2 * Asterisk -- An open source telephony toolkit.
4 * Copyright (C) 1999 - 2008, 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>
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
38 #include <sys/signal.h>
40 #include <netinet/in.h>
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/cdr.h"
59 #include "asterisk/manager.h"
60 #include "asterisk/privacy.h"
61 #include "asterisk/stringfields.h"
62 #include "asterisk/global_datastores.h"
63 #include "asterisk/dsp.h"
64 #include "asterisk/cel.h"
65 #include "asterisk/aoc.h"
66 #include "asterisk/ccss.h"
67 #include "asterisk/indications.h"
70 <application name="Dial" language="en_US">
72 Attempt to connect to another device or endpoint and bridge the call.
75 <parameter name="Technology/Resource" required="true" argsep="&">
76 <argument name="Technology/Resource" required="true">
77 <para>Specification of the device(s) to dial. These must be in the format of
78 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
79 represents a particular channel driver, and <replaceable>Resource</replaceable>
80 represents a resource available to that particular channel driver.</para>
82 <argument name="Technology2/Resource2" required="false" multiple="true">
83 <para>Optional extra devices to dial in parallel</para>
84 <para>If you need more then one enter them as
85 Technology2/Resource2&Technology3/Resourse3&.....</para>
88 <parameter name="timeout" required="false">
89 <para>Specifies the number of seconds we attempt to dial the specified devices</para>
90 <para>If not specified, this defaults to 136 years.</para>
92 <parameter name="options" required="false">
95 <argument name="x" required="true">
96 <para>The file to play to the called party</para>
98 <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
101 <para>Immediately answer the calling channel when the called channel answers in
102 all cases. Normally, the calling channel is answered when the called channel
103 answers, but when options such as A() and M() are used, the calling channel is
104 not answered until all actions on the called channel (such as playing an
105 announcement) are completed. This option can be used to answer the calling
106 channel before doing anything on the called channel. You will rarely need to use
107 this option, the default behavior is adequate in most cases.</para>
110 <para>Reset the call detail record (CDR) for this call.</para>
113 <para>If the Dial() application cancels this call, always set the flag to tell the channel
114 driver that the call is answered elsewhere.</para>
117 <para>Allow the calling user to dial a 1 digit extension while waiting for
118 a call to be answered. Exit to that extension if it exists in the
119 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
122 <option name="D" argsep=":">
123 <argument name="called" />
124 <argument name="calling" />
125 <argument name="progress" />
126 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
127 party has answered, but before the call gets bridged. The
128 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
129 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
130 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
131 immediately after receiving a PROGRESS message.</para>
134 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
137 <argument name="x" required="false" />
138 <para>If <replaceable>x</replaceable> is not provided, force the callerid of the <emphasis>calling</emphasis>
139 channel to be set as the extension associated with the channel using a dialplan <literal>hint</literal>.
140 For example, some PSTNs do not allow CallerID to be set to anything
141 other than the number assigned to the caller. If <replaceable>x</replaceable> is provided, though, then
142 this option behaves quite differently. Any outgoing channel created will have its connected party information
143 set to <replaceable>x</replaceable></para>
145 <option name="F" argsep="^">
146 <argument name="context" required="false" />
147 <argument name="exten" required="false" />
148 <argument name="priority" required="true" />
149 <para>When the caller hangs up, transfer the called party
150 to the specified destination and continue execution at that location.</para>
153 <para>Proceed with dialplan execution at the next priority in the current extension if the
154 source channel hangs up.</para>
157 <para>Proceed with dialplan execution at the next priority in the current extension if the
158 destination channel hangs up.</para>
160 <option name="G" argsep="^">
161 <argument name="context" required="false" />
162 <argument name="exten" required="false" />
163 <argument name="priority" required="true" />
164 <para>If the call is answered, transfer the calling party to
165 the specified <replaceable>priority</replaceable> and the called party to the specified
166 <replaceable>priority</replaceable> plus one.</para>
168 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
172 <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
175 <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
178 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
181 <para>Asterisk will ignore any connected line update requests or redirecting party update
182 requests it may receiveon this dial attempt.</para>
185 <para>Allow the called party to enable parking of the call by sending
186 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
189 <para>Allow the calling party to enable parking of the call by sending
190 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
192 <option name="L" argsep=":">
193 <argument name="x" required="true">
194 <para>Maximum call time, in milliseconds</para>
197 <para>Warning time, in milliseconds</para>
200 <para>Repeat time, in milliseconds</para>
202 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
203 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
204 <para>This option is affected by the following variables:</para>
206 <variable name="LIMIT_PLAYAUDIO_CALLER">
207 <value name="yes" default="true" />
209 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
211 <variable name="LIMIT_PLAYAUDIO_CALLEE">
213 <value name="no" default="true"/>
214 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
216 <variable name="LIMIT_TIMEOUT_FILE">
217 <value name="filename"/>
218 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
219 If not set, the time remaining will be announced.</para>
221 <variable name="LIMIT_CONNECT_FILE">
222 <value name="filename"/>
223 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
224 If not set, the time remaining will be announced.</para>
226 <variable name="LIMIT_WARNING_FILE">
227 <value name="filename"/>
228 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
229 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
234 <argument name="class" required="false"/>
235 <para>Provide hold music to the calling party until a requested
236 channel answers. A specific music on hold <replaceable>class</replaceable>
237 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
239 <option name="M" argsep="^">
240 <argument name="macro" required="true">
241 <para>Name of the macro that should be executed.</para>
243 <argument name="arg" multiple="true">
244 <para>Macro arguments</para>
246 <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
247 before connecting to the calling channel. Arguments can be specified to the Macro
248 using <literal>^</literal> as a delimiter. The macro can set the variable
249 <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
250 finished executing:</para>
252 <variable name="MACRO_RESULT">
253 <para>If set, this action will be taken after the macro finished executing.</para>
255 Hangup both legs of the call
257 <value name="CONGESTION">
258 Behave as if line congestion was encountered
261 Behave as if a busy signal was encountered
263 <value name="CONTINUE">
264 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
266 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
267 <value name="GOTO:<context>^<exten>^<priority>">
268 Transfer the call to the specified destination.
273 <para>You cannot use any additional action post answer options in conjunction
274 with this option. Also, pbx services are not run on the peer (called) channel,
275 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
277 <warning><para>Be aware of the limitations that macros have, specifically with regards to use of
278 the <literal>WaitExten</literal> application. For more information, see the documentation for
279 Macro()</para></warning>
282 <argument name="delete">
283 <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
284 the recorded introduction will not be deleted if the caller hangs up while the remote party has not
286 <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
287 always be deleted.</para>
289 <para>This option is a modifier for the call screening/privacy mode. (See the
290 <literal>p</literal> and <literal>P</literal> options.) It specifies
291 that no introductions are to be saved in the <directory>priv-callerintros</directory>
295 <para>This option is a modifier for the call screening/privacy mode. It specifies
296 that if Caller*ID is present, do not screen the call.</para>
299 <para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
300 be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
301 behavior of Asterisk 1.0 and earlier.</para>
304 <argument name="mode">
305 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
306 the originator hanging up will cause the phone to ring back immediately.</para>
307 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
308 flashes the trunk, it will ring their phone back.</para>
310 <para>Enables <emphasis>operator services</emphasis> mode. This option only
311 works when bridging a DAHDI channel to another DAHDI channel
312 only. if specified on non-DAHDI interfaces, it will be ignored.
313 When the destination answers (presumably an operator services
314 station), the originator no longer has control of their line.
315 They may hang up, but the switch will not release their line
316 until the destination party (the operator) hangs up.</para>
319 <para>This option enables screening mode. This is basically Privacy mode
320 without memory.</para>
323 <argument name="x" />
324 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
325 it is provided. The current extension is used if a database family/key is not specified.</para>
328 <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
329 party until the called channel has answered.</para>
330 <argument name="tone" required="false">
331 <para>Indicate progress to calling party. Send audio 'tone' from indications.conf</para>
335 <argument name="x" required="true" />
336 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
337 answered the call.</para>
340 <argument name="x" required="true" />
341 <para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable></para>
344 <para>Allow the called party to transfer the calling party by sending the
345 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
346 transfers initiated by other methods.</para>
349 <para>Allow the calling party to transfer the called party by sending the
350 DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
351 transfers initiated by other methods.</para>
353 <option name="U" argsep="^">
354 <argument name="x" required="true">
355 <para>Name of the subroutine to execute via Gosub</para>
357 <argument name="arg" multiple="true" required="false">
358 <para>Arguments for the Gosub routine</para>
360 <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
361 to the calling channel. Arguments can be specified to the Gosub
362 using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
363 <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
365 <variable name="GOSUB_RESULT">
367 Hangup both legs of the call.
369 <value name="CONGESTION">
370 Behave as if line congestion was encountered.
373 Behave as if a busy signal was encountered.
375 <value name="CONTINUE">
376 Hangup the called party and allow the calling party
377 to continue dialplan execution at the next priority.
379 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
380 <value name="GOTO:<context>^<exten>^<priority>">
381 Transfer the call to the specified priority. Optionally, an extension, or
382 extension and priority can be specified.
387 <para>You cannot use any additional action post answer options in conjunction
388 with this option. Also, pbx services are not run on the peer (called) channel,
389 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
393 <argument name = "x" required="true">
394 <para>Force the outgoing callerid presentation indicator parameter to be set
395 to one of the values passed in <replaceable>x</replaceable>:
396 <literal>allowed_not_screened</literal>
397 <literal>allowed_passed_screen</literal>
398 <literal>allowed_failed_screen</literal>
399 <literal>allowed</literal>
400 <literal>prohib_not_screened</literal>
401 <literal>prohib_passed_screen</literal>
402 <literal>prohib_failed_screen</literal>
403 <literal>prohib</literal>
404 <literal>unavailable</literal></para>
408 <para>Allow the called party to enable recording of the call by sending
409 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
412 <para>Allow the calling party to enable recording of the call by sending
413 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
416 <para>Allow the called party to enable recording of the call by sending
417 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
420 <para>Allow the calling party to enable recording of the call by sending
421 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
424 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
428 <parameter name="URL">
429 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
433 <para>This application will place calls to one or more specified channels. As soon
434 as one of the requested channels answers, the originating channel will be
435 answered, if it has not already been answered. These two channels will then
436 be active in a bridged call. All other channels that were requested will then
439 <para>Unless there is a timeout specified, the Dial application will wait
440 indefinitely until one of the called channels answers, the user hangs up, or
441 if all of the called channels are busy or unavailable. Dialplan executing will
442 continue if no requested channels can be called, or if the timeout expires.
443 This application will report normal termination if the originating channel
444 hangs up, or if the call is bridged and either of the parties in the bridge
445 ends the call.</para>
446 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
447 application will be put into that group (as in Set(GROUP()=...).
448 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
449 application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
450 however, the variable will be unset after use.</para>
452 <para>This application sets the following channel variables:</para>
454 <variable name="DIALEDTIME">
455 <para>This is the time from dialing a channel until when it is disconnected.</para>
457 <variable name="ANSWEREDTIME">
458 <para>This is the amount of time for actual call.</para>
460 <variable name="DIALSTATUS">
461 <para>This is the status of the call</para>
462 <value name="CHANUNAVAIL" />
463 <value name="CONGESTION" />
464 <value name="NOANSWER" />
465 <value name="BUSY" />
466 <value name="ANSWER" />
467 <value name="CANCEL" />
468 <value name="DONTCALL">
469 For the Privacy and Screening Modes.
470 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
472 <value name="TORTURE">
473 For the Privacy and Screening Modes.
474 Will be set if the called party chooses to send the calling party to the 'torture' script.
476 <value name="INVALIDARGS" />
481 <application name="RetryDial" language="en_US">
483 Place a call, retrying on failure allowing an optional exit extension.
486 <parameter name="announce" required="true">
487 <para>Filename of sound that will be played when no channel can be reached</para>
489 <parameter name="sleep" required="true">
490 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
492 <parameter name="retries" required="true">
493 <para>Number of retries</para>
494 <para>When this is reached flow will continue at the next priority in the dialplan</para>
496 <parameter name="dialargs" required="true">
497 <para>Same format as arguments provided to the Dial application</para>
501 <para>This application will attempt to place a call using the normal Dial application.
502 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
503 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
504 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
505 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
506 While waiting to retry a call, a 1 digit extension may be dialed. If that
507 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
508 one, The call will jump to that extension immediately.
509 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
510 to the Dial application.</para>
515 static const char app[] = "Dial";
516 static const char rapp[] = "RetryDial";
519 OPT_ANNOUNCE = (1 << 0),
520 OPT_RESETCDR = (1 << 1),
521 OPT_DTMF_EXIT = (1 << 2),
522 OPT_SENDDTMF = (1 << 3),
523 OPT_FORCECLID = (1 << 4),
524 OPT_GO_ON = (1 << 5),
525 OPT_CALLEE_HANGUP = (1 << 6),
526 OPT_CALLER_HANGUP = (1 << 7),
527 OPT_ORIGINAL_CLID = (1 << 8),
528 OPT_DURATION_LIMIT = (1 << 9),
529 OPT_MUSICBACK = (1 << 10),
530 OPT_CALLEE_MACRO = (1 << 11),
531 OPT_SCREEN_NOINTRO = (1 << 12),
532 OPT_SCREEN_NOCALLERID = (1 << 13),
533 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
534 OPT_SCREENING = (1 << 15),
535 OPT_PRIVACY = (1 << 16),
536 OPT_RINGBACK = (1 << 17),
537 OPT_DURATION_STOP = (1 << 18),
538 OPT_CALLEE_TRANSFER = (1 << 19),
539 OPT_CALLER_TRANSFER = (1 << 20),
540 OPT_CALLEE_MONITOR = (1 << 21),
541 OPT_CALLER_MONITOR = (1 << 22),
542 OPT_GOTO = (1 << 23),
543 OPT_OPERMODE = (1 << 24),
544 OPT_CALLEE_PARK = (1 << 25),
545 OPT_CALLER_PARK = (1 << 26),
546 OPT_IGNORE_FORWARDING = (1 << 27),
547 OPT_CALLEE_GOSUB = (1 << 28),
548 OPT_CALLEE_MIXMONITOR = (1 << 29),
549 OPT_CALLER_MIXMONITOR = (1 << 30),
550 OPT_CALLER_ANSWER = (1 << 31),
553 #define DIAL_STILLGOING (1 << 31)
554 #define DIAL_NOFORWARDHTML ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
555 #define DIAL_CALLERID_ABSENT ((uint64_t)1 << 33) /* TRUE if caller id is not available for connected line. */
556 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 34)
557 #define OPT_PEER_H ((uint64_t)1 << 35)
558 #define OPT_CALLEE_GO_ON ((uint64_t)1 << 36)
559 #define OPT_CANCEL_TIMEOUT ((uint64_t)1 << 37)
560 #define OPT_FORCE_CID_TAG ((uint64_t)1 << 38)
561 #define OPT_FORCE_CID_PRES ((uint64_t)1 << 39)
564 OPT_ARG_ANNOUNCE = 0,
567 OPT_ARG_DURATION_LIMIT,
569 OPT_ARG_CALLEE_MACRO,
571 OPT_ARG_CALLEE_GOSUB,
572 OPT_ARG_CALLEE_GO_ON,
574 OPT_ARG_DURATION_STOP,
576 OPT_ARG_SCREEN_NOINTRO,
578 OPT_ARG_FORCE_CID_TAG,
579 OPT_ARG_FORCE_CID_PRES,
580 /* note: this entry _MUST_ be the last one in the enum */
584 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
585 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
586 AST_APP_OPTION('a', OPT_CALLER_ANSWER),
587 AST_APP_OPTION('C', OPT_RESETCDR),
588 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
589 AST_APP_OPTION('d', OPT_DTMF_EXIT),
590 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
591 AST_APP_OPTION('e', OPT_PEER_H),
592 AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
593 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
594 AST_APP_OPTION('g', OPT_GO_ON),
595 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
596 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
597 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
598 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
599 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
600 AST_APP_OPTION('k', OPT_CALLEE_PARK),
601 AST_APP_OPTION('K', OPT_CALLER_PARK),
602 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
603 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
604 AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
605 AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
606 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
607 AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
608 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
609 AST_APP_OPTION('p', OPT_SCREENING),
610 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
611 AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
612 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
613 AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
614 AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
615 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
616 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
617 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
618 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
619 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
620 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
621 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
622 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
625 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
626 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
627 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
628 OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
629 !chan->audiohooks && !peer->audiohooks)
632 * The list of active channels
635 struct chanlist *next;
636 struct ast_channel *chan;
638 /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
639 struct ast_party_connected_line connected;
640 /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
641 unsigned int pending_connected_update:1;
642 struct ast_aoc_decoded *aoc_s_rate_list;
645 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
647 static void chanlist_free(struct chanlist *outgoing)
649 ast_party_connected_line_free(&outgoing->connected);
650 ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
654 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
656 /* Hang up a tree of stuff */
659 /* Hangup any existing lines we have open */
660 if (outgoing->chan && (outgoing->chan != exception)) {
661 if (answered_elsewhere) {
662 /* The flag is used for local channel inheritance and stuff */
663 ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
664 /* This is for the channel drivers */
665 outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
667 ast_hangup(outgoing->chan);
670 outgoing = outgoing->next;
675 #define AST_MAX_WATCHERS 256
678 * argument to handle_cause() and other functions.
681 struct ast_channel *chan;
687 static void handle_cause(int cause, struct cause_args *num)
689 struct ast_cdr *cdr = num->chan->cdr;
698 case AST_CAUSE_CONGESTION:
704 case AST_CAUSE_NO_ROUTE_DESTINATION:
705 case AST_CAUSE_UNREGISTERED:
711 case AST_CAUSE_NO_ANSWER:
713 ast_cdr_noanswer(cdr);
716 case AST_CAUSE_NORMAL_CLEARING:
725 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
727 char rexten[2] = { exten, '\0' };
730 if (!ast_goto_if_exists(chan, context, rexten, pri))
733 if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
735 else if (!ast_strlen_zero(chan->macrocontext)) {
736 if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
743 /* do not call with chan lock held */
744 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
749 ast_channel_lock(chan);
750 context = ast_strdupa(S_OR(chan->macrocontext, chan->context));
751 exten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
752 ast_channel_unlock(chan);
754 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
757 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
759 struct ast_channel *chans[] = { src, dst };
760 ast_manager_event_multichan(EVENT_FLAG_CALL, "Dial", 2, chans,
761 "SubEvent: Begin\r\n"
763 "Destination: %s\r\n"
764 "CallerIDNum: %s\r\n"
765 "CallerIDName: %s\r\n"
767 "DestUniqueID: %s\r\n"
768 "Dialstring: %s\r\n",
769 src->name, dst->name,
770 S_COR(src->caller.id.number.valid, src->caller.id.number.str, "<unknown>"),
771 S_COR(src->caller.id.name.valid, src->caller.id.name.str, "<unknown>"),
772 src->uniqueid, dst->uniqueid,
773 dialstring ? dialstring : "");
776 static void senddialendevent(struct ast_channel *src, const char *dialstatus)
778 ast_manager_event(src, EVENT_FLAG_CALL, "Dial",
782 "DialStatus: %s\r\n",
783 src->name, src->uniqueid, dialstatus);
787 * helper function for wait_for_answer()
789 * XXX this code is highly suspicious, as it essentially overwrites
790 * the outgoing channel without properly deleting it.
792 * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
794 static void do_forward(struct chanlist *o,
795 struct cause_args *num, struct ast_flags64 *peerflags, int single, int *to)
798 struct ast_channel *original = o->chan;
799 struct ast_channel *c = o->chan; /* the winner */
800 struct ast_channel *in = num->chan; /* the input channel */
805 ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
806 if ((stuff = strchr(tmpchan, '/'))) {
810 const char *forward_context;
812 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
813 if (ast_strlen_zero(forward_context)) {
814 forward_context = NULL;
816 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
817 ast_channel_unlock(c);
822 ast_cel_report_event(in, AST_CEL_FORWARD, NULL, c->call_forward, NULL);
824 /* Before processing channel, go ahead and check for forwarding */
825 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
826 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
827 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
828 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
830 cause = AST_CAUSE_BUSY;
832 /* Setup parameters */
833 c = o->chan = ast_request(tech, in->nativeformats, in, stuff, &cause);
836 ast_channel_make_compatible(o->chan, in);
837 ast_channel_inherit_variables(in, o->chan);
838 ast_channel_datastore_inherit(in, o->chan);
839 /* When a call is forwarded, we don't want to track new interfaces
840 * dialed for CC purposes. Setting the done flag will ensure that
841 * any Dial operations that happen later won't record CC interfaces.
843 ast_ignore_cc(o->chan);
844 ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", o->chan->name);
847 "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
851 ast_clear_flag64(o, DIAL_STILLGOING);
852 handle_cause(cause, num);
853 ast_hangup(original);
855 struct ast_party_redirecting redirecting;
857 if (single && CAN_EARLY_BRIDGE(peerflags, c, in)) {
858 ast_rtp_instance_early_bridge_make_compatible(c, in);
861 ast_channel_set_redirecting(c, &original->redirecting, NULL);
863 while (ast_channel_trylock(in)) {
864 CHANNEL_DEADLOCK_AVOIDANCE(c);
866 if (!c->redirecting.from.number.valid
867 || ast_strlen_zero(c->redirecting.from.number.str)) {
869 * The call was not previously redirected so it is
870 * now redirected from this number.
872 ast_party_number_free(&c->redirecting.from.number);
873 ast_party_number_init(&c->redirecting.from.number);
874 c->redirecting.from.number.valid = 1;
875 c->redirecting.from.number.str =
876 ast_strdup(S_OR(in->macroexten, in->exten));
879 c->dialed.transit_network_select = in->dialed.transit_network_select;
881 if (ast_test_flag64(o, OPT_FORCECLID)) {
882 ast_party_id_free(&c->caller.id);
883 ast_party_id_init(&c->caller.id);
884 c->caller.id.number.valid = 1;
885 c->caller.id.number.str = ast_strdup(S_OR(in->macroexten, in->exten));
886 ast_string_field_set(c, accountcode, c->accountcode);
888 ast_party_caller_copy(&c->caller, &in->caller);
889 ast_string_field_set(c, accountcode, in->accountcode);
891 ast_party_connected_line_copy(&c->connected, &original->connected);
893 c->data = "(Outgoing Line)";
895 * We must unlock c before calling ast_channel_redirecting_macro, because
896 * we put c into autoservice there. That is pretty much a guaranteed
897 * deadlock. This is why the handling of c's lock may seem a bit unusual
900 ast_party_redirecting_init(&redirecting);
901 ast_party_redirecting_copy(&redirecting, &c->redirecting);
902 ast_channel_unlock(c);
903 if (ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
904 ast_channel_update_redirecting(in, &redirecting, NULL);
906 ast_party_redirecting_free(&redirecting);
907 ast_channel_unlock(in);
909 ast_clear_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE);
910 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
914 if (ast_call(c, stuff, 0)) {
915 ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
917 ast_clear_flag64(o, DIAL_STILLGOING);
918 ast_hangup(original);
924 while (ast_channel_trylock(in)) {
925 CHANNEL_DEADLOCK_AVOIDANCE(c);
927 senddialevent(in, c, stuff);
928 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
929 char cidname[AST_MAX_EXTENSION] = "";
930 const char *tmpexten;
931 tmpexten = ast_strdupa(S_OR(in->macroexten, in->exten));
932 ast_channel_unlock(in);
933 ast_channel_unlock(c);
934 ast_set_callerid(c, tmpexten, get_cid_name(cidname, sizeof(cidname), in), NULL);
936 ast_channel_unlock(in);
937 ast_channel_unlock(c);
939 /* Hangup the original channel now, in case we needed it */
940 ast_hangup(original);
943 ast_indicate(in, -1);
948 /* argument used for some functions. */
949 struct privacy_args {
953 char privintro[1024];
957 static struct ast_channel *wait_for_answer(struct ast_channel *in,
958 struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
960 struct privacy_args *pa,
961 const struct cause_args *num_in, int *result, char *dtmf_progress,
964 struct cause_args num = *num_in;
965 int prestart = num.busy + num.congestion + num.nochan;
967 struct ast_channel *peer = NULL;
968 /* single is set if only one destination is enabled */
969 int single = outgoing && !outgoing->next;
971 struct chanlist *epollo;
973 struct ast_party_connected_line connected_caller;
974 struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
975 int cc_recall_core_id;
977 int cc_frame_received = 0;
980 ast_party_connected_line_init(&connected_caller);
982 /* Turn off hold music, etc */
983 if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK)) {
984 ast_deactivate_generator(in);
985 /* If we are calling a single channel, and not providing ringback or music, */
986 /* then, make them compatible for in-band tone purpose */
987 ast_channel_make_compatible(outgoing->chan, in);
990 if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
991 ast_channel_lock(outgoing->chan);
992 ast_connected_line_copy_from_caller(&connected_caller, &outgoing->chan->caller);
993 ast_channel_unlock(outgoing->chan);
994 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
995 ast_channel_update_connected_line(in, &connected_caller, NULL);
996 ast_party_connected_line_free(&connected_caller);
1000 is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
1003 for (epollo = outgoing; epollo; epollo = epollo->next)
1004 ast_poll_channel_add(in, epollo->chan);
1007 while (*to && !peer) {
1009 int pos = 0; /* how many channels do we handle */
1010 int numlines = prestart;
1011 struct ast_channel *winner;
1012 struct ast_channel *watchers[AST_MAX_WATCHERS];
1014 watchers[pos++] = in;
1015 for (o = outgoing; o; o = o->next) {
1016 /* Keep track of important channels */
1017 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
1018 watchers[pos++] = o->chan;
1021 if (pos == 1) { /* only the input channel is available */
1022 if (numlines == (num.busy + num.congestion + num.nochan)) {
1023 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1025 strcpy(pa->status, "BUSY");
1026 else if (num.congestion)
1027 strcpy(pa->status, "CONGESTION");
1028 else if (num.nochan)
1029 strcpy(pa->status, "CHANUNAVAIL");
1031 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
1035 ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
1039 winner = ast_waitfor_n(watchers, pos, to);
1040 for (o = outgoing; o; o = o->next) {
1041 struct ast_frame *f;
1042 struct ast_channel *c = o->chan;
1046 if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
1048 ast_verb(3, "%s answered %s\n", c->name, in->name);
1049 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1050 if (o->pending_connected_update) {
1051 if (ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1052 ast_channel_update_connected_line(in, &o->connected, NULL);
1054 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1055 ast_channel_lock(c);
1056 ast_connected_line_copy_from_caller(&connected_caller, &c->caller);
1057 ast_channel_unlock(c);
1058 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1059 ast_channel_update_connected_line(in, &connected_caller, NULL);
1060 ast_party_connected_line_free(&connected_caller);
1063 if (o->aoc_s_rate_list) {
1064 size_t encoded_size;
1065 struct ast_aoc_encoded *encoded;
1066 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1067 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1068 ast_aoc_destroy_encoded(encoded);
1072 ast_copy_flags64(peerflags, o,
1073 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1074 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1075 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1076 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1077 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1078 DIAL_NOFORWARDHTML);
1079 ast_string_field_set(c, dialcontext, "");
1080 ast_copy_string(c->exten, "", sizeof(c->exten));
1086 /* here, o->chan == c == winner */
1087 if (!ast_strlen_zero(c->call_forward)) {
1088 pa->sentringing = 0;
1089 if (!ignore_cc && (f = ast_read(c))) {
1090 if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
1091 /* This channel is forwarding the call, and is capable of CC, so
1092 * be sure to add the new device interface to the list
1094 ast_handle_cc_control_frame(in, c, f->data.ptr);
1098 do_forward(o, &num, peerflags, single, to);
1101 f = ast_read(winner);
1103 in->hangupcause = c->hangupcause;
1105 ast_poll_channel_del(in, c);
1109 ast_clear_flag64(o, DIAL_STILLGOING);
1110 handle_cause(in->hangupcause, &num);
1113 if (f->frametype == AST_FRAME_CONTROL) {
1114 switch (f->subclass.integer) {
1115 case AST_CONTROL_ANSWER:
1116 /* This is our guy if someone answered. */
1118 ast_verb(3, "%s answered %s\n", c->name, in->name);
1119 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1120 if (o->pending_connected_update) {
1121 if (ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
1122 ast_channel_update_connected_line(in, &o->connected, NULL);
1124 } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
1125 ast_channel_lock(c);
1126 ast_connected_line_copy_from_caller(&connected_caller, &c->caller);
1127 ast_channel_unlock(c);
1128 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
1129 ast_channel_update_connected_line(in, &connected_caller, NULL);
1130 ast_party_connected_line_free(&connected_caller);
1133 if (o->aoc_s_rate_list) {
1134 size_t encoded_size;
1135 struct ast_aoc_encoded *encoded;
1136 if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
1137 ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
1138 ast_aoc_destroy_encoded(encoded);
1143 peer->cdr->answer = ast_tvnow();
1144 peer->cdr->disposition = AST_CDR_ANSWERED;
1146 ast_copy_flags64(peerflags, o,
1147 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1148 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1149 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1150 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1151 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1152 DIAL_NOFORWARDHTML);
1153 ast_string_field_set(c, dialcontext, "");
1154 ast_copy_string(c->exten, "", sizeof(c->exten));
1155 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
1156 /* Setup early bridge if appropriate */
1157 ast_channel_early_bridge(in, peer);
1159 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1160 in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1161 c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1163 case AST_CONTROL_BUSY:
1164 ast_verb(3, "%s is busy\n", c->name);
1165 in->hangupcause = c->hangupcause;
1168 ast_clear_flag64(o, DIAL_STILLGOING);
1169 handle_cause(AST_CAUSE_BUSY, &num);
1171 case AST_CONTROL_CONGESTION:
1172 ast_verb(3, "%s is circuit-busy\n", c->name);
1173 in->hangupcause = c->hangupcause;
1176 ast_clear_flag64(o, DIAL_STILLGOING);
1177 handle_cause(AST_CAUSE_CONGESTION, &num);
1179 case AST_CONTROL_RINGING:
1180 /* This is a tricky area to get right when using a native
1181 * CC agent. The reason is that we do the best we can to send only a
1182 * single ringing notification to the caller.
1184 * Call completion complicates the logic used here. CCNR is typically
1185 * offered during a ringing message. Let's say that party A calls
1186 * parties B, C, and D. B and C do not support CC requests, but D
1187 * does. If we were to receive a ringing notification from B before
1188 * the others, then we would end up sending a ringing message to
1189 * A with no CCNR offer present.
1191 * The approach that we have taken is that if we receive a ringing
1192 * response from a party and no CCNR offer is present, we need to
1193 * wait. Specifically, we need to wait until either a) a called party
1194 * offers CCNR in its ringing response or b) all called parties have
1195 * responded in some way to our call and none offers CCNR.
1197 * The drawback to this is that if one of the parties has a delayed
1198 * response or, god forbid, one just plain doesn't respond to our
1199 * outgoing call, then this will result in a significant delay between
1200 * when the caller places the call and hears ringback.
1202 * Note also that if CC is disabled for this call, then it is perfectly
1203 * fine for ringing frames to get sent through.
1206 if (ignore_cc || cc_frame_received || num_ringing == numlines) {
1207 ast_verb(3, "%s is ringing\n", c->name);
1208 /* Setup early media if appropriate */
1209 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1210 ast_channel_early_bridge(in, c);
1211 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
1212 ast_indicate(in, AST_CONTROL_RINGING);
1217 case AST_CONTROL_PROGRESS:
1218 ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
1219 /* Setup early media if appropriate */
1220 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1221 ast_channel_early_bridge(in, c);
1222 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1223 if (single || (!single && !pa->sentringing)) {
1224 ast_indicate(in, AST_CONTROL_PROGRESS);
1226 if(!ast_strlen_zero(dtmf_progress)) {
1227 ast_verb(3, "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n", dtmf_progress);
1228 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1231 case AST_CONTROL_VIDUPDATE:
1232 ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
1233 ast_indicate(in, AST_CONTROL_VIDUPDATE);
1235 case AST_CONTROL_SRCUPDATE:
1236 ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
1237 ast_indicate(in, AST_CONTROL_SRCUPDATE);
1239 case AST_CONTROL_CONNECTED_LINE:
1240 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1241 ast_verb(3, "Connected line update to %s prevented.\n", in->name);
1242 } else if (!single) {
1243 struct ast_party_connected_line connected;
1244 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n", c->name, in->name);
1245 ast_party_connected_line_set_init(&connected, &o->connected);
1246 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1247 ast_party_connected_line_set(&o->connected, &connected, NULL);
1248 ast_party_connected_line_free(&connected);
1249 o->pending_connected_update = 1;
1251 if (ast_channel_connected_line_macro(c, in, f, 1, 1)) {
1252 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1256 case AST_CONTROL_AOC:
1258 struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
1259 if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
1260 ast_aoc_destroy_decoded(o->aoc_s_rate_list);
1261 o->aoc_s_rate_list = decoded;
1263 ast_aoc_destroy_decoded(decoded);
1267 case AST_CONTROL_REDIRECTING:
1268 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1269 ast_verb(3, "Redirecting update to %s prevented.\n", in->name);
1271 ast_verb(3, "%s redirecting info has changed, passing it to %s\n", c->name, in->name);
1272 if (ast_channel_redirecting_macro(c, in, f, 1, 1)) {
1273 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1275 pa->sentringing = 0;
1278 case AST_CONTROL_PROCEEDING:
1279 ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
1280 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1281 ast_channel_early_bridge(in, c);
1282 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1283 ast_indicate(in, AST_CONTROL_PROCEEDING);
1285 case AST_CONTROL_HOLD:
1286 ast_verb(3, "Call on %s placed on hold\n", c->name);
1287 ast_indicate(in, AST_CONTROL_HOLD);
1289 case AST_CONTROL_UNHOLD:
1290 ast_verb(3, "Call on %s left from hold\n", c->name);
1291 ast_indicate(in, AST_CONTROL_UNHOLD);
1293 case AST_CONTROL_OFFHOOK:
1294 case AST_CONTROL_FLASH:
1295 /* Ignore going off hook and flash */
1297 case AST_CONTROL_CC:
1299 ast_handle_cc_control_frame(in, c, f->data.ptr);
1300 cc_frame_received = 1;
1304 if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
1305 ast_verb(3, "%s stopped sounds\n", c->name);
1306 ast_indicate(in, -1);
1307 pa->sentringing = 0;
1311 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
1313 } else if (single) {
1314 switch (f->frametype) {
1315 case AST_FRAME_VOICE:
1316 case AST_FRAME_IMAGE:
1317 case AST_FRAME_TEXT:
1318 if (ast_write(in, f)) {
1319 ast_log(LOG_WARNING, "Unable to write frame\n");
1322 case AST_FRAME_HTML:
1323 if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
1324 ast_log(LOG_WARNING, "Unable to send URL\n");
1334 struct ast_frame *f = ast_read(in);
1336 if (f && (f->frametype != AST_FRAME_VOICE))
1337 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1338 else if (!f || (f->frametype != AST_FRAME_VOICE))
1339 printf("Hangup received on %s\n", in->name);
1341 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
1344 strcpy(pa->status, "CANCEL");
1345 ast_cdr_noanswer(in->cdr);
1347 if (f->data.uint32) {
1348 in->hangupcause = f->data.uint32;
1353 ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
1358 /* now f is guaranteed non-NULL */
1359 if (f->frametype == AST_FRAME_DTMF) {
1360 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1361 const char *context;
1362 ast_channel_lock(in);
1363 context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1364 if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
1365 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
1367 ast_cdr_noanswer(in->cdr);
1368 *result = f->subclass.integer;
1369 strcpy(pa->status, "CANCEL");
1371 ast_channel_unlock(in);
1373 ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
1377 ast_channel_unlock(in);
1380 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1381 detect_disconnect(in, f->subclass.integer, featurecode)) {
1382 ast_verb(3, "User requested call disconnect.\n");
1384 strcpy(pa->status, "CANCEL");
1385 ast_cdr_noanswer(in->cdr);
1388 ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
1394 /* Forward HTML stuff */
1395 if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
1396 if (ast_channel_sendhtml(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1)
1397 ast_log(LOG_WARNING, "Unable to send URL\n");
1399 if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
1400 if (ast_write(outgoing->chan, f))
1401 ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
1403 if (single && (f->frametype == AST_FRAME_CONTROL)) {
1404 if ((f->subclass.integer == AST_CONTROL_HOLD) ||
1405 (f->subclass.integer == AST_CONTROL_UNHOLD) ||
1406 (f->subclass.integer == AST_CONTROL_VIDUPDATE) ||
1407 (f->subclass.integer == AST_CONTROL_SRCUPDATE)) {
1408 ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass.integer, outgoing->chan->name);
1409 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1410 } else if (f->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
1411 if (ast_channel_connected_line_macro(in, outgoing->chan, f, 0, 1)) {
1412 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1414 } else if (f->subclass.integer == AST_CONTROL_REDIRECTING) {
1415 if (ast_channel_redirecting_macro(in, outgoing->chan, f, 0, 1)) {
1416 ast_indicate_data(outgoing->chan, f->subclass.integer, f->data.ptr, f->datalen);
1423 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1424 if (!*to || ast_check_hangup(in))
1425 ast_cdr_noanswer(in->cdr);
1429 for (epollo = outgoing; epollo; epollo = epollo->next) {
1431 ast_poll_channel_del(in, epollo->chan);
1436 ast_cc_completed(in, "Recall completed!");
1441 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
1443 struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
1444 struct ast_call_feature feature = { 0, };
1447 ast_str_append(&featurecode, 1, "%c", code);
1449 res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
1451 if (res != AST_FEATURE_RETURN_STOREDIGITS) {
1452 ast_str_reset(featurecode);
1454 if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
1461 static void replace_macro_delimiter(char *s)
1468 /* returns true if there is a valid privacy reply */
1469 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1473 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1475 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1480 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1481 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1487 /* Get the user's intro, store it in priv-callerintros/$CID,
1488 unless it is already there-- this should be done before the
1489 call is actually dialed */
1491 /* all ring indications and moh for the caller has been halted as soon as the
1492 target extension was picked up. We are going to have to kill some
1493 time and make the caller believe the peer hasn't picked up yet */
1495 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1496 char *original_moh = ast_strdupa(chan->musicclass);
1497 ast_indicate(chan, -1);
1498 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1499 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1500 ast_string_field_set(chan, musicclass, original_moh);
1501 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1502 ast_indicate(chan, AST_CONTROL_RINGING);
1506 /* Start autoservice on the other chan ?? */
1507 res2 = ast_autoservice_start(chan);
1508 /* Now Stream the File */
1509 for (loopcount = 0; loopcount < 3; loopcount++) {
1510 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1512 if (!res2) /* on timeout, play the message again */
1513 res2 = ast_play_and_wait(peer, "priv-callpending");
1514 if (!valid_priv_reply(opts, res2))
1516 /* priv-callpending script:
1517 "I have a caller waiting, who introduces themselves as:"
1520 res2 = ast_play_and_wait(peer, pa->privintro);
1521 if (!valid_priv_reply(opts, res2))
1523 /* now get input from the called party, as to their choice */
1525 /* XXX can we have both, or they are mutually exclusive ? */
1526 if (ast_test_flag64(opts, OPT_PRIVACY))
1527 res2 = ast_play_and_wait(peer, "priv-callee-options");
1528 if (ast_test_flag64(opts, OPT_SCREENING))
1529 res2 = ast_play_and_wait(peer, "screen-callee-options");
1531 /*! \page DialPrivacy Dial Privacy scripts
1532 \par priv-callee-options script:
1533 "Dial 1 if you wish this caller to reach you directly in the future,
1534 and immediately connect to their incoming call
1535 Dial 2 if you wish to send this caller to voicemail now and
1537 Dial 3 to send this caller to the torture menus, now and forevermore.
1538 Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1539 Dial 5 to allow this caller to come straight thru to you in the future,
1540 but right now, just this once, send them to voicemail."
1541 \par screen-callee-options script:
1542 "Dial 1 if you wish to immediately connect to the incoming call
1543 Dial 2 if you wish to send this caller to voicemail.
1544 Dial 3 to send this caller to the torture menus.
1545 Dial 4 to send this caller to a simple "go away" menu.
1547 if (valid_priv_reply(opts, res2))
1549 /* invalid option */
1550 res2 = ast_play_and_wait(peer, "vm-sorry");
1553 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1555 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1556 ast_indicate(chan, -1);
1557 pa->sentringing = 0;
1559 ast_autoservice_stop(chan);
1560 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1561 /* map keypresses to various things, the index is res2 - '1' */
1562 static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1563 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1565 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1566 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1567 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1573 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1576 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1579 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1582 /* XXX should we set status to DENY ? */
1583 if (ast_test_flag64(opts, OPT_PRIVACY))
1585 /* if not privacy, then 5 is the same as "default" case */
1586 default: /* bad input or -1 if failure to start autoservice */
1587 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
1588 /* well, there seems basically two choices. Just patch the caller thru immediately,
1589 or,... put 'em thru to voicemail. */
1590 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1591 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1592 /* XXX should we set status to DENY ? */
1593 /* XXX what about the privacy flags ? */
1597 if (res2 == '1') { /* the only case where we actually connect */
1598 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1599 just clog things up, and it's not useful information, not being tied to a CID */
1600 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1601 ast_filedelete(pa->privintro, NULL);
1602 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1603 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1605 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1607 return 0; /* the good exit path */
1609 ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
1614 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1615 static int setup_privacy_args(struct privacy_args *pa,
1616 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1621 int silencethreshold;
1623 if (chan->caller.id.number.valid
1624 && !ast_strlen_zero(chan->caller.id.number.str)) {
1625 l = ast_strdupa(chan->caller.id.number.str);
1626 ast_shrink_phone_number(l);
1627 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1628 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1629 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1631 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1632 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1637 tnam = ast_strdupa(chan->name);
1638 /* clean the channel name so slashes don't try to end up in disk file name */
1639 for (tn2 = tnam; *tn2; tn2++) {
1640 if (*tn2 == '/') /* any other chars to be afraid of? */
1643 ast_verb(3, "Privacy-- callerid is empty\n");
1645 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
1647 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1650 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1652 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1653 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1654 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1655 pa->privdb_val = AST_PRIVACY_ALLOW;
1656 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1657 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1660 if (pa->privdb_val == AST_PRIVACY_DENY) {
1661 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1662 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1664 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1665 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1666 return 0; /* Is this right? */
1667 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1668 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1669 return 0; /* is this right??? */
1670 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1671 /* Get the user's intro, store it in priv-callerintros/$CID,
1672 unless it is already there-- this should be done before the
1673 call is actually dialed */
1675 /* make sure the priv-callerintros dir actually exists */
1676 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1677 if ((res = ast_mkdir(pa->privintro, 0755))) {
1678 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1682 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1683 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1684 /* the DELUX version of this code would allow this caller the
1685 option to hear and retape their previously recorded intro.
1688 int duration; /* for feedback from play_and_wait */
1689 /* the file doesn't exist yet. Let the caller submit his
1690 vocal intro for posterity */
1691 /* priv-recordintro script:
1693 "At the tone, please say your name:"
1696 silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1698 res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
1699 /* don't think we'll need a lock removed, we took care of
1700 conflicts by naming the pa.privintro file */
1702 /* Delete the file regardless since they hung up during recording */
1703 ast_filedelete(pa->privintro, NULL);
1704 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1705 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1707 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1710 if (!ast_streamfile(chan, "vm-dialout", chan->language) )
1711 ast_waitstream(chan, "");
1714 return 1; /* success */
1717 static void end_bridge_callback(void *data)
1721 struct ast_channel *chan = data;
1729 ast_channel_lock(chan);
1730 if (chan->cdr->answer.tv_sec) {
1731 snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->answer.tv_sec);
1732 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1735 if (chan->cdr->start.tv_sec) {
1736 snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->start.tv_sec);
1737 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1739 ast_channel_unlock(chan);
1742 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
1743 bconfig->end_bridge_callback_data = originator;
1746 static int dial_handle_playtones(struct ast_channel *chan, const char *data)
1748 struct ast_tone_zone_sound *ts = NULL;
1750 const char *str = data;
1752 if (ast_strlen_zero(str)) {
1753 ast_debug(1,"Nothing to play\n");
1757 ts = ast_get_indication_tone(chan->zone, str);
1759 if (ts && ts->data[0]) {
1760 res = ast_playtones_start(chan, 0, ts->data, 0);
1766 ts = ast_tone_zone_sound_unref(ts);
1770 ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
1776 static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
1778 int res = -1; /* default: error */
1779 char *rest, *cur; /* scan the list of destinations */
1780 struct chanlist *outgoing = NULL; /* list of destinations */
1781 struct ast_channel *peer;
1782 int to; /* timeout */
1783 struct cause_args num = { chan, 0, 0, 0 };
1786 char *cid_num = NULL, *cid_name = NULL, *cid_tag = NULL, *cid_pres = NULL;
1788 struct ast_bridge_config config = { { 0, } };
1789 struct timeval calldurationlimit = { 0, };
1790 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
1791 struct privacy_args pa = {
1794 .status = "INVALIDARGS",
1796 int sentringing = 0, moh = 0;
1797 const char *outbound_group = NULL;
1801 int delprivintro = 0;
1802 AST_DECLARE_APP_ARGS(args,
1804 AST_APP_ARG(timeout);
1805 AST_APP_ARG(options);
1808 struct ast_flags64 opts = { 0, };
1809 char *opt_args[OPT_ARG_ARRAY_SIZE];
1810 struct ast_datastore *datastore = NULL;
1811 int fulldial = 0, num_dialed = 0;
1813 char device_name[AST_CHANNEL_NAME];
1815 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
1816 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
1817 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
1818 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
1819 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
1820 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
1822 if (ast_strlen_zero(data)) {
1823 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1824 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1828 parse = ast_strdupa(data);
1830 AST_STANDARD_APP_ARGS(args, parse);
1832 if (!ast_strlen_zero(args.options) &&
1833 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
1834 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1838 if (ast_strlen_zero(args.peers)) {
1839 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1840 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1844 if (ast_cc_call_init(chan, &ignore_cc)) {
1848 if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
1849 delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
1851 if (delprivintro < 0 || delprivintro > 1) {
1852 ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
1857 if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
1858 opt_args[OPT_ARG_RINGBACK] = NULL;
1861 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
1862 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
1863 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
1866 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
1867 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
1868 if (!calldurationlimit.tv_sec) {
1869 ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
1870 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1873 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
1876 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
1877 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
1878 dtmfcalled = strsep(&dtmf_progress, ":");
1879 dtmfcalling = strsep(&dtmf_progress, ":");
1882 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
1883 if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
1887 if (ast_test_flag64(&opts, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID]))
1888 ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &cid_name, &cid_num);
1889 if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG) && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG]))
1890 cid_tag = ast_strdupa(opt_args[OPT_ARG_FORCE_CID_TAG]);
1891 if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES) && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES]))
1892 cid_pres = ast_strdupa(opt_args[OPT_ARG_FORCE_CID_PRES]);
1893 if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
1894 ast_cdr_reset(chan->cdr, NULL);
1895 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
1896 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
1898 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
1899 res = setup_privacy_args(&pa, &opts, opt_args, chan);
1902 res = -1; /* reset default */
1905 if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
1906 __ast_answer(chan, 0, 0);
1912 /* If a channel group has been specified, get it for use when we create peer channels */
1914 ast_channel_lock(chan);
1915 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
1916 outbound_group = ast_strdupa(outbound_group);
1917 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
1918 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
1919 outbound_group = ast_strdupa(outbound_group);
1921 ast_channel_unlock(chan);
1922 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE |
1923 OPT_CANCEL_TIMEOUT | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
1925 /* loop through the list of dial destinations */
1927 while ((cur = strsep(&rest, "&")) ) {
1928 struct chanlist *tmp;
1929 struct ast_channel *tc; /* channel for this destination */
1930 /* Get a technology/[device:]number pair */
1932 char *interface = ast_strdupa(number);
1933 char *tech = strsep(&number, "/");
1934 /* find if we already dialed this interface */
1935 struct ast_dialed_interface *di;
1936 AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
1939 ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
1942 if (!(tmp = ast_calloc(1, sizeof(*tmp))))
1945 ast_copy_flags64(tmp, &opts,
1946 OPT_CANCEL_ELSEWHERE |
1947 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1948 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1949 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1950 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1951 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1952 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
1953 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
1955 ast_copy_string(numsubst, number, sizeof(numsubst));
1956 /* Request the peer */
1958 ast_channel_lock(chan);
1959 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
1961 * Seed the chanlist's connected line information with previously
1962 * acquired connected line info from the incoming channel. The
1963 * previously acquired connected line info could have been set
1964 * through the CONNECTED_LINE dialplan function.
1966 ast_party_connected_line_copy(&tmp->connected, &chan->connected);
1967 ast_channel_unlock(chan);
1970 dialed_interfaces = datastore->data;
1972 if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
1973 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
1978 datastore->inheritance = DATASTORE_INHERIT_FOREVER;
1980 if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
1981 ast_datastore_free(datastore);
1986 datastore->data = dialed_interfaces;
1987 AST_LIST_HEAD_INIT(dialed_interfaces);
1989 ast_channel_lock(chan);
1990 ast_channel_datastore_add(chan, datastore);
1991 ast_channel_unlock(chan);
1994 AST_LIST_LOCK(dialed_interfaces);
1995 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
1996 if (!strcasecmp(di->interface, interface)) {
1997 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
2002 AST_LIST_UNLOCK(dialed_interfaces);
2010 /* It is always ok to dial a Local interface. We only keep track of
2011 * which "real" interfaces have been dialed. The Local channel will
2012 * inherit this list so that if it ends up dialing a real interface,
2013 * it won't call one that has already been called. */
2014 if (strcasecmp(tech, "Local")) {
2015 if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
2016 AST_LIST_UNLOCK(dialed_interfaces);
2020 strcpy(di->interface, interface);
2022 AST_LIST_LOCK(dialed_interfaces);
2023 AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
2024 AST_LIST_UNLOCK(dialed_interfaces);
2027 tc = ast_request(tech, chan->nativeformats, chan, numsubst, &cause);
2029 /* If we can't, just go on to the next call */
2030 ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
2031 tech, cause, ast_cause2str(cause));
2032 handle_cause(cause, &num);
2033 if (!rest) /* we are on the last destination */
2034 chan->hangupcause = cause;
2036 if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
2037 if (!ast_cc_callback(chan, tech, numsubst, ast_cc_busy_interface)) {
2038 ast_cc_extension_monitor_add_dialstring(chan, interface, "");
2043 ast_channel_get_device_name(tc, device_name, sizeof(device_name));
2045 ast_cc_extension_monitor_add_dialstring(chan, interface, device_name);
2047 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
2049 ast_channel_lock(tc);
2050 while (ast_channel_trylock(chan)) {
2051 CHANNEL_DEADLOCK_AVOIDANCE(tc);
2053 /* Setup outgoing SDP to match incoming one */
2054 if (!outgoing && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
2055 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
2058 /* Inherit specially named variables from parent channel */
2059 ast_channel_inherit_variables(chan, tc);
2060 ast_channel_datastore_inherit(chan, tc);
2062 tc->appl = "AppDial";
2063 tc->data = "(Outgoing Line)";
2064 memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
2066 /* If the new channel has no callerid, try to guess what it should be */
2067 if (!tc->caller.id.number.valid) {
2068 if (chan->connected.id.number.valid) {
2069 struct ast_party_caller caller;
2071 ast_party_caller_set_init(&caller, &tc->caller);
2072 caller.id = chan->connected.id;
2073 caller.ani = chan->connected.ani;
2074 ast_channel_set_caller_event(tc, &caller, NULL);
2075 } else if (!ast_strlen_zero(chan->dialed.number.str)) {
2076 ast_set_callerid(tc, chan->dialed.number.str, NULL, NULL);
2077 } else if (!ast_strlen_zero(S_OR(chan->macroexten, chan->exten))) {
2078 ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), NULL, NULL);
2080 ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
2083 if (ast_test_flag64(peerflags, OPT_FORCECLID) && !ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
2084 struct ast_party_connected_line connected;
2087 ast_party_connected_line_set_init(&connected, &tmp->chan->connected);
2089 pres = ast_parse_caller_presentation(cid_pres);
2091 pres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2094 pres = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
2097 connected.id.number.valid = 1;
2098 connected.id.number.str = cid_num;
2099 connected.id.number.presentation = pres;
2102 connected.id.name.valid = 1;
2103 connected.id.name.str = cid_name;
2104 connected.id.name.presentation = pres;
2106 connected.id.tag = cid_tag;
2107 ast_channel_set_connected_line(tmp->chan, &connected, NULL);
2109 ast_connected_line_copy_from_caller(&tc->connected, &chan->caller);
2112 ast_party_redirecting_copy(&tc->redirecting, &chan->redirecting);
2114 tc->dialed.transit_network_select = chan->dialed.transit_network_select;
2116 if (!ast_strlen_zero(chan->accountcode)) {
2117 ast_string_field_set(tc, peeraccount, chan->accountcode);
2119 if (ast_strlen_zero(tc->musicclass))
2120 ast_string_field_set(tc, musicclass, chan->musicclass);
2122 /* Pass ADSI CPE and transfer capability */
2123 tc->adsicpe = chan->adsicpe;
2124 tc->transfercapability = chan->transfercapability;
2126 /* If we have an outbound group, set this peer channel to it */
2128 ast_app_group_set_channel(tc, outbound_group);
2129 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
2130 if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
2131 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
2133 /* Check if we're forced by configuration */
2134 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
2135 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
2138 /* Inherit context and extension */
2139 ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
2140 if (!ast_strlen_zero(chan->macroexten))
2141 ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
2143 ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
2145 ast_channel_unlock(tc);
2146 res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
2148 /* Save the info in cdr's that we called them */
2150 ast_cdr_setdestchan(chan->cdr, tc->name);
2152 /* check the results of ast_call */
2154 /* Again, keep going even if there's an error */
2155 ast_debug(1, "ast call on peer returned %d\n", res);
2156 ast_verb(3, "Couldn't call %s\n", numsubst);
2157 if (tc->hangupcause) {
2158 chan->hangupcause = tc->hangupcause;
2160 ast_channel_unlock(chan);
2161 ast_cc_call_failed(chan, tc, interface);
2167 const char *tmpexten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
2168 senddialevent(chan, tc, numsubst);
2169 ast_verb(3, "Called %s\n", numsubst);
2170 ast_channel_unlock(chan);
2171 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
2172 char cidname[AST_MAX_EXTENSION];
2173 ast_set_callerid(tc, tmpexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
2176 /* Put them in the list of outgoing thingies... We're ready now.
2177 XXX If we're forcibly removed, these outgoing calls won't get
2179 ast_set_flag64(tmp, DIAL_STILLGOING);
2181 tmp->next = outgoing;
2183 /* If this line is up, don't try anybody else */
2184 if (outgoing->chan->_state == AST_STATE_UP)
2188 if (ast_strlen_zero(args.timeout)) {
2191 to = atoi(args.timeout);
2195 ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
2201 strcpy(pa.status, "CHANUNAVAIL");
2202 if (fulldial == num_dialed) {
2207 /* Our status will at least be NOANSWER */
2208 strcpy(pa.status, "NOANSWER");
2209 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
2211 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
2212 char *original_moh = ast_strdupa(chan->musicclass);
2213 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
2214 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
2215 ast_string_field_set(chan, musicclass, original_moh);
2217 ast_moh_start(chan, NULL, NULL);
2219 ast_indicate(chan, AST_CONTROL_PROGRESS);
2220 } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
2221 if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
2222 if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
2223 ast_indicate(chan, AST_CONTROL_RINGING);
2226 ast_indicate(chan, AST_CONTROL_PROGRESS);
2229 ast_indicate(chan, AST_CONTROL_RINGING);
2235 peer = wait_for_answer(chan, outgoing, &to, peerflags, opt_args, &pa, &num, &result, dtmf_progress, ignore_cc);
2237 /* The ast_channel_datastore_remove() function could fail here if the
2238 * datastore was moved to another channel during a masquerade. If this is
2239 * the case, don't free the datastore here because later, when the channel
2240 * to which the datastore was moved hangs up, it will attempt to free this
2241 * datastore again, causing a crash
2243 if (!ast_channel_datastore_remove(chan, datastore))
2244 ast_datastore_free(datastore);
2248 } else if (to) { /* Musta gotten hung up */
2250 } else { /* Nobody answered, next please? */
2254 /* SIP, in particular, sends back this error code to indicate an
2255 * overlap dialled number needs more digits. */
2256 if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
2257 res = AST_PBX_INCOMPLETE;
2260 /* almost done, although the 'else' block is 400 lines */
2264 if (ast_test_flag64(&opts, OPT_CALLER_ANSWER))
2267 strcpy(pa.status, "ANSWER");
2268 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2269 /* Ah ha! Someone answered within the desired timeframe. Of course after this
2270 we will always return with -1 so that it is hung up properly after the
2272 hanguptree(outgoing, peer, 1);
2274 /* If appropriate, log that we have a destination channel and set the answer time */
2276 ast_cdr_setdestchan(chan->cdr, peer->name);
2277 ast_cdr_setanswer(chan->cdr, peer->cdr->answer);
2280 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
2282 ast_channel_lock(peer);
2283 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
2286 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
2287 ast_channel_unlock(peer);
2289 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
2290 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
2291 ast_channel_sendurl( peer, args.url );
2293 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
2294 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
2299 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
2303 struct ast_channel *chans[2];
2304 struct ast_channel *active_chan;
2309 /* we need to stream the announcment while monitoring the caller for a hangup */
2311 /* stream the file */
2312 res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
2315 ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
2318 ast_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
2319 while (peer->stream) {
2322 ms = ast_sched_wait(peer->sched);
2324 if (ms < 0 && !peer->timingfunc) {
2325 ast_stopstream(peer);
2331 active_chan = ast_waitfor_n(chans, 2, &ms);
2333 struct ast_frame *fr = ast_read(active_chan);
2339 switch(fr->frametype) {
2340 case AST_FRAME_DTMF_END:
2341 digit = fr->subclass.integer;
2342 if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
2343 ast_stopstream(peer);
2344 res = ast_senddigit(chan, digit, 0);
2347 case AST_FRAME_CONTROL:
2348 switch (fr->subclass.integer) {
2349 case AST_CONTROL_HANGUP:
2359 /* Ignore all others */
2364 ast_sched_runq(peer->sched);
2366 ast_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
2369 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
2370 /* chan and peer are going into the PBX, they both
2371 * should probably get CDR records. */
2372 ast_clear_flag(chan->cdr, AST_CDR_FLAG_DIALED);
2373 ast_clear_flag(peer->cdr, AST_CDR_FLAG_DIALED);
2375 replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
2376 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
2377 /* peer goes to the same context and extension as chan, so just copy info from chan*/
2378 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2379 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2380 peer->priority = chan->priority + 2;
2381 ast_pbx_start(peer);
2382 hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
2389 if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
2390 struct ast_app *theapp;
2391 const char *macro_result;
2393 res = ast_autoservice_start(chan);
2395 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2399 theapp = pbx_findapp("Macro");
2401 if (theapp && !res) { /* XXX why check res here ? */
2402 /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
2403 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2404 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2406 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
2407 res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
2408 ast_debug(1, "Macro exited with status %d\n", res);
2411 ast_log(LOG_ERROR, "Could not find application Macro\n");
2415 if (ast_autoservice_stop(chan) < 0) {
2419 ast_channel_lock(peer);
2421 if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
2422 char *macro_transfer_dest;
2424 if (!strcasecmp(macro_result, "BUSY")) {
2425 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2426 ast_set_flag64(peerflags, OPT_GO_ON);
2428 } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
2429 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2430 ast_set_flag64(peerflags, OPT_GO_ON);
2432 } else if (!strcasecmp(macro_result, "CONTINUE")) {
2433 /* hangup peer and keep chan alive assuming the macro has changed
2434 the context / exten / priority or perhaps
2435 the next priority in the current exten is desired.
2437 ast_set_flag64(peerflags, OPT_GO_ON);
2439 } else if (!strcasecmp(macro_result, "ABORT")) {
2440 /* Hangup both ends unless the caller has the g flag */
2442 } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
2444 /* perform a transfer to a new extension */
2445 if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
2446 replace_macro_delimiter(macro_transfer_dest);
2447 if (!ast_parseable_goto(chan, macro_transfer_dest))
2448 ast_set_flag64(peerflags, OPT_GO_ON);
2453 ast_channel_unlock(peer);
2456 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
2457 struct ast_app *theapp;
2458 const char *gosub_result;
2459 char *gosub_args, *gosub_argstart;
2462 res9 = ast_autoservice_start(chan);
2464 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2468 theapp = pbx_findapp("Gosub");
2470 if (theapp && !res9) {
2471 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
2473 /* Set where we came from */
2474 ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
2475 ast_copy_string(peer->exten, "s", sizeof(peer->exten));
2478 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
2479 if (gosub_argstart) {
2480 *gosub_argstart = 0;
2481 if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
2482 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2485 *gosub_argstart = ',';
2487 if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
2488 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2494 res9 = pbx_exec(peer, theapp, gosub_args);
2496 struct ast_pbx_args args;
2497 /* A struct initializer fails to compile for this case ... */
2498 memset(&args, 0, sizeof(args));
2499 args.no_hangup_chan = 1;
2500 ast_pbx_run_args(peer, &args);
2502 ast_free(gosub_args);
2503 ast_debug(1, "Gosub exited with status %d\n", res9);
2505 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
2509 ast_log(LOG_ERROR, "Could not find application Gosub\n");
2513 if (ast_autoservice_stop(chan) < 0) {
2514 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2518 ast_channel_lock(peer);
2520 if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
2521 char *gosub_transfer_dest;
2522 const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
2524 /* Inherit return value from the peer, so it can be used in the master */
2526 pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
2529 if (!strcasecmp(gosub_result, "BUSY")) {
2530 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2531 ast_set_flag64(peerflags, OPT_GO_ON);
2533 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
2534 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2535 ast_set_flag64(peerflags, OPT_GO_ON);
2537 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
2538 /* hangup peer and keep chan alive assuming the macro has changed
2539 the context / exten / priority or perhaps
2540 the next priority in the current exten is desired.
2542 ast_set_flag64(peerflags, OPT_GO_ON);
2544 } else if (!strcasecmp(gosub_result, "ABORT")) {
2545 /* Hangup both ends unless the caller has the g flag */
2547 } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
2549 /* perform a transfer to a new extension */
2550 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
2551 replace_macro_delimiter(gosub_transfer_dest);
2552 if (!ast_parseable_goto(chan, gosub_transfer_dest))
2553 ast_set_flag64(peerflags, OPT_GO_ON);
2558 ast_channel_unlock(peer);
2562 if (!ast_tvzero(calldurationlimit)) {
2563 struct timeval whentohangup = calldurationlimit;
2564 peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
2566 if (!ast_strlen_zero(dtmfcalled)) {
2567 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
2568 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
2570 if (!ast_strlen_zero(dtmfcalling)) {
2571 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
2572 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
2576 if (res) { /* some error */
2579 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
2580 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
2581 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
2582 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
2583 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
2584 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
2585 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
2586 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
2587 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
2588 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
2589 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
2590 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
2591 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
2592 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
2593 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
2594 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
2595 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
2596 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
2597 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
2598 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
2599 if (ast_test_flag64(peerflags, OPT_GO_ON))
2600 ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
2602 config.end_bridge_callback = end_bridge_callback;
2603 config.end_bridge_callback_data = chan;
2604 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
2609 } else if (sentringing) {
2611 ast_indicate(chan, -1);
2613 /* Be sure no generators are left on it and reset the visible indication */
2614 ast_deactivate_generator(chan);
2615 chan->visible_indication = 0;
2616 /* Make sure channels are compatible */
2617 res = ast_channel_make_compatible(chan, peer);
2619 ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
2625 struct oprmode oprmode;
2627 oprmode.peer = peer;
2628 oprmode.mode = opermode;
2630 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
2632 res = ast_bridge_call(chan, peer, &config);
2635 strcpy(peer->context, chan->context);
2637 if (ast_test_flag64(&opts, OPT_PEER_H)
2638 && ast_exists_extension(peer, peer->context, "h", 1,
2639 S_COR(peer->caller.id.number.valid, peer->caller.id.number.str, NULL))) {
2644 strcpy(peer->exten, "h");
2646 autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
2647 ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
2649 while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten,
2651 S_COR(peer->caller.id.number.valid, peer->caller.id.number.str, NULL),
2656 if (found && res9) {
2657 /* Something bad happened, or a hangup has been requested. */
2658 ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2659 ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2661 ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP); /* set it back the way it was */
2663 if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
2664 if(!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
2665 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
2666 ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
2669 res = ast_goto_if_exists(peer, chan->context, chan->exten, (chan->priority) + 1);
2670 if (res == AST_PBX_GOTO_FAILED) {
2675 ast_pbx_start(peer);
2677 if (!ast_check_hangup(chan))
2678 chan->hangupcause = peer->hangupcause;
2686 } else if (sentringing) {
2688 ast_indicate(chan, -1);
2691 if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
2692 ast_filedelete(pa.privintro, NULL);
2693 if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
2694 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
2696 ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
2700 ast_channel_early_bridge(chan, NULL);
2701 hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
2702 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2703 senddialendevent(chan, pa.status);
2704 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
2706 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
2707 if (!ast_tvzero(calldurationlimit))
2708 memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
2713 if (config.warning_sound) {
2714 ast_free((char *)config.warning_sound);
2716 if (config.end_sound) {
2717 ast_free((char *)config.end_sound);
2719 if (config.start_sound) {
2720 ast_free((char *)config.start_sound);
2722 ast_ignore_cc(chan);
2726 static int dial_exec(struct ast_channel *chan, const char *data)
2728 struct ast_flags64 peerflags;
2730 memset(&peerflags, 0, sizeof(peerflags));
2732 return dial_exec_full(chan, data, &peerflags, NULL);
2735 static int retrydial_exec(struct ast_channel *chan, const char *data)
2738 const char *context = NULL;
2739 int sleepms = 0, loops = 0, res = -1;
2740 struct ast_flags64 peerflags = { 0, };
2741 AST_DECLARE_APP_ARGS(args,
2742 AST_APP_ARG(announce);
2744 AST_APP_ARG(retries);
2745 AST_APP_ARG(dialdata);
2748 if (ast_strlen_zero(data)) {
2749 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
2753 parse = ast_strdupa(data);
2754 AST_STANDARD_APP_ARGS(args, parse);
2756 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
2759 if (!ast_strlen_zero(args.retries)) {
2760 loops = atoi(args.retries);
2763 if (!args.dialdata) {
2764 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
2772 loops = -1; /* run forever */
2774 ast_channel_lock(chan);
2775 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
2776 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
2777 ast_channel_unlock(chan);
2783 chan->data = "Retrying";
2784 if (ast_test_flag(chan, AST_FLAG_MOH))
2787 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
2792 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
2793 if (!ast_strlen_zero(args.announce)) {
2794 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2795 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2796 ast_waitstream(chan, AST_DIGIT_ANY);
2798 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2800 if (!res && sleepms) {
2801 if (!ast_test_flag(chan, AST_FLAG_MOH))
2802 ast_moh_start(chan, NULL, NULL);
2803 res = ast_waitfordigit(chan, sleepms);
2806 if (!ast_strlen_zero(args.announce)) {
2807 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2808 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2809 res = ast_waitstream(chan, "");
2811 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2814 if (!ast_test_flag(chan, AST_FLAG_MOH))
2815 ast_moh_start(chan, NULL, NULL);
2817 res = ast_waitfordigit(chan, sleepms);
2822 if (res < 0 || res == AST_PBX_INCOMPLETE) {
2824 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
2825 if (onedigit_goto(chan, context, (char) res, 1)) {
2837 if (ast_test_flag(chan, AST_FLAG_MOH))
2843 static int unload_module(void)
2846 struct ast_context *con;
2848 res = ast_unregister_application(app);
2849 res |= ast_unregister_application(rapp);
2851 if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
2852 ast_context_remove_extension2(con, "s", 1, NULL, 0);
2853 ast_context_destroy(con, "app_dial"); /* leave nothing behind */
2859 static int load_module(void)
2862 struct ast_context *con;
2864 con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
2866 ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
2868 ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
2870 res = ast_register_application_xml(app, dial_exec);
2871 res |= ast_register_application_xml(rapp, retrydial_exec);
projects / asterisk/asterisk.git / blob