the top of the source tree.
21 * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
23 * \author Mark Spencer <markster@digium.com>
25 * \ingroup applications
29 <depend>chan_local</depend>
30 <support_level>core</support_level>
36 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
39 #include <sys/signal.h>
41 #include <netinet/in.h>
43 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
44 #include "asterisk/lock.h"
45 #include "asterisk/file.h"
46 #include "asterisk/channel.h"
47 #include "asterisk/pbx.h"
48 #include "asterisk/module.h"
49 #include "asterisk/translate.h"
50 #include "asterisk/say.h"
51 #include "asterisk/config.h"
52 #include "asterisk/features.h"
53 #include "asterisk/musiconhold.h"
54 #include "asterisk/callerid.h"
55 #include "asterisk/utils.h"
56 #include "asterisk/app.h"
57 #include "asterisk/causes.h"
58 #include "asterisk/rtp_engine.h"
59 #include "asterisk/cdr.h"
60 #include "asterisk/manager.h"
61 #include "asterisk/privacy.h"
62 #include "asterisk/stringfields.h"
63 #include "asterisk/global_datastores.h"
64 #include "asterisk/dsp.h"
65 #include "asterisk/cel.h"
66 #include "asterisk/aoc.h"
67 #include "asterisk/ccss.h"
68 #include "asterisk/indications.h"
69 #include "asterisk/framehook.h"
72 <application name="Dial" language="en_US">
74 Attempt to connect to another device or endpoint and bridge the call.
77 <parameter name="Technology/Resource" required="true" argsep="&">
78 <argument name="Technology/Resource" required="true">
79 <para>Specification of the device(s) to dial. These must be in the format of
80 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
81 represents a particular channel driver, and <replaceable>Resource</replaceable>
82 represents a resource available to that particular channel driver.</para>
84 <argument name="Technology2/Resource2" required="false" multiple="true">
85 <para>Optional extra devices to dial in parallel</para>
86 <para>If you need more then one enter them as
87 Technology2/Resource2&Technology3/Resourse3&.....</para>
90 <parameter name="timeout" required="false">
91 <para>Specifies the number of seconds we attempt to dial the specified devices</para>
92 <para>If not specified, this defaults to 136 years.</para>
94 <parameter name="options" required="false">
97 <argument name="x" required="true">
98 <para>The file to play to the called party</para>
100 <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
103 <para>Immediately answer the calling channel when the called channel answers in
104 all cases. Normally, the calling channel is answered when the called channel
105 answers, but when options such as A() and M() are used, the calling channel is
106 not answered until all actions on the called channel (such as playing an
107 announcement) are completed. This option can be used to answer the calling
108 channel before doing anything on the called channel. You will rarely need to use
109 this option, the default behavior is adequate in most cases.</para>
111 <option name="b" argsep="^">
112 <para>Before initiating an outgoing call, Gosub to the specified
113 location using the newly created channel. The Gosub will be
114 executed for each destination channel.</para>
115 <argument name="context" required="false" />
116 <argument name="exten" required="false" />
117 <argument name="priority" required="true" hasparams="optional" argsep="^">
118 <argument name="arg1" multiple="true" required="true" />
119 <argument name="argN" />
122 <option name="B" argsep="^">
123 <para>Before initiating the outgoing call(s), Gosub to the specified
124 location using the current channel.</para>
125 <argument name="context" required="false" />
126 <argument name="exten" required="false" />
127 <argument name="priority" required="true" hasparams="optional" argsep="^">
128 <argument name="arg1" multiple="true" required="true" />
129 <argument name="argN" />
133 <para>Reset the call detail record (CDR) for this call.</para>
136 <para>If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere'</para>
139 <para>Allow the calling user to dial a 1 digit extension while waiting for
140 a call to be answered. Exit to that extension if it exists in the
141 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
144 <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
145 connected. If you wish to use this option with these phones, you
146 can use the <literal>Answer</literal> application before dialing.</para>
149 <option name="D" argsep=":">
150 <argument name="called" />
151 <argument name="calling" />
152 <argument name="progress" />
153 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
154 party has answered, but before the call gets bridged. The
155 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
156 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
157 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
158 to the called party immediately after receiving a PROGRESS message.</para>
159 <para>See SendDTMF for valid digits.</para>
162 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
165 <argument name="x" required="false" />
166 <para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
167 deflection to the dialplan extension of this Dial() using a dialplan <literal>hint</literal>.
168 For example, some PSTNs do not allow CallerID to be set to anything
169 other than the numbers assigned to you.
170 If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
172 <option name="F" argsep="^">
173 <argument name="context" required="false" />
174 <argument name="exten" required="false" />
175 <argument name="priority" required="true" />
176 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
177 to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
179 <para>Any channel variables you want the called channel to inherit from the caller channel must be
180 prefixed with one or two underbars ('_').</para>
184 <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
185 and <emphasis>start</emphasis> execution at that location.</para>
187 <para>Any channel variables you want the called channel to inherit from the caller channel must be
188 prefixed with one or two underbars ('_').</para>
191 <para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
195 <para>Proceed with dialplan execution at the next priority in the current extension if the
196 destination channel hangs up.</para>
198 <option name="G" argsep="^">
199 <argument name="context" required="false" />
200 <argument name="exten" required="false" />
201 <argument name="priority" required="true" />
202 <para>If the call is answered, transfer the calling party to
203 the specified <replaceable>priority</replaceable> and the called party to the specified
204 <replaceable>priority</replaceable> plus one.</para>
206 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
210 <para>Allow the called party to hang up by sending the DTMF sequence
211 defined for disconnect in <filename>features.conf</filename>.</para>
214 <para>Allow the calling party to hang up by sending the DTMF sequence
215 defined for disconnect in <filename>features.conf</filename>.</para>
217 <para>Many SIP and ISDN phones cannot send DTMF digits until the call is
218 connected. If you wish to allow DTMF disconnect before the dialed
219 party answers with these phones, you can use the <literal>Answer</literal>
220 application before dialing.</para>
224 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
227 <para>Asterisk will ignore any connected line update requests or any redirecting party
228 update requests it may receive on this dial attempt.</para>
231 <para>Allow the called party to enable parking of the call by sending
232 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
235 <para>Allow the calling party to enable parking of the call by sending
236 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
238 <option name="L" argsep=":">
239 <argument name="x" required="true">
240 <para>Maximum call time, in milliseconds</para>
243 <para>Warning time, in milliseconds</para>
246 <para>Repeat time, in milliseconds</para>
248 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
249 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
250 <para>This option is affected by the following variables:</para>
252 <variable name="LIMIT_PLAYAUDIO_CALLER">
253 <value name="yes" default="true" />
255 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
257 <variable name="LIMIT_PLAYAUDIO_CALLEE">
259 <value name="no" default="true"/>
260 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
262 <variable name="LIMIT_TIMEOUT_FILE">
263 <value name="filename"/>
264 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
265 If not set, the time remaining will be announced.</para>
267 <variable name="LIMIT_CONNECT_FILE">
268 <value name="filename"/>
269 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
270 If not set, the time remaining will be announced.</para>
272 <variable name="LIMIT_WARNING_FILE">
273 <value name="filename"/>
274 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
275 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
280 <argument name="class" required="false"/>
281 <para>Provide hold music to the calling party until a requested
282 channel answers. A specific music on hold <replaceable>class</replaceable>
283 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
285 <option name="M" argsep="^">
286 <argument name="macro" required="true">
287 <para>Name of the macro that should be executed.</para>
289 <argument name="arg" multiple="true">
290 <para>Macro arguments</para>
292 <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
293 before connecting to the calling channel. Arguments can be specified to the Macro
294 using <literal>^</literal> as a delimiter. The macro can set the variable
295 <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
296 finished executing:</para>
298 <variable name="MACRO_RESULT">
299 <para>If set, this action will be taken after the macro finished executing.</para>
301 Hangup both legs of the call
303 <value name="CONGESTION">
304 Behave as if line congestion was encountered
307 Behave as if a busy signal was encountered
309 <value name="CONTINUE">
310 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
312 <value name="GOTO:[[<context>^]<exten>^]<priority>">
313 Transfer the call to the specified destination.
318 <para>You cannot use any additional action post answer options in conjunction
319 with this option. Also, pbx services are run on the peer (called) channel,
320 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
322 <warning><para>Be aware of the limitations that macros have, specifically with regards to use of
323 the <literal>WaitExten</literal> application. For more information, see the documentation for
324 Macro()</para></warning>
327 <argument name="delete">
328 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
329 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
331 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
332 always be deleted.</para>
334 <para>This option is a modifier for the call screening/privacy mode. (See the
335 <literal>p</literal> and <literal>P</literal> options.) It specifies
336 that no introductions are to be saved in the <directory>priv-callerintros</directory>
340 <para>This option is a modifier for the call screening/privacy mode. It specifies
341 that if Caller*ID is present, do not screen the call.</para>
344 <argument name="x" required="false" />
345 <para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
346 <emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
347 This was the behavior of Asterisk 1.0 and earlier.
348 If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
349 Note that o(${CALLERID(all)}) is similar to option o without the parameter.</para>
352 <argument name="mode">
353 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
354 the originator hanging up will cause the phone to ring back immediately.</para>
355 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
356 flashes the trunk, it will ring their phone back.</para>
358 <para>Enables <emphasis>operator services</emphasis> mode. This option only
359 works when bridging a DAHDI channel to another DAHDI channel
360 only. if specified on non-DAHDI interfaces, it will be ignored.
361 When the destination answers (presumably an operator services
362 station), the originator no longer has control of their line.
363 They may hang up, but the switch will not release their line
364 until the destination party (the operator) hangs up.</para>
367 <para>This option enables screening mode. This is basically Privacy mode
368 without memory.</para>
371 <argument name="x" />
372 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
373 it is provided. The current extension is used if a database family/key is not specified.</para>
376 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
377 party until the called channel has answered.</para>
378 <argument name="tone" required="false">
379 <para>Indicate progress to calling party. Send audio 'tone' from indications.conf</para>
383 <argument name="x" required="true" />
384 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
385 answered the call.</para>
388 <argument name="x" required="true" />
389 <para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable>.</para>
390 <para>Works with the f option.</para>
393 <para>Allow the called party to transfer the calling party by sending the
394 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
395 transfers initiated by other methods.</para>
398 <para>Allow the calling party to transfer the called 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>
402 <option name="U" argsep="^">
403 <argument name="x" required="true">
404 <para>Name of the subroutine to execute via Gosub</para>
406 <argument name="arg" multiple="true" required="false">
407 <para>Arguments for the Gosub routine</para>
409 <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
410 to the calling channel. Arguments can be specified to the Gosub
411 using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
412 <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
414 <variable name="GOSUB_RESULT">
416 Hangup both legs of the call.
418 <value name="CONGESTION">
419 Behave as if line congestion was encountered.
422 Behave as if a busy signal was encountered.
424 <value name="CONTINUE">
425 Hangup the called party and allow the calling party
426 to continue dialplan execution at the next priority.
428 <value name="GOTO:[[<context>^]<exten>^]<priority>">
429 Transfer the call to the specified destination.
434 <para>You cannot use any additional action post answer options in conjunction
435 with this option. Also, pbx services are run on the peer (called) channel,
436 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
440 <argument name = "x" required="true">
441 <para>Force the outgoing callerid presentation indicator parameter to be set
442 to one of the values passed in <replaceable>x</replaceable>:
443 <literal>allowed_not_screened</literal>
444 <literal>allowed_passed_screen</literal>
445 <literal>allowed_failed_screen</literal>
446 <literal>allowed</literal>
447 <literal>prohib_not_screened</literal>
448 <literal>prohib_passed_screen</literal>
449 <literal>prohib_failed_screen</literal>
450 <literal>prohib</literal>
451 <literal>unavailable</literal></para>
453 <para>Works with the f option.</para>
456 <para>Allow the called party to enable recording of the call by sending
457 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
460 <para>Allow the calling party to enable recording of the call by sending
461 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
464 <para>Allow the called party to enable recording of the call by sending
465 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
468 <para>Allow the calling party to enable recording of the call by sending
469 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
472 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
476 <parameter name="URL">
477 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
481 <para>This application will place calls to one or more specified channels. As soon
482 as one of the requested channels answers, the originating channel will be
483 answered, if it has not already been answered. These two channels will then
484 be active in a bridged call. All other channels that were requested will then
487 <para>Unless there is a timeout specified, the Dial application will wait
488 indefinitely until one of the called channels answers, the user hangs up, or
489 if all of the called channels are busy or unavailable. Dialplan executing will
490 continue if no requested channels can be called, or if the timeout expires.
491 This application will report normal termination if the originating channel
492 hangs up, or if the call is bridged and either of the parties in the bridge
493 ends the call.</para>
494 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
495 application will be put into that group (as in Set(GROUP()=...).
496 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
497 application will be put into that group (as in Set(GROUP()=...). Unlike <variable>OUTBOUND_GROUP</variable>,
498 however, the variable will be unset after use.</para>
500 <para>This application sets the following channel variables:</para>
502 <variable name="DIALEDTIME">
503 <para>This is the time from dialing a channel until when it is disconnected.</para>
505 <variable name="ANSWEREDTIME">
506 <para>This is the amount of time for actual call.</para>
508 <variable name="DIALSTATUS">
509 <para>This is the status of the call</para>
510 <value name="CHANUNAVAIL" />
511 <value name="CONGESTION" />
512 <value name="NOANSWER" />
513 <value name="BUSY" />
514 <value name="ANSWER" />
515 <value name="CANCEL" />
516 <value name="DONTCALL">
517 For the Privacy and Screening Modes.
518 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
520 <value name="TORTURE">
521 For the Privacy and Screening Modes.
522 Will be set if the called party chooses to send the calling party to the 'torture' script.
524 <value name="INVALIDARGS" />
529 <application name="RetryDial" language="en_US">
531 Place a call, retrying on failure allowing an optional exit extension.
534 <parameter name="announce" required="true">
535 <para>Filename of sound that will be played when no channel can be reached</para>
537 <parameter name="sleep" required="true">
538 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
540 <parameter name="retries" required="true">
541 <para>Number of retries</para>
542 <para>When this is reached flow will continue at the next priority in the dialplan</para>
544 <parameter name="dialargs" required="true">
545 <para>Same format as arguments provided to the Dial application</para>
549 <para>This application will attempt to place a call using the normal Dial application.
550 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
551 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
552 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
553 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
554 While waiting to retry a call, a 1 digit extension may be dialed. If that
555 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
556 one, The call will jump to that extension immediately.
557 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
558 to the Dial application.</para>
563 static const char app[] = "Dial";
564 static const char rapp[] = "RetryDial";
567 OPT_ANNOUNCE = (1 << 0),
568 OPT_RESETCDR = (1 << 1),
569 OPT_DTMF_EXIT = (1 << 2),
570 OPT_SENDDTMF = (1 << 3),
571 OPT_FORCECLID = (1 << 4),
572 OPT_GO_ON = (1 << 5),
573 OPT_CALLEE_HANGUP = (1 << 6),
574 OPT_CALLER_HANGUP = (1 << 7),
575 OPT_ORIGINAL_CLID = (1 << 8),
576 OPT_DURATION_LIMIT = (1 << 9),
577 OPT_MUSICBACK = (1 << 10),
578 OPT_CALLEE_MACRO = (1 << 11),
579 OPT_SCREEN_NOINTRO = (1 << 12),
580 OPT_SCREEN_NOCALLERID = (1 << 13),
581 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
582 OPT_SCREENING = (1 << 15),
583 OPT_PRIVACY = (1 << 16),
584 OPT_RINGBACK = (1 << 17),
585 OPT_DURATION_STOP = (1 << 18),
586 OPT_CALLEE_TRANSFER = (1 << 19),
587 OPT_CALLER_TRANSFER = (1 << 20),
588 OPT_CALLEE_MONITOR = (1 << 21),
589 OPT_CALLER_MONITOR = (1 << 22),
590 OPT_GOTO = (1 << 23),
591 OPT_OPERMODE = (1 << 24),
592 OPT_CALLEE_PARK = (1 << 25),
593 OPT_CALLER_PARK = (1 << 26),
594 OPT_IGNORE_FORWARDING = (1 << 27),
595 OPT_CALLEE_GOSUB = (1 << 28),
596 OPT_CALLEE_MIXMONITOR = (1 << 29),
597 OPT_CALLER_MIXMONITOR = (1 << 30),
600 /* flags are now 64 bits, so keep it up! */
601 #define DIAL_STILLGOING (1LLU << 31)
602 #define DIAL_NOFORWARDHTML (1LLU << 32)
603 #define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */
604 #define OPT_CANCEL_ELSEWHERE (1LLU << 34)
605 #define OPT_PEER_H (1LLU << 35)
606 #define OPT_CALLEE_GO_ON (1LLU << 36)
607 #define OPT_CANCEL_TIMEOUT (1LLU << 37)
608 #define OPT_FORCE_CID_TAG (1LLU << 38)
609 #define OPT_FORCE_CID_PRES (1LLU << 39)
610 #define OPT_CALLER_ANSWER (1LLU << 40)
611 #define OPT_PREDIAL_CALLEE (1LLU << 41)
612 #define OPT_PREDIAL_CALLER (1LLU << 42)
615 OPT_ARG_ANNOUNCE = 0,
618 OPT_ARG_DURATION_LIMIT,
620 OPT_ARG_CALLEE_MACRO,
622 OPT_ARG_CALLEE_GOSUB,
623 OPT_ARG_CALLEE_GO_ON,
625 OPT_ARG_DURATION_STOP,
627 OPT_ARG_SCREEN_NOINTRO,
628 OPT_ARG_ORIGINAL_CLID,
630 OPT_ARG_FORCE_CID_TAG,
631 OPT_ARG_FORCE_CID_PRES,
632 OPT_ARG_PREDIAL_CALLEE,
633 OPT_ARG_PREDIAL_CALLER,
634 /* note: this entry _MUST_ be the last one in the enum */
638 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
639 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
640 AST_APP_OPTION('a', OPT_CALLER_ANSWER),
641 AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
642 AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
643 AST_APP_OPTION('C', OPT_RESETCDR),
644 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
645 AST_APP_OPTION('d', OPT_DTMF_EXIT),
646 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
647 AST_APP_OPTION('e', OPT_PEER_H),
648 AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
649 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
650 AST_APP_OPTION('g', OPT_GO_ON),
651 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
652 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
653 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
654 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
655 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
656 AST_APP_OPTION('k', OPT_CALLEE_PARK),
657 AST_APP_OPTION('K', OPT_CALLER_PARK),
658 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
659 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
660 AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
661 AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
662 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
663 AST_APP_OPTION_ARG('o', OPT_ORIGINAL_CLID, OPT_ARG_ORIGINAL_CLID),
664 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
665 AST_APP_OPTION('p', OPT_SCREENING),
666 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
667 AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
668 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
669 AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
670 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
671 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
672 AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
673 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
674 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
675 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
676 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
677 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
678 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
681 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
682 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
683 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
684 OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
685 !ast_channel_audiohooks(chan) && !ast_channel_audiohooks(peer) && \
686 ast_framehook_list_is_empty(ast_channel_framehooks(chan)) && ast_framehook_list_is_empty(ast_channel_framehooks(peer)))
689 * The list of active channels
692 AST_LIST_ENTRY(chanlist) node;
693 struct ast_channel *chan;
694 /*! Channel interface dialing string (is tech/number). (Stored in stuff[]) */
695 const char *interface;
696 /*! Channel technology name. (Stored in stuff[]) */
698 /*! Channel device addressing. (Stored in stuff[]) */
701 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
702 struct ast_party_connected_line connected;
703 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
704 unsigned int pending_connected_update:1;
705 struct ast_aoc_decoded *aoc_s_rate_list;
706 /*! The interface, tech, and number strings are stuffed here. */
710 AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
712 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
714 static void chanlist_free(struct chanlist *outgoing)
716 ast_party_connected_line_free(&outgoing->connected);
717 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
721 static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int answered_elsewhere)
723 /* Hang up a tree of stuff */
724 struct chanlist *outgoing;
726 while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
727 /* Hangup any existing lines we have open */
728 if (outgoing->chan && (outgoing->chan != exception)) {
729 if (answered_elsewhere) {
730 /* This is for the channel drivers */
731 ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
733 ast_hangup(outgoing->chan);
735 chanlist_free(outgoing);
739 #define AST_MAX_WATCHERS 256
742 * argument to handle_cause() and other functions.
745 struct ast_channel *chan;
751 static void handle_cause(int cause, struct cause_args *num)
753 struct ast_cdr *cdr = ast_channel_cdr(num->chan);
762 case AST_CAUSE_CONGESTION:
768 case AST_CAUSE_NO_ROUTE_DESTINATION:
769 case AST_CAUSE_UNREGISTERED:
775 case AST_CAUSE_NO_ANSWER:
777 ast_cdr_noanswer(cdr);
780 case AST_CAUSE_NORMAL_CLEARING:
789 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
791 char rexten[2] = { exten, '\0' };
794 if (!ast_goto_if_exists(chan, context, rexten, pri))
797 if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
799 else if (!ast_strlen_zero(ast_channel_macrocontext(chan))) {
800 if (!ast_goto_if_exists(chan, ast_channel_macrocontext(chan), rexten, pri))
807 /* do not call with chan lock held */
808 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
813 ast_channel_lock(chan);
814 context = ast_strdupa(S_OR(ast_channel_macrocontext(chan), ast_channel_context(chan)));
815 exten = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
816 ast_channel_unlock(chan);
818 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
821 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
823 struct ast_channel *chans[] = { src, dst };
825 <managerEventInstance>
826 <synopsis>Raised when a dial action has started.</synopsis>
828 <parameter name="SubEvent">
829 <para>A sub event type, specifying whether the dial action has begun or ended.</para>
836 </managerEventInstance>
838 ast_manager_event_multichan(EVENT_FLAG_CALL, "Dial", 2, chans,
839 "SubEvent: Begin\r\n"
841 "Destination: %s\r\n"
842 "CallerIDNum: %s\r\n"
843 "CallerIDName: %s\r\n"
844 "ConnectedLineNum: %s\r\n"
845 "ConnectedLineName: %s\r\n"
847 "DestUniqueID: %s\r\n"
848 "Dialstring: %s\r\n",
849 ast_channel_name(src), ast_channel_name(dst),
850 S_COR(ast_channel_caller(src)->id.number.valid, ast_channel_caller(src)->id.number.str, "<unknown>"),
851 S_COR(ast_channel_caller(src)->id.name.valid, ast_channel_caller(src)->id.name.str, "<unknown>"),
852 S_COR(ast_channel_connected(src)->id.number.valid, ast_channel_connected(src)->id.number.str, "<unknown>"),
853 S_COR(ast_channel_connected(src)->id.name.valid, ast_channel_connected(src)->id.name.str, "<unknown>"),
854 ast_channel_uniqueid(src), ast_channel_uniqueid(dst),
855 dialstring ? dialstring : "");
858 static void senddialendevent(struct ast_channel *src, const char *dialstatus)
861 <managerEventInstance>
862 <synopsis>Raised when a dial action has ended.</synopsis>
864 <parameter name="DialStatus">
865 <para>The value of the <variable>DIALSTATUS</variable> channel variable.</para>
868 </managerEventInstance>
870 ast_manager_event(src, EVENT_FLAG_CALL, "Dial",
874 "DialStatus: %s\r\n",
875 ast_channel_name(src), ast_channel_uniqueid(src), dialstatus);
879 * helper function for wait_for_answer()
881 * \param o Outgoing call channel list.
882 * \param num Incoming call channel cause accumulation
883 * \param peerflags Dial option flags
884 * \param single TRUE if there is only one outgoing call.
885 * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
886 * \param to Remaining call timeout time.
887 * \param forced_clid OPT_FORCECLID caller id to send
888 * \param stored_clid Caller id representing the called party if needed
890 * XXX this code is highly suspicious, as it essentially overwrites
891 * the outgoing channel without properly deleting it.
893 * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
895 static void do_forward(struct chanlist *o, struct cause_args *num,
896 struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
897 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
900 struct ast_channel *original = o->chan;
901 struct ast_channel *c = o->chan; /* the winner */
902 struct ast_channel *in = num->chan; /* the input channel */
906 struct ast_party_caller caller;
908 ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
909 if ((stuff = strchr(tmpchan, '/'))) {
913 const char *forward_context;
915 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
916 if (ast_strlen_zero(forward_context)) {
917 forward_context = NULL;
919 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
920 ast_channel_unlock(c);
924 if (!strcasecmp(tech, "Local")) {
926 * Drop the connected line update block for local channels since
927 * this is going to run dialplan and the user can change his
928 * mind about what connected line information he wants to send.
930 ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
933 ast_cel_report_event(in, AST_CEL_FORWARD, NULL, ast_channel_call_forward(c), NULL);
935 /* Before processing channel, go ahead and check for forwarding */
936 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
937 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
938 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
939 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
941 cause = AST_CAUSE_BUSY;
943 /* Setup parameters */
944 c = o->chan = ast_request(tech, ast_channel_nativeformats(in), in, stuff, &cause);
946 if (single && !caller_entertained) {
947 ast_channel_make_compatible(o->chan, in);
949 ast_channel_lock_both(in, o->chan);
950 ast_channel_inherit_variables(in, o->chan);
951 ast_channel_datastore_inherit(in, o->chan);
952 ast_channel_unlock(in);
953 ast_channel_unlock(o->chan);
954 /* When a call is forwarded, we don't want to track new interfaces
955 * dialed for CC purposes. Setting the done flag will ensure that
956 * any Dial operations that happen later won't record CC interfaces.
958 ast_ignore_cc(o->chan);
959 ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
962 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
966 ast_clear_flag64(o, DIAL_STILLGOING);
967 handle_cause(cause, num);
968 ast_hangup(original);
970 ast_channel_lock_both(c, original);
971 ast_party_redirecting_copy(ast_channel_redirecting(c),
972 ast_channel_redirecting(original));
973 ast_channel_unlock(c);
974 ast_channel_unlock(original);
976 ast_channel_lock_both(c, in);
978 if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
979 ast_rtp_instance_early_bridge_make_compatible(c, in);
982 if (!ast_channel_redirecting(c)->from.number.valid
983 || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
985 * The call was not previously redirected so it is
986 * now redirected from this number.
988 ast_party_number_free(&ast_channel_redirecting(c)->from.number);
989 ast_party_number_init(&ast_channel_redirecting(c)->from.number);
990 ast_channel_redirecting(c)->from.number.valid = 1;
991 ast_channel_redirecting(c)->from.number.str =
992 ast_strdup(S_OR(ast_channel_macroexten(in), ast_channel_exten(in)));
995 ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
997 /* Determine CallerID to store in outgoing channel. */
998 ast_party_caller_set_init(&caller, ast_channel_caller(c));
999 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1000 caller.id = *stored_clid;
1001 ast_channel_set_caller_event(c, &caller, NULL);
1002 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1003 } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
1004 ast_channel_caller(c)->id.number.str, NULL))) {
1006 * The new channel has no preset CallerID number by the channel
1007 * driver. Use the dialplan extension and hint name.
1009 caller.id = *stored_clid;
1010 ast_channel_set_caller_event(c, &caller, NULL);
1011 ast_set_flag64(o, DIAL_CALLERID_ABSENT);
1013 ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
1016 /* Determine CallerID for outgoing channel to send. */
1017 if (ast_test_flag64(o, OPT_FORCECLID)) {
1018 struct ast_party_connected_line connected;
1020 ast_party_connected_line_init(&connected);
1021 connected.id = *forced_clid;
1022 ast_party_connected_line_copy(ast_channel_connected(c), &connected);
1024 ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
1027 ast_channel_accountcode_set(c, ast_channel_accountcode(in));
1029 ast_channel_appl_set(c, "AppDial");
1030 ast_channel_data_set(c, "(Outgoing Line)");
1032 ast_channel_unlock(in);
1033 if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1034 struct ast_party_redirecting redirecting;
1037 * Redirecting updates to the caller make sense only on single
1040 * We must unlock c before calling
1041 * ast_channel_redirecting_macro, because we put c into
1042 * autoservice there. That is pretty much a guaranteed
1043 * deadlock. This is why the handling of c's lock may seem a
1046 ast_party_redirecting_init(&redirecting);
1047 ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
1048 ast_channel_unlock(c);
1049 if (ast_channel_redirecting_sub(c, in, &redirecting, 0) &&
1050 ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
1051 ast_channel_update_redirecting(in, &redirecting, NULL);
1053 ast_party_redirecting_free(&redirecting);
1055 ast_channel_unlock(c);
1058 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
1062 if (ast_call(c, stuff, 0)) {
1063 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
1065 ast_clear_flag64(o, DIAL_STILLGOING);
1066 ast_hangup(original);
1071 ast_channel_lock_both(c, in);
1072 senddialevent(in, c, stuff);
1073 ast_channel_unlock(in);
1074 ast_channel_unlock(c);
1075 /* Hangup the original channel now, in case we needed it */
1076 ast_hangup(original);
1078 if (single && !caller_entertained) {
1079 ast_indicate(in, -1);
1084 /* argument used for some functions. */
1085 struct privacy_args {
1089 char privintro[1024];
1093 static struct ast_channel *wait_for_answer(struct ast_channel *in,
1094 struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
1096 struct privacy_args *pa,
1097 const struct cause_args *num_in, int *result, char *dtmf_progress,
1098 const int ignore_cc,
1099 struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
1101 struct cause_args num = *num_in;
1102 int prestart = num.busy + num.congestion + num.nochan;
1104 struct ast_channel *peer = NULL;
1106 struct chanlist *epollo;
1108 struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
1109 /* single is set if only one destination is enabled */
1110 int single = outgoing && !AST_LIST_NEXT(outgoing, node);
1111 int caller_entertained = outgoing
1112 && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
1113 struct ast_party_connected_line connected_caller;
1114 struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
1115 int cc_recall_core_id;
1117 int cc_frame_received = 0;
1118 int num_ringing = 0;
1119 struct timeval start = ast_tvnow();
1121 ast_party_connected_line_init(&connected_caller);
1123 /* Turn off hold music, etc */
1124 if (!caller_entertained) {
1125 ast_deactivate_generator(in);
1126 /* If we are calling a single channel, and not providing ringback or music, */
1127 /* then, make them compatible for in-band tone purpose */
1128 if (ast_channel_make_compatible(outgoing->chan, in) < 0) {
1129 /* If these channels can not be made compatible,
1130 * there is no point in continuing. The bridge
1131 * will just fail if it gets that far.
1134 strcpy(pa->status, "CONGESTION");
1135 ast_cdr_failed(ast_channel_cdr(in));
1140 if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
1141 && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
1142 ast_channel_lock(outgoing->chan);
1143 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(outgoing->chan));
1144 ast_channel_unlock(outgoing->chan);
1145 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1146 if (ast_channel_connected_line_sub(outgoing->chan, in, &connected_caller, 0) &&
1147 ast_channel_connected_line_macro(outgoing->chan, in, &connected_caller, 1, 0)) {
1148 ast_channel_update_connected_line(in, &connected_caller, NULL);
1150 ast_party_connected_line_free(&connected_caller);
1154 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1157 AST_LIST_TRAVERSE(out_chans, epollo, node) {
1158 ast_poll_channel_add(in, epollo->chan);
1162 while ((*to = ast_remaining_ms(start, orig)) && !peer) {
1164 int pos = 0; /* how many channels do we handle */
1165 int numlines = prestart;
1166 struct ast_channel *winner;
1167 struct ast_channel *watchers[AST_MAX_WATCHERS];
1169 watchers[pos++] = in;
1170 AST_LIST_TRAVERSE(out_chans, o, node) {
1171 /* Keep track of important channels */
1172 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1173 watchers[pos++] = o->chan;
1176 if (pos == 1) { /* only the input channel is available */
1177 if (numlines == (num.busy + num.congestion + num.nochan)) {
1178 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1180 strcpy(pa->status, "BUSY");
1181 else if (num.congestion)
1182 strcpy(pa->status, "CONGESTION");
1183 else if (num.nochan)
1184 strcpy(pa->status, "CHANUNAVAIL");
1186 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1190 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1194 winner = ast_waitfor_n(watchers, pos, to);
1195 AST_LIST_TRAVERSE(out_chans, o, node) {
1196 struct ast_frame *f;
1197 struct ast_channel *c = o->chan;
1201 if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
1203 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1204 if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1205 if (o->pending_connected_update) {
1206 if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1207 ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1208 ast_channel_update_connected_line(in, &o->connected, NULL);
1210 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1211 ast_channel_lock(c);
1212 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
1213 ast_channel_unlock(c);
1214 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1215 if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
1216 ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
1217 ast_channel_update_connected_line(in, &connected_caller, NULL);
1219 ast_party_connected_line_free(&connected_caller);
1222 if (o->aoc_s_rate_list) {
1223 size_t encoded_size;
1224 struct ast_aoc_encoded *encoded;
1225 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1226 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1227 ast_aoc_destroy_encoded(encoded);
1231 ast_copy_flags64(peerflags, o,
1232 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1233 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1234 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1235 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1236 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1237 DIAL_NOFORWARDHTML);
1238 ast_channel_dialcontext_set(c, "");
1239 ast_channel_exten_set(c, "");
1245 /* here, o->chan == c == winner */
1246 if (!ast_strlen_zero(ast_channel_call_forward(c))) {
1247 pa->sentringing = 0;
1248 if (!ignore_cc && (f = ast_read(c))) {
1249 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1250 /* This channel is forwarding the call, and is capable of CC, so
1251 * be sure to add the new device interface to the list
1253 ast_handle_cc_control_frame(in, c, f->data.ptr);
1258 if (o->pending_connected_update) {
1260 * Re-seed the chanlist's connected line information with
1261 * previously acquired connected line info from the incoming
1262 * channel. The previously acquired connected line info could
1263 * have been set through the CONNECTED_LINE dialplan function.
1265 o->pending_connected_update = 0;
1266 ast_channel_lock(in);
1267 ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
1268 ast_channel_unlock(in);
1271 do_forward(o, &num, peerflags, single, caller_entertained, to,
1272 forced_clid, stored_clid);
1274 if (single && o->chan
1275 && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
1276 && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1277 ast_channel_lock(o->chan);
1278 ast_connected_line_copy_from_caller(&connected_caller,
1279 ast_channel_caller(o->chan));
1280 ast_channel_unlock(o->chan);
1281 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1282 if (ast_channel_connected_line_sub(o->chan, in, &connected_caller, 0) &&
1283 ast_channel_connected_line_macro(o->chan, in, &connected_caller, 1, 0)) {
1284 ast_channel_update_connected_line(in, &connected_caller, NULL);
1286 ast_party_connected_line_free(&connected_caller);
1290 f = ast_read(winner);
1292 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1294 ast_poll_channel_del(in, c);
1298 ast_clear_flag64(o, DIAL_STILLGOING);
1299 handle_cause(ast_channel_hangupcause(in), &num);
1302 switch (f->frametype) {
1303 case AST_FRAME_CONTROL:
1304 switch (f->subclass.integer) {
1305 case AST_CONTROL_ANSWER:
1306 /* This is our guy if someone answered. */
1308 ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
1309 if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1310 if (o->pending_connected_update) {
1311 if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
1312 ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1313 ast_channel_update_connected_line(in, &o->connected, NULL);
1315 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1316 ast_channel_lock(c);
1317 ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
1318 ast_channel_unlock(c);
1319 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1320 if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
1321 ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
1322 ast_channel_update_connected_line(in, &connected_caller, NULL);
1324 ast_party_connected_line_free(&connected_caller);
1327 if (o->aoc_s_rate_list) {
1328 size_t encoded_size;
1329 struct ast_aoc_encoded *encoded;
1330 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1331 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1332 ast_aoc_destroy_encoded(encoded);
1336 if (ast_channel_cdr(peer)) {
1337 ast_channel_cdr(peer)->answer = ast_tvnow();
1338 ast_channel_cdr(peer)->disposition = AST_CDR_ANSWERED;
1340 ast_copy_flags64(peerflags, o,
1341 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1342 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1343 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1344 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1345 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1346 DIAL_NOFORWARDHTML);
1347 ast_channel_dialcontext_set(c, "");
1348 ast_channel_exten_set(c, "");
1349 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
1350 /* Setup early bridge if appropriate */
1351 ast_channel_early_bridge(in, peer);
1353 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1354 ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
1355 ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
1357 case AST_CONTROL_BUSY:
1358 ast_verb(3, "%s is busy\n", ast_channel_name(c));
1359 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1362 ast_clear_flag64(o, DIAL_STILLGOING);
1363 handle_cause(AST_CAUSE_BUSY, &num);
1365 case AST_CONTROL_CONGESTION:
1366 ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
1367 ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
1370 ast_clear_flag64(o, DIAL_STILLGOING);
1371 handle_cause(AST_CAUSE_CONGESTION, &num);
1373 case AST_CONTROL_RINGING:
1374 /* This is a tricky area to get right when using a native
1375 * CC agent. The reason is that we do the best we can to send only a
1376 * single ringing notification to the caller.
1378 * Call completion complicates the logic used here. CCNR is typically
1379 * offered during a ringing message. Let's say that party A calls
1380 * parties B, C, and D. B and C do not support CC requests, but D
1381 * does. If we were to receive a ringing notification from B before
1382 * the others, then we would end up sending a ringing message to
1383 * A with no CCNR offer present.
1385 * The approach that we have taken is that if we receive a ringing
1386 * response from a party and no CCNR offer is present, we need to
1387 * wait. Specifically, we need to wait until either a) a called party
1388 * offers CCNR in its ringing response or b) all called parties have
1389 * responded in some way to our call and none offers CCNR.
1391 * The drawback to this is that if one of the parties has a delayed
1392 * response or, god forbid, one just plain doesn't respond to our
1393 * outgoing call, then this will result in a significant delay between
1394 * when the caller places the call and hears ringback.
1396 * Note also that if CC is disabled for this call, then it is perfectly
1397 * fine for ringing frames to get sent through.
1400 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1401 ast_verb(3, "%s is ringing\n", ast_channel_name(c));
1402 /* Setup early media if appropriate */
1403 if (single && !caller_entertained
1404 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1405 ast_channel_early_bridge(in, c);
1407 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1408 ast_indicate(in, AST_CONTROL_RINGING);
1413 case AST_CONTROL_PROGRESS:
1414 ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1415 /* Setup early media if appropriate */
1416 if (single && !caller_entertained
1417 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1418 ast_channel_early_bridge(in, c);
1420 if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
1421 if (single || (!single && !pa->sentringing)) {
1422 ast_indicate(in, AST_CONTROL_PROGRESS);
1425 if (!ast_strlen_zero(dtmf_progress)) {
1427 "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
1429 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1432 case AST_CONTROL_VIDUPDATE:
1433 case AST_CONTROL_SRCUPDATE:
1434 case AST_CONTROL_SRCCHANGE:
1435 if (!single || caller_entertained) {
1438 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1439 ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
1440 ast_indicate(in, f->subclass.integer);
1442 case AST_CONTROL_CONNECTED_LINE:
1443 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1444 ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
1448 struct ast_party_connected_line connected;
1450 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
1451 ast_channel_name(c), ast_channel_name(in));
1452 ast_party_connected_line_set_init(&connected, &o->connected);
1453 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1454 ast_party_connected_line_set(&o->connected, &connected, NULL);
1455 ast_party_connected_line_free(&connected);
1456 o->pending_connected_update = 1;
1459 if (ast_channel_connected_line_sub(c, in, f, 1) &&
1460 ast_channel_connected_line_macro(c, in, f, 1, 1)) {
1461 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1464 case AST_CONTROL_AOC:
1466 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1467 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1468 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1469 o->aoc_s_rate_list = decoded;
1471 ast_aoc_destroy_decoded(decoded);
1475 case AST_CONTROL_REDIRECTING:
1478 * Redirecting updates to the caller make sense only on single
1483 if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
1484 ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
1487 ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
1488 ast_channel_name(c), ast_channel_name(in));
1489 if (ast_channel_redirecting_sub(c, in, f, 1) &&
1490 ast_channel_redirecting_macro(c, in, f, 1, 1)) {
1491 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1493 pa->sentringing = 0;
1495 case AST_CONTROL_PROCEEDING:
1496 ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
1497 if (single && !caller_entertained
1498 && CAN_EARLY_BRIDGE(peerflags, in, c)) {
1499 ast_channel_early_bridge(in, c);
1501 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1502 ast_indicate(in, AST_CONTROL_PROCEEDING);
1504 case AST_CONTROL_HOLD:
1505 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1506 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
1507 ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1509 case AST_CONTROL_UNHOLD:
1510 /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
1511 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
1512 ast_indicate(in, AST_CONTROL_UNHOLD);
1514 case AST_CONTROL_OFFHOOK:
1515 case AST_CONTROL_FLASH:
1516 /* Ignore going off hook and flash */
1518 case AST_CONTROL_CC:
1520 ast_handle_cc_control_frame(in, c, f->data.ptr);
1521 cc_frame_received = 1;
1524 case AST_CONTROL_PVT_CAUSE_CODE:
1525 ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
1528 if (single && !caller_entertained) {
1529 ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
1530 ast_indicate(in, -1);
1531 pa->sentringing = 0;
1535 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1539 case AST_FRAME_VOICE:
1540 case AST_FRAME_IMAGE:
1541 if (caller_entertained) {
1545 case AST_FRAME_TEXT:
1546 if (single && ast_write(in, f)) {
1547 ast_log(LOG_WARNING, "Unable to write frametype: %d\n",
1551 case AST_FRAME_HTML:
1552 if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
1553 && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1554 ast_log(LOG_WARNING, "Unable to send URL\n");
1563 struct ast_frame *f = ast_read(in);
1565 if (f && (f->frametype != AST_FRAME_VOICE))
1566 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1567 else if (!f || (f->frametype != AST_FRAME_VOICE))
1568 printf("Hangup received on %s\n", in->name);
1570 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1573 strcpy(pa->status, "CANCEL");
1574 ast_cdr_noanswer(ast_channel_cdr(in));
1576 if (f->data.uint32) {
1577 ast_channel_hangupcause_set(in, f->data.uint32);
1582 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1587 /* now f is guaranteed non-NULL */
1588 if (f->frametype == AST_FRAME_DTMF) {
1589 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1590 const char *context;
1591 ast_channel_lock(in);
1592 context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1593 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1594 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1596 ast_cdr_noanswer(ast_channel_cdr(in));
1597 *result = f->subclass.integer;
1598 strcpy(pa->status, "CANCEL");
1600 ast_channel_unlock(in);
1602 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1606 ast_channel_unlock(in);
1609 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1610 detect_disconnect(in, f->subclass.integer, &featurecode)) {
1611 ast_verb(3, "User requested call disconnect.\n");
1613 strcpy(pa->status, "CANCEL");
1614 ast_cdr_noanswer(ast_channel_cdr(in));
1617 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1623 /* Send the frame from the in channel to all outgoing channels. */
1624 AST_LIST_TRAVERSE(out_chans, o, node) {
1625 if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
1626 /* This outgoing channel has died so don't send the frame to it. */
1629 switch (f->frametype) {
1630 case AST_FRAME_HTML:
1631 /* Forward HTML stuff */
1632 if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
1633 && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1634 ast_log(LOG_WARNING, "Unable to send URL\n");
1637 case AST_FRAME_VOICE:
1638 case AST_FRAME_IMAGE:
1639 if (!single || caller_entertained) {
1641 * We are calling multiple parties or caller is being
1642 * entertained and has thus not been made compatible.
1643 * No need to check any other called parties.
1648 case AST_FRAME_TEXT:
1649 case AST_FRAME_DTMF_BEGIN:
1650 case AST_FRAME_DTMF_END:
1651 if (ast_write(o->chan, f)) {
1652 ast_log(LOG_WARNING, "Unable to forward frametype: %d\n",
1656 case AST_FRAME_CONTROL:
1657 switch (f->subclass.integer) {
1658 case AST_CONTROL_HOLD:
1659 ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
1660 ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
1662 case AST_CONTROL_UNHOLD:
1663 ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
1664 ast_indicate(o->chan, AST_CONTROL_UNHOLD);
1666 case AST_CONTROL_VIDUPDATE:
1667 case AST_CONTROL_SRCUPDATE:
1668 case AST_CONTROL_SRCCHANGE:
1669 if (!single || caller_entertained) {
1671 * We are calling multiple parties or caller is being
1672 * entertained and has thus not been made compatible.
1673 * No need to check any other called parties.
1677 ast_verb(3, "%s requested media update control %d, passing it to %s\n",
1678 ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
1679 ast_indicate(o->chan, f->subclass.integer);
1681 case AST_CONTROL_CONNECTED_LINE:
1682 if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
1683 ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
1684 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1687 case AST_CONTROL_REDIRECTING:
1688 if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
1689 ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
1690 ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
1694 /* We are not going to do anything with this frame. */
1699 /* We are not going to do anything with this frame. */
1709 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1711 if (!*to || ast_check_hangup(in)) {
1712 ast_cdr_noanswer(ast_channel_cdr(in));
1716 AST_LIST_TRAVERSE(out_chans, epollo, node) {
1718 ast_poll_channel_del(in, epollo->chan);
1723 ast_cc_completed(in, "Recall completed!");
1728 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
1730 struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
1731 struct ast_call_feature feature = { 0, };
1734 ast_str_append(featurecode, 1, "%c", code);
1736 res = ast_feature_detect(chan, &features, ast_str_buffer(*featurecode), &feature);
1738 if (res != AST_FEATURE_RETURN_STOREDIGITS) {
1739 ast_str_reset(*featurecode);
1741 if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
1748 /* returns true if there is a valid privacy reply */
1749 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1753 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1755 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1760 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1761 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1767 /* Get the user's intro, store it in priv-callerintros/$CID,
1768 unless it is already there-- this should be done before the
1769 call is actually dialed */
1771 /* all ring indications and moh for the caller has been halted as soon as the
1772 target extension was picked up. We are going to have to kill some
1773 time and make the caller believe the peer hasn't picked up yet */
1775 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1776 char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
1777 ast_indicate(chan, -1);
1778 ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
1779 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1780 ast_channel_musicclass_set(chan, original_moh);
1781 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1782 ast_indicate(chan, AST_CONTROL_RINGING);
1786 /* Start autoservice on the other chan ?? */
1787 res2 = ast_autoservice_start(chan);
1788 /* Now Stream the File */
1789 for (loopcount = 0; loopcount < 3; loopcount++) {
1790 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1792 if (!res2) /* on timeout, play the message again */
1793 res2 = ast_play_and_wait(peer, "priv-callpending");
1794 if (!valid_priv_reply(opts, res2))
1796 /* priv-callpending script:
1797 "I have a caller waiting, who introduces themselves as:"
1800 res2 = ast_play_and_wait(peer, pa->privintro);
1801 if (!valid_priv_reply(opts, res2))
1803 /* now get input from the called party, as to their choice */
1805 /* XXX can we have both, or they are mutually exclusive ? */
1806 if (ast_test_flag64(opts, OPT_PRIVACY))
1807 res2 = ast_play_and_wait(peer, "priv-callee-options");
1808 if (ast_test_flag64(opts, OPT_SCREENING))
1809 res2 = ast_play_and_wait(peer, "screen-callee-options");
1812 /*! \page DialPrivacy Dial Privacy scripts
1813 * \par priv-callee-options script:
1814 * \li Dial 1 if you wish this caller to reach you directly in the future,
1815 * and immediately connect to their incoming call.
1816 * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
1817 * \li Dial 3 to send this caller to the torture menus, now and forevermore.
1818 * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1819 * \li Dial 5 to allow this caller to come straight thru to you in the future,
1820 * but right now, just this once, send them to voicemail.
1822 * \par screen-callee-options script:
1823 * \li Dial 1 if you wish to immediately connect to the incoming call
1824 * \li Dial 2 if you wish to send this caller to voicemail.
1825 * \li Dial 3 to send this caller to the torture menus.
1826 * \li Dial 4 to send this caller to a simple "go away" menu.
1828 if (valid_priv_reply(opts, res2))
1830 /* invalid option */
1831 res2 = ast_play_and_wait(peer, "vm-sorry");
1834 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1836 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1837 ast_indicate(chan, -1);
1838 pa->sentringing = 0;
1840 ast_autoservice_stop(chan);
1841 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1842 /* map keypresses to various things, the index is res2 - '1' */
1843 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1844 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1846 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1847 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1848 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1854 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1857 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1860 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1863 /* XXX should we set status to DENY ? */
1864 if (ast_test_flag64(opts, OPT_PRIVACY))
1866 /* if not privacy, then 5 is the same as "default" case */
1867 default: /* bad input or -1 if failure to start autoservice */
1868 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
1869 /* well, there seems basically two choices. Just patch the caller thru immediately,
1870 or,... put 'em thru to voicemail. */
1871 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1872 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1873 /* XXX should we set status to DENY ? */
1874 /* XXX what about the privacy flags ? */
1878 if (res2 == '1') { /* the only case where we actually connect */
1879 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1880 just clog things up, and it's not useful information, not being tied to a CID */
1881 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1882 ast_filedelete(pa->privintro, NULL);
1883 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1884 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1886 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1888 return 0; /* the good exit path */
1890 /* hang up on the callee -- he didn't want to talk anyway! */
1891 ast_autoservice_chan_hangup_peer(chan, peer);
1896 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1897 static int setup_privacy_args(struct privacy_args *pa,
1898 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1904 if (ast_channel_caller(chan)->id.number.valid
1905 && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
1906 l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
1907 ast_shrink_phone_number(l);
1908 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1909 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1910 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1912 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1913 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1918 tnam = ast_strdupa(ast_channel_name(chan));
1919 /* clean the channel name so slashes don't try to end up in disk file name */
1920 for (tn2 = tnam; *tn2; tn2++) {
1921 if (*tn2 == '/') /* any other chars to be afraid of? */
1924 ast_verb(3, "Privacy-- callerid is empty\n");
1926 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
1928 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1931 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1933 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1934 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1935 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1936 pa->privdb_val = AST_PRIVACY_ALLOW;
1937 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1938 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1941 if (pa->privdb_val == AST_PRIVACY_DENY) {
1942 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1943 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1945 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1946 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1947 return 0; /* Is this right? */
1948 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1949 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1950 return 0; /* is this right??? */
1951 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1952 /* Get the user's intro, store it in priv-callerintros/$CID,
1953 unless it is already there-- this should be done before the
1954 call is actually dialed */
1956 /* make sure the priv-callerintros dir actually exists */
1957 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1958 if ((res = ast_mkdir(pa->privintro, 0755))) {
1959 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1963 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1964 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1965 /* the DELUX version of this code would allow this caller the
1966 option to hear and retape their previously recorded intro.
1969 int duration; /* for feedback from play_and_wait */
1970 /* the file doesn't exist yet. Let the caller submit his
1971 vocal intro for posterity */
1972 /* priv-recordintro script:
1974 "At the tone, please say your name:"
1977 int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1979 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 */
1980 /* don't think we'll need a lock removed, we took care of
1981 conflicts by naming the pa.privintro file */
1983 /* Delete the file regardless since they hung up during recording */
1984 ast_filedelete(pa->privintro, NULL);
1985 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1986 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1988 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1991 if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
1992 ast_waitstream(chan, "");
1995 return 1; /* success */
1998 static void end_bridge_callback(void *data)
2002 struct ast_channel *chan = data;
2004 if (!ast_channel_cdr(chan)) {
2010 ast_channel_lock(chan);
2011 if (ast_channel_cdr(chan)->answer.tv_sec) {
2012 snprintf(buf, sizeof(buf), "%ld", (long) end - ast_channel_cdr(chan)->answer.tv_sec);
2013 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
2016 if (ast_channel_cdr(chan)->start.tv_sec) {
2017 snprintf(buf, sizeof(buf), "%ld", (long) end - ast_channel_cdr(chan)->start.tv_sec);
2018 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
2020 ast_channel_unlock(chan);
2023 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
2024 bconfig->end_bridge_callback_data = originator;
2027 static int dial_handle_playtones(struct ast_channel *chan, const char *data)
2029 struct ast_tone_zone_sound *ts = NULL;
2031 const char *str = data;
2033 if (ast_strlen_zero(str)) {
2034 ast_debug(1,"Nothing to play\n");
2038 ts = ast_get_indication_tone(ast_channel_zone(chan), str);
2040 if (ts && ts->data[0]) {
2041 res = ast_playtones_start(chan, 0, ts->data, 0);
2047 ts = ast_tone_zone_sound_unref(ts);
2051 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
2057 static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
2059 int res = -1; /* default: error */
2060 char *rest, *cur; /* scan the list of destinations */
2061 struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
2062 struct chanlist *outgoing;
2063 struct chanlist *tmp;
2064 struct ast_channel *peer;
2065 int to; /* timeout */
2066 struct cause_args num = { chan, 0, 0, 0 };
2069 struct ast_bridge_config config = { { 0, } };
2070 struct timeval calldurationlimit = { 0, };
2071 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
2072 struct privacy_args pa = {
2075 .status = "INVALIDARGS",
2077 int sentringing = 0, moh = 0;
2078 const char *outbound_group = NULL;
2082 int delprivintro = 0;
2083 AST_DECLARE_APP_ARGS(args,
2085 AST_APP_ARG(timeout);
2086 AST_APP_ARG(options);
2089 struct ast_flags64 opts = { 0, };
2090 char *opt_args[OPT_ARG_ARRAY_SIZE];
2091 struct ast_datastore *datastore = NULL;
2092 int fulldial = 0, num_dialed = 0;
2094 char device_name[AST_CHANNEL_NAME];
2095 char forced_clid_name[AST_MAX_EXTENSION];
2096 char stored_clid_name[AST_MAX_EXTENSION];
2097 int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
2099 * \brief Forced CallerID party information to send.
2100 * \note This will not have any malloced strings so do not free it.
2102 struct ast_party_id forced_clid;
2104 * \brief Stored CallerID information if needed.
2106 * \note If OPT_ORIGINAL_CLID set then this is the o option
2107 * CallerID. Otherwise it is the dialplan extension and hint
2110 * \note This will not have any malloced strings so do not free it.
2112 struct ast_party_id stored_clid;
2114 * \brief CallerID party information to store.
2115 * \note This will not have any malloced strings so do not free it.
2117 struct ast_party_caller caller;
2119 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
2120 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
2121 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
2122 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
2123 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
2124 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
2126 if (ast_strlen_zero(data)) {
2127 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2128 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2132 parse = ast_strdupa(data);
2134 AST_STANDARD_APP_ARGS(args, parse);
2136 if (!ast_strlen_zero(args.options) &&
2137 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
2138 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2142 if (ast_strlen_zero(args.peers)) {
2143 ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
2144 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2148 if (ast_cc_call_init(chan, &ignore_cc)) {
2152 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
2153 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
2155 if (delprivintro < 0 || delprivintro > 1) {
2156 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
2161 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
2162 opt_args[OPT_ARG_RINGBACK] = NULL;
2165 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
2166 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
2167 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
2170 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
2171 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
2172 if (!calldurationlimit.tv_sec) {
2173 ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
2174 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2177 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
2180 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
2181 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
2182 dtmfcalled = strsep(&dtmf_progress, ":");
2183 dtmfcalling = strsep(&dtmf_progress, ":");
2186 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
2187 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
2191 /* Setup the forced CallerID information to send if used. */
2192 ast_party_id_init(&forced_clid);
2193 force_forwards_only = 0;
2194 if (ast_test_flag64(&opts, OPT_FORCECLID)) {
2195 if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2196 ast_channel_lock(chan);
2197 forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2198 ast_channel_unlock(chan);
2199 forced_clid_name[0] = '\0';
2200 forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
2201 sizeof(forced_clid_name), chan);
2202 force_forwards_only = 1;
2204 /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
2205 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
2206 &forced_clid.number.str);
2208 if (!ast_strlen_zero(forced_clid.name.str)) {
2209 forced_clid.name.valid = 1;
2211 if (!ast_strlen_zero(forced_clid.number.str)) {
2212 forced_clid.number.valid = 1;
2215 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
2216 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
2217 forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
2219 forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2220 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
2221 && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
2224 pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
2226 forced_clid.number.presentation = pres;
2230 /* Setup the stored CallerID information if needed. */
2231 ast_party_id_init(&stored_clid);
2232 if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
2233 if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
2234 ast_channel_lock(chan);
2235 ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
2236 if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
2237 stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
2239 if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
2240 stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
2242 if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
2243 stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
2245 if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
2246 stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
2248 ast_channel_unlock(chan);
2250 /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
2251 ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
2252 &stored_clid.number.str);
2253 if (!ast_strlen_zero(stored_clid.name.str)) {
2254 stored_clid.name.valid = 1;
2256 if (!ast_strlen_zero(stored_clid.number.str)) {
2257 stored_clid.number.valid = 1;
2262 * In case the new channel has no preset CallerID number by the
2263 * channel driver, setup the dialplan extension and hint name.
2265 stored_clid_name[0] = '\0';
2266 stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
2267 sizeof(stored_clid_name), chan);
2268 if (ast_strlen_zero(stored_clid.name.str)) {
2269 stored_clid.name.str = NULL;
2271 stored_clid.name.valid = 1;
2273 ast_channel_lock(chan);
2274 stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
2275 stored_clid.number.valid = 1;
2276 ast_channel_unlock(chan);
2279 if (ast_test_flag64(&opts, OPT_RESETCDR) && ast_channel_cdr(chan))
2280 ast_cdr_reset(ast_channel_cdr(chan), NULL);
2281 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
2282 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
2284 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
2285 res = setup_privacy_args(&pa, &opts, opt_args, chan);
2288 res = -1; /* reset default */
2294 /* If a channel group has been specified, get it for use when we create peer channels */
2296 ast_channel_lock(chan);
2297 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
2298 outbound_group = ast_strdupa(outbound_group);
2299 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
2300 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
2301 outbound_group = ast_strdupa(outbound_group);
2303 ast_channel_unlock(chan);
2305 /* Set per dial instance flags. These flags are also passed back to RetryDial. */
2306 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
2307 | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
2308 | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
2310 /* PREDIAL: Run gosub on the caller's channel */
2311 if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
2312 && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
2313 ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
2314 ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
2317 /* loop through the list of dial destinations */
2319 while ((cur = strsep(&rest, "&")) ) {
2320 struct ast_channel *tc; /* channel for this destination */
2321 /* Get a technology/resource pair */
2323 char *tech = strsep(&number, "/");
2326 /* find if we already dialed this interface */
2327 struct ast_dialed_interface *di;
2328 AST_LIST_HEAD(,ast_dialed_interface) *dialed_interfaces;
2331 if (ast_strlen_zero(number)) {
2332 ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
2336 tech_len = strlen(tech) + 1;
2337 number_len = strlen(number) + 1;
2338 tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
2343 /* Save tech, number, and interface. */
2349 cur[tech_len - 1] = '/';
2350 tmp->interface = cur;
2352 strcpy(cur, number);
2356 /* Set per outgoing call leg options. */
2357 ast_copy_flags64(tmp, &opts,
2358 OPT_CANCEL_ELSEWHERE |
2359 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
2360 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
2361 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
2362 OPT_CALLEE_PARK | OPT_CALLER_PARK |
2363 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
2364 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE);
2365 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
2368 /* Request the peer */
2370 ast_channel_lock(chan);
2371 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
2373 * Seed the chanlist's connected line information with previously
2374 * acquired connected line info from the incoming channel. The
2375 * previously acquired connected line info could have been set
2376 * through the CONNECTED_LINE dialplan function.
2378 ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
2379 ast_channel_unlock(chan);
2382 dialed_interfaces = datastore->data;
2384 if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
2385 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
2389 datastore->inheritance = DATASTORE_INHERIT_FOREVER;
2391 if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
2392 ast_datastore_free(datastore);
2397 datastore->data = dialed_interfaces;
2398 AST_LIST_HEAD_INIT(dialed_interfaces);
2400 ast_channel_lock(chan);
2401 ast_channel_datastore_add(chan, datastore);
2402 ast_channel_unlock(chan);
2405 AST_LIST_LOCK(dialed_interfaces);
2406 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
2407 if (!strcasecmp(di->interface, tmp->interface)) {
2408 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
2413 AST_LIST_UNLOCK(dialed_interfaces);
2420 /* It is always ok to dial a Local interface. We only keep track of
2421 * which "real" interfaces have been dialed. The Local channel will
2422 * inherit this list so that if it ends up dialing a real interface,
2423 * it won't call one that has already been called. */
2424 if (strcasecmp(tmp->tech, "Local")) {
2425 if (!(di = ast_calloc(1, sizeof(*di) + strlen(tmp->interface)))) {
2429 strcpy(di->interface, tmp->interface);
2431 AST_LIST_LOCK(dialed_interfaces);
2432 AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
2433 AST_LIST_UNLOCK(dialed_interfaces);
2436 tc = ast_request(tmp->tech, ast_channel_nativeformats(chan), chan, tmp->number, &cause);
2438 /* If we can't, just go on to the next call */
2439 ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
2440 tmp->tech, cause, ast_cause2str(cause));
2441 handle_cause(cause, &num);
2443 /* we are on the last destination */
2444 ast_channel_hangupcause_set(chan, cause);
2446 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2447 if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
2448 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
2454 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2456 ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
2458 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
2460 ast_channel_lock_both(tc, chan);
2462 /* Setup outgoing SDP to match incoming one */
2463 if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2464 /* We are on the only destination. */
2465 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2468 /* Inherit specially named variables from parent channel */
2469 ast_channel_inherit_variables(chan, tc);
2470 ast_channel_datastore_inherit(chan, tc);
2472 ast_channel_appl_set(tc, "AppDial");
2473 ast_channel_data_set(tc, "(Outgoing Line)");
2474 memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
2476 /* Determine CallerID to store in outgoing channel. */
2477 ast_party_caller_set_init(&caller, ast_channel_caller(tc));
2478 if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2479 caller.id = stored_clid;
2480 ast_channel_set_caller_event(tc, &caller, NULL);
2481 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2482 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
2483 ast_channel_caller(tc)->id.number.str, NULL))) {
2485 * The new channel has no preset CallerID number by the channel
2486 * driver. Use the dialplan extension and hint name.
2488 caller.id = stored_clid;
2489 if (!caller.id.name.valid
2490 && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2491 ast_channel_connected(chan)->id.name.str, NULL))) {
2493 * No hint name available. We have a connected name supplied by
2494 * the dialplan we can use instead.
2496 caller.id.name.valid = 1;
2497 caller.id.name = ast_channel_connected(chan)->id.name;
2499 ast_channel_set_caller_event(tc, &caller, NULL);
2500 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2501 } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
2503 /* The new channel has no preset CallerID name by the channel driver. */
2504 if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
2505 ast_channel_connected(chan)->id.name.str, NULL))) {
2507 * We have a connected name supplied by the dialplan we can
2510 caller.id.name.valid = 1;
2511 caller.id.name = ast_channel_connected(chan)->id.name;
2512 ast_channel_set_caller_event(tc, &caller, NULL);
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