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"
66 <application name="Dial" language="en_US">
68 Attempt to connect to another device or endpoint and bridge the call.
71 <parameter name="Technology/Resource" required="true" argsep="&">
72 <argument name="Technology/Resource" required="true">
73 <para>Specification of the device(s) to dial. These must be in the format of
74 <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
75 represents a particular channel driver, and <replaceable>Resource</replaceable>
76 represents a resource available to that particular channel driver.</para>
78 <argument name="Technology2/Resource2" required="false" multiple="true">
79 <para>Optional extra devices to dial in parallel</para>
80 <para>If you need more then one enter them as
81 Technology2/Resource2&Technology3/Resourse3&.....</para>
84 <parameter name="timeout" required="false">
85 <para>Specifies the number of seconds we attempt to dial the specified devices</para>
86 <para>If not specified, this defaults to 136 years.</para>
88 <parameter name="options" required="false">
91 <argument name="x" required="true">
92 <para>The file to play to the called party</para>
94 <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
97 <para>Reset the call detail record (CDR) for this call.</para>
100 <para>If the Dial() application cancels this call, always set the flag to tell the channel
101 driver that the call is answered elsewhere.</para>
104 <para>Allow the calling user to dial a 1 digit extension while waiting for
105 a call to be answered. Exit to that extension if it exists in the
106 current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
109 <option name="D" argsep=":">
110 <argument name="called" />
111 <argument name="calling" />
112 <argument name="progress" />
113 <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
114 party has answered, but before the call gets bridged. The
115 <replaceable>called</replaceable> DTMF string is sent to the called party, and the
116 <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
117 can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
118 immediately after receiving a PROGRESS message.</para>
121 <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
124 <para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
125 extension associated with the channel using a dialplan <literal>hint</literal>.
126 For example, some PSTNs do not allow CallerID to be set to anything
127 other than the number assigned to the caller.</para>
129 <option name="F" argsep="^">
130 <argument name="context" required="false" />
131 <argument name="exten" required="false" />
132 <argument name="priority" required="true" />
133 <para>When the caller hangs up, transfer the called party
134 to the specified destination and continue execution at that location.</para>
137 <para>Proceed with dialplan execution at the next priority in the current extension if the
138 source channel hangs up.</para>
141 <para>Proceed with dialplan execution at the next priority in the current extension if the
142 destination channel hangs up.</para>
144 <option name="G" argsep="^">
145 <argument name="context" required="false" />
146 <argument name="exten" required="false" />
147 <argument name="priority" required="true" />
148 <para>If the call is answered, transfer the calling party to
149 the specified <replaceable>priority</replaceable> and the called party to the specified
150 <replaceable>priority</replaceable> plus one.</para>
152 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
156 <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
159 <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
162 <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
165 <para>Asterisk will ignore any connected line update requests or redirecting party update
166 requests it may receiveon this dial attempt.</para>
169 <para>Allow the called party to enable parking of the call by sending
170 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
173 <para>Allow the calling party to enable parking of the call by sending
174 the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
176 <option name="L" argsep=":">
177 <argument name="x" required="true">
178 <para>Maximum call time, in milliseconds</para>
181 <para>Warning time, in milliseconds</para>
184 <para>Repeat time, in milliseconds</para>
186 <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
187 left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
188 <para>This option is affected by the following variables:</para>
190 <variable name="LIMIT_PLAYAUDIO_CALLER">
191 <value name="yes" default="true" />
193 <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
195 <variable name="LIMIT_PLAYAUDIO_CALLEE">
197 <value name="no" default="true"/>
198 <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
200 <variable name="LIMIT_TIMEOUT_FILE">
201 <value name="filename"/>
202 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
203 If not set, the time remaining will be announced.</para>
205 <variable name="LIMIT_CONNECT_FILE">
206 <value name="filename"/>
207 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
208 If not set, the time remaining will be announced.</para>
210 <variable name="LIMIT_WARNING_FILE">
211 <value name="filename"/>
212 <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
213 a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
218 <argument name="class" required="false"/>
219 <para>Provide hold music to the calling party until a requested
220 channel answers. A specific music on hold <replaceable>class</replaceable>
221 (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
223 <option name="M" argsep="^">
224 <argument name="macro" required="true">
225 <para>Name of the macro that should be executed.</para>
227 <argument name="arg" multiple="true">
228 <para>Macro arguments</para>
230 <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
231 before connecting to the calling channel. Arguments can be specified to the Macro
232 using <literal>^</literal> as a delimiter. The macro can set the variable
233 <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
234 finished executing:</para>
236 <variable name="MACRO_RESULT">
237 <para>If set, this action will be taken after the macro finished executing.</para>
239 Hangup both legs of the call
241 <value name="CONGESTION">
242 Behave as if line congestion was encountered
245 Behave as if a busy signal was encountered
247 <value name="CONTINUE">
248 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
250 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
251 <value name="GOTO:<context>^<exten>^<priority>">
252 Transfer the call to the specified destination.
257 <para>You cannot use any additional action post answer options in conjunction
258 with this option. Also, pbx services are not run on the peer (called) channel,
259 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
263 <para>This option is a modifier for the call screening/privacy mode. (See the
264 <literal>p</literal> and <literal>P</literal> options.) It specifies
265 that no introductions are to be saved in the <directory>priv-callerintros</directory>
269 <para>This option is a modifier for the call screening/privacy mode. It specifies
270 that if Caller*ID is present, do not screen the call.</para>
273 <para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
274 be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
275 behavior of Asterisk 1.0 and earlier.</para>
278 <argument name="mode">
279 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
280 the originator hanging up will cause the phone to ring back immediately.</para>
281 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
282 flashes the trunk, it will ring their phone back.</para>
284 <para>Enables <emphasis>operator services</emphasis> mode. This option only
285 works when bridging a DAHDI channel to another DAHDI channel
286 only. if specified on non-DAHDI interfaces, it will be ignored.
287 When the destination answers (presumably an operator services
288 station), the originator no longer has control of their line.
289 They may hang up, but the switch will not release their line
290 until the destination party (the operator) hangs up.</para>
293 <para>This option enables screening mode. This is basically Privacy mode
294 without memory.</para>
297 <argument name="x" />
298 <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
299 it is provided. The current extension is used if a database family/key is not specified.</para>
302 <para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
303 party until the called channel has answered.</para>
306 <argument name="x" required="true" />
307 <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
308 answered the call.</para>
311 <para>Allow the called party to transfer the calling party by sending the
312 DTMF sequence defined in <filename>features.conf</filename>.</para>
315 <para>Allow the calling party to transfer the called party by sending the
316 DTMF sequence defined in <filename>features.conf</filename>.</para>
318 <option name="U" argsep="^">
319 <argument name="x" required="true">
320 <para>Name of the subroutine to execute via Gosub</para>
322 <argument name="arg" multiple="true" required="false">
323 <para>Arguments for the Gosub routine</para>
325 <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
326 to the calling channel. Arguments can be specified to the Gosub
327 using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
328 <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
330 <variable name="GOSUB_RESULT">
332 Hangup both legs of the call.
334 <value name="CONGESTION">
335 Behave as if line congestion was encountered.
338 Behave as if a busy signal was encountered.
340 <value name="CONTINUE">
341 Hangup the called party and allow the calling party
342 to continue dialplan execution at the next priority.
344 <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
345 <value name="GOTO:<context>^<exten>^<priority>">
346 Transfer the call to the specified priority. Optionally, an extension, or
347 extension and priority can be specified.
352 <para>You cannot use any additional action post answer options in conjunction
353 with this option. Also, pbx services are not run on the peer (called) channel,
354 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
358 <para>Allow the called party to enable recording of the call by sending
359 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
362 <para>Allow the calling party to enable recording of the call by sending
363 the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
366 <para>Allow the called party to enable recording of the call by sending
367 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
370 <para>Allow the calling party to enable recording of the call by sending
371 the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
374 <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
378 <parameter name="URL">
379 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
383 <para>This application will place calls to one or more specified channels. As soon
384 as one of the requested channels answers, the originating channel will be
385 answered, if it has not already been answered. These two channels will then
386 be active in a bridged call. All other channels that were requested will then
389 <para>Unless there is a timeout specified, the Dial application will wait
390 indefinitely until one of the called channels answers, the user hangs up, or
391 if all of the called channels are busy or unavailable. Dialplan executing will
392 continue if no requested channels can be called, or if the timeout expires.
393 This application will report normal termination if the originating channel
394 hangs up, or if the call is bridged and either of the parties in the bridge
395 ends the call.</para>
396 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
397 application will be put into that group (as in Set(GROUP()=...).
398 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
399 application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
400 however, the variable will be unset after use.</para>
402 <para>This application sets the following channel variables:</para>
404 <variable name="DIALEDTIME">
405 <para>This is the time from dialing a channel until when it is disconnected.</para>
407 <variable name="ANSWEREDTIME">
408 <para>This is the amount of time for actual call.</para>
410 <variable name="DIALSTATUS">
411 <para>This is the status of the call</para>
412 <value name="CHANUNAVAIL" />
413 <value name="CONGESTION" />
414 <value name="NOANSWER" />
415 <value name="BUSY" />
416 <value name="ANSWER" />
417 <value name="CANCEL" />
418 <value name="DONTCALL">
419 For the Privacy and Screening Modes.
420 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
422 <value name="TORTURE">
423 For the Privacy and Screening Modes.
424 Will be set if the called party chooses to send the calling party to the 'torture' script.
426 <value name="INVALIDARGS" />
431 <application name="RetryDial" language="en_US">
433 Place a call, retrying on failure allowing an optional exit extension.
436 <parameter name="announce" required="true">
437 <para>Filename of sound that will be played when no channel can be reached</para>
439 <parameter name="sleep" required="true">
440 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
442 <parameter name="retries" required="true">
443 <para>Number of retries</para>
444 <para>When this is reached flow will continue at the next priority in the dialplan</para>
446 <parameter name="dialargs" required="true">
447 <para>Same format as arguments provided to the Dial application</para>
451 <para>This application will attempt to place a call using the normal Dial application.
452 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
453 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
454 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
455 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
456 While waiting to retry a call, a 1 digit extension may be dialed. If that
457 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
458 one, The call will jump to that extension immediately.
459 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
460 to the Dial application.</para>
465 static char *app = "Dial";
466 static char *rapp = "RetryDial";
469 OPT_ANNOUNCE = (1 << 0),
470 OPT_RESETCDR = (1 << 1),
471 OPT_DTMF_EXIT = (1 << 2),
472 OPT_SENDDTMF = (1 << 3),
473 OPT_FORCECLID = (1 << 4),
474 OPT_GO_ON = (1 << 5),
475 OPT_CALLEE_HANGUP = (1 << 6),
476 OPT_CALLER_HANGUP = (1 << 7),
477 OPT_ORIGINAL_CLID = (1 << 8),
478 OPT_DURATION_LIMIT = (1 << 9),
479 OPT_MUSICBACK = (1 << 10),
480 OPT_CALLEE_MACRO = (1 << 11),
481 OPT_SCREEN_NOINTRO = (1 << 12),
482 OPT_SCREEN_NOCALLERID = (1 << 13),
483 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
484 OPT_SCREENING = (1 << 15),
485 OPT_PRIVACY = (1 << 16),
486 OPT_RINGBACK = (1 << 17),
487 OPT_DURATION_STOP = (1 << 18),
488 OPT_CALLEE_TRANSFER = (1 << 19),
489 OPT_CALLER_TRANSFER = (1 << 20),
490 OPT_CALLEE_MONITOR = (1 << 21),
491 OPT_CALLER_MONITOR = (1 << 22),
492 OPT_GOTO = (1 << 23),
493 OPT_OPERMODE = (1 << 24),
494 OPT_CALLEE_PARK = (1 << 25),
495 OPT_CALLER_PARK = (1 << 26),
496 OPT_IGNORE_FORWARDING = (1 << 27),
497 OPT_CALLEE_GOSUB = (1 << 28),
498 OPT_CALLEE_MIXMONITOR = (1 << 29),
499 OPT_CALLER_MIXMONITOR = (1 << 30),
502 #define DIAL_STILLGOING (1 << 31)
503 #define DIAL_NOFORWARDHTML ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
504 #define DIAL_NOCONNECTEDLINE ((uint64_t)1 << 33)
505 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 34)
506 #define OPT_PEER_H ((uint64_t)1 << 35)
507 #define OPT_CALLEE_GO_ON ((uint64_t)1 << 36)
508 #define OPT_CANCEL_TIMEOUT ((uint64_t)1 << 37)
511 OPT_ARG_ANNOUNCE = 0,
514 OPT_ARG_DURATION_LIMIT,
516 OPT_ARG_CALLEE_MACRO,
517 OPT_ARG_CALLEE_GOSUB,
518 OPT_ARG_CALLEE_GO_ON,
520 OPT_ARG_DURATION_STOP,
522 /* note: this entry _MUST_ be the last one in the enum */
526 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
527 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
528 AST_APP_OPTION('C', OPT_RESETCDR),
529 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
530 AST_APP_OPTION('d', OPT_DTMF_EXIT),
531 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
532 AST_APP_OPTION('e', OPT_PEER_H),
533 AST_APP_OPTION('f', OPT_FORCECLID),
534 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
535 AST_APP_OPTION('g', OPT_GO_ON),
536 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
537 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
538 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
539 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
540 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
541 AST_APP_OPTION('k', OPT_CALLEE_PARK),
542 AST_APP_OPTION('K', OPT_CALLER_PARK),
543 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
544 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
545 AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
546 AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
547 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
548 AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
549 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
550 AST_APP_OPTION('p', OPT_SCREENING),
551 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
552 AST_APP_OPTION('r', OPT_RINGBACK),
553 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
554 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
555 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
556 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
557 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
558 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
559 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
560 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
561 AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
564 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
565 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
566 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
567 !chan->audiohooks && !peer->audiohooks)
570 * The list of active channels
573 struct chanlist *next;
574 struct ast_channel *chan;
576 struct ast_party_connected_line connected;
579 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
581 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
583 /* Hang up a tree of stuff */
586 /* Hangup any existing lines we have open */
587 if (outgoing->chan && (outgoing->chan != exception)) {
588 if (answered_elsewhere) {
589 /* The flag is used for local channel inheritance and stuff */
590 ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
591 /* This is for the channel drivers */
592 outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
594 ast_hangup(outgoing->chan);
597 outgoing = outgoing->next;
602 #define AST_MAX_WATCHERS 256
605 * argument to handle_cause() and other functions.
608 struct ast_channel *chan;
614 static void handle_cause(int cause, struct cause_args *num)
616 struct ast_cdr *cdr = num->chan->cdr;
625 case AST_CAUSE_CONGESTION:
631 case AST_CAUSE_NO_ROUTE_DESTINATION:
632 case AST_CAUSE_UNREGISTERED:
638 case AST_CAUSE_NORMAL_CLEARING:
647 /* free the buffer if allocated, and set the pointer to the second arg */
648 #define S_REPLACE(s, new_val) \
655 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
657 char rexten[2] = { exten, '\0' };
660 if (!ast_goto_if_exists(chan, context, rexten, pri))
663 if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
665 else if (!ast_strlen_zero(chan->macrocontext)) {
666 if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
673 /* do not call with chan lock held */
674 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
679 ast_channel_lock(chan);
680 context = ast_strdupa(S_OR(chan->macrocontext, chan->context));
681 exten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
682 ast_channel_unlock(chan);
684 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
687 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
689 manager_event(EVENT_FLAG_CALL, "Dial",
690 "SubEvent: Begin\r\n"
692 "Destination: %s\r\n"
693 "CallerIDNum: %s\r\n"
694 "CallerIDName: %s\r\n"
696 "DestUniqueID: %s\r\n"
697 "Dialstring: %s\r\n",
698 src->name, dst->name, S_OR(src->cid.cid_num, "<unknown>"),
699 S_OR(src->cid.cid_name, "<unknown>"), src->uniqueid,
700 dst->uniqueid, dialstring ? dialstring : "");
703 static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
705 manager_event(EVENT_FLAG_CALL, "Dial",
709 "DialStatus: %s\r\n",
710 src->name, src->uniqueid, dialstatus);
714 * helper function for wait_for_answer()
716 * XXX this code is highly suspicious, as it essentially overwrites
717 * the outgoing channel without properly deleting it.
719 static void do_forward(struct chanlist *o,
720 struct cause_args *num, struct ast_flags64 *peerflags, int single, int *to)
723 struct ast_channel *original = o->chan;
724 struct ast_channel *c = o->chan; /* the winner */
725 struct ast_channel *in = num->chan; /* the input channel */
726 struct ast_party_redirecting *apr = &o->chan->redirecting;
727 struct ast_party_connected_line *apc = &o->chan->connected;
732 ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
733 if ((stuff = strchr(tmpchan, '/'))) {
737 const char *forward_context;
739 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
740 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
741 ast_channel_unlock(c);
745 /* Before processing channel, go ahead and check for forwarding */
746 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
747 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
748 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
749 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
751 cause = AST_CAUSE_BUSY;
753 /* Setup parameters */
754 c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
757 ast_channel_make_compatible(o->chan, in);
758 ast_channel_inherit_variables(in, o->chan);
759 ast_channel_datastore_inherit(in, o->chan);
761 ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
764 ast_clear_flag64(o, DIAL_STILLGOING);
765 handle_cause(cause, num);
766 ast_hangup(original);
769 ast_rtp_instance_early_bridge_make_compatible(c, in);
772 c->cdrflags = in->cdrflags;
774 ast_channel_set_redirecting(c, apr);
776 while (ast_channel_trylock(in)) {
777 CHANNEL_DEADLOCK_AVOIDANCE(c);
779 S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(original->cid.cid_rdnis, S_OR(in->macroexten, in->exten))));
781 c->cid.cid_tns = in->cid.cid_tns;
783 if (ast_test_flag64(o, OPT_FORCECLID)) {
784 S_REPLACE(c->cid.cid_num, ast_strdupa(S_OR(in->macroexten, in->exten)));
785 S_REPLACE(c->cid.cid_name, NULL);
786 ast_string_field_set(c, accountcode, c->accountcode);
788 ast_party_caller_copy(&c->cid, &in->cid);
789 ast_string_field_set(c, accountcode, in->accountcode);
791 ast_party_connected_line_copy(&c->connected, apc);
793 S_REPLACE(in->cid.cid_rdnis, ast_strdup(c->cid.cid_rdnis));
794 ast_channel_update_redirecting(in, apr);
796 ast_clear_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE);
797 if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
801 ast_channel_unlock(in);
802 ast_channel_unlock(c);
804 if (ast_call(c, tmpchan, 0)) {
805 ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
806 ast_clear_flag64(o, DIAL_STILLGOING);
807 ast_hangup(original);
813 while (ast_channel_trylock(in)) {
814 CHANNEL_DEADLOCK_AVOIDANCE(c);
816 senddialevent(in, c, stuff);
817 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
818 char cidname[AST_MAX_EXTENSION] = "";
819 const char *tmpexten;
820 tmpexten = ast_strdupa(S_OR(in->macroexten, in->exten));
821 ast_channel_unlock(in);
822 ast_channel_unlock(c);
823 ast_set_callerid(c, tmpexten, get_cid_name(cidname, sizeof(cidname), in), NULL);
825 ast_channel_unlock(in);
826 ast_channel_unlock(c);
828 /* Hangup the original channel now, in case we needed it */
829 ast_hangup(original);
832 ast_indicate(in, -1);
837 /* argument used for some functions. */
838 struct privacy_args {
842 char privintro[1024];
846 static struct ast_channel *wait_for_answer(struct ast_channel *in,
847 struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
848 struct privacy_args *pa,
849 const struct cause_args *num_in, int *result, char *dtmf_progress)
851 struct cause_args num = *num_in;
852 int prestart = num.busy + num.congestion + num.nochan;
854 struct ast_channel *peer = NULL;
855 /* single is set if only one destination is enabled */
856 int single = outgoing && !outgoing->next;
858 struct chanlist *epollo;
860 struct ast_party_connected_line connected_caller;
861 struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
863 /* Turn off hold music, etc */
864 if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK))
865 ast_deactivate_generator(in);
867 /* If we are calling a single channel, make them compatible for in-band tone purpose */
868 ast_channel_make_compatible(outgoing->chan, in);
870 if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_NOCONNECTEDLINE)) {
871 ast_channel_lock(outgoing->chan);
872 ast_connected_line_copy_from_caller(&connected_caller, &outgoing->chan->cid);
873 ast_channel_unlock(outgoing->chan);
874 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
875 ast_channel_update_connected_line(in, &connected_caller);
876 ast_party_connected_line_free(&connected_caller);
881 for (epollo = outgoing; epollo; epollo = epollo->next)
882 ast_poll_channel_add(in, epollo->chan);
885 while (*to && !peer) {
887 int pos = 0; /* how many channels do we handle */
888 int numlines = prestart;
889 struct ast_channel *winner;
890 struct ast_channel *watchers[AST_MAX_WATCHERS];
892 watchers[pos++] = in;
893 for (o = outgoing; o; o = o->next) {
894 /* Keep track of important channels */
895 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
896 watchers[pos++] = o->chan;
899 if (pos == 1) { /* only the input channel is available */
900 if (numlines == (num.busy + num.congestion + num.nochan)) {
901 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
903 strcpy(pa->status, "BUSY");
904 else if (num.congestion)
905 strcpy(pa->status, "CONGESTION");
907 strcpy(pa->status, "CHANUNAVAIL");
909 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
914 winner = ast_waitfor_n(watchers, pos, to);
915 for (o = outgoing; o; o = o->next) {
917 struct ast_channel *c = o->chan;
921 if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
923 ast_verb(3, "%s answered %s\n", c->name, in->name);
924 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
925 if (o->connected.id.number) {
926 ast_channel_update_connected_line(in, &o->connected);
927 } else if (!ast_test_flag64(o, DIAL_NOCONNECTEDLINE)) {
929 ast_connected_line_copy_from_caller(&connected_caller, &c->cid);
930 ast_channel_unlock(c);
931 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
932 ast_channel_update_connected_line(in, &connected_caller);
933 ast_party_connected_line_free(&connected_caller);
937 ast_copy_flags64(peerflags, o,
938 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
939 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
940 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
941 OPT_CALLEE_PARK | OPT_CALLER_PARK |
942 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
944 ast_string_field_set(c, dialcontext, "");
945 ast_copy_string(c->exten, "", sizeof(c->exten));
951 /* here, o->chan == c == winner */
952 if (!ast_strlen_zero(c->call_forward)) {
953 do_forward(o, &num, peerflags, single, to);
956 f = ast_read(winner);
958 in->hangupcause = c->hangupcause;
960 ast_poll_channel_del(in, c);
964 ast_clear_flag64(o, DIAL_STILLGOING);
965 handle_cause(in->hangupcause, &num);
968 if (f->frametype == AST_FRAME_CONTROL) {
969 switch(f->subclass) {
970 case AST_CONTROL_ANSWER:
971 /* This is our guy if someone answered. */
973 ast_verb(3, "%s answered %s\n", c->name, in->name);
974 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
975 if (o->connected.id.number) {
976 ast_channel_update_connected_line(in, &o->connected);
977 } else if (!ast_test_flag64(o, DIAL_NOCONNECTEDLINE)) {
979 ast_connected_line_copy_from_caller(&connected_caller, &c->cid);
980 ast_channel_unlock(c);
981 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
982 ast_channel_update_connected_line(in, &connected_caller);
983 ast_party_connected_line_free(&connected_caller);
988 peer->cdr->answer = ast_tvnow();
989 peer->cdr->disposition = AST_CDR_ANSWERED;
991 ast_copy_flags64(peerflags, o,
992 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
993 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
994 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
995 OPT_CALLEE_PARK | OPT_CALLER_PARK |
996 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
998 ast_string_field_set(c, dialcontext, "");
999 ast_copy_string(c->exten, "", sizeof(c->exten));
1000 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
1001 /* Setup early bridge if appropriate */
1002 ast_channel_early_bridge(in, peer);
1004 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
1005 in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1006 c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1008 case AST_CONTROL_BUSY:
1009 ast_verb(3, "%s is busy\n", c->name);
1010 in->hangupcause = c->hangupcause;
1013 ast_clear_flag64(o, DIAL_STILLGOING);
1014 handle_cause(AST_CAUSE_BUSY, &num);
1016 case AST_CONTROL_CONGESTION:
1017 ast_verb(3, "%s is circuit-busy\n", c->name);
1018 in->hangupcause = c->hangupcause;
1021 ast_clear_flag64(o, DIAL_STILLGOING);
1022 handle_cause(AST_CAUSE_CONGESTION, &num);
1024 case AST_CONTROL_RINGING:
1025 ast_verb(3, "%s is ringing\n", c->name);
1026 /* Setup early media if appropriate */
1027 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1028 ast_channel_early_bridge(in, c);
1029 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
1030 ast_indicate(in, AST_CONTROL_RINGING);
1034 case AST_CONTROL_PROGRESS:
1035 ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
1036 /* Setup early media if appropriate */
1037 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1038 ast_channel_early_bridge(in, c);
1039 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1040 ast_indicate(in, AST_CONTROL_PROGRESS);
1041 if(!ast_strlen_zero(dtmf_progress)) {
1042 ast_verb(3, "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n", dtmf_progress);
1043 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1046 case AST_CONTROL_VIDUPDATE:
1047 ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
1048 ast_indicate(in, AST_CONTROL_VIDUPDATE);
1050 case AST_CONTROL_SRCUPDATE:
1051 ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
1052 ast_indicate(in, AST_CONTROL_SRCUPDATE);
1054 case AST_CONTROL_CONNECTED_LINE:
1055 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1056 ast_verb(3, "Connected line update to %s prevented.\n", in->name);
1057 } else if (!single) {
1058 struct ast_party_connected_line connected;
1059 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n", c->name, in->name);
1060 ast_party_connected_line_set_init(&connected, &o->connected);
1061 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1062 ast_party_connected_line_set(&o->connected, &connected);
1063 ast_party_connected_line_free(&connected);
1065 ast_verb(3, "%s connected line has changed, passing it to %s\n", c->name, in->name);
1066 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1069 case AST_CONTROL_REDIRECTING:
1070 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1071 ast_verb(3, "Redirecting update to %s prevented.\n", in->name);
1073 ast_verb(3, "%s redirecting info has changed, passing it to %s\n", c->name, in->name);
1074 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1077 case AST_CONTROL_PROCEEDING:
1078 ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
1079 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1080 ast_channel_early_bridge(in, c);
1081 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1082 ast_indicate(in, AST_CONTROL_PROCEEDING);
1084 case AST_CONTROL_HOLD:
1085 ast_verb(3, "Call on %s placed on hold\n", c->name);
1086 ast_indicate(in, AST_CONTROL_HOLD);
1088 case AST_CONTROL_UNHOLD:
1089 ast_verb(3, "Call on %s left from hold\n", c->name);
1090 ast_indicate(in, AST_CONTROL_UNHOLD);
1092 case AST_CONTROL_OFFHOOK:
1093 case AST_CONTROL_FLASH:
1094 /* Ignore going off hook and flash */
1097 if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
1098 ast_verb(3, "%s stopped sounds\n", c->name);
1099 ast_indicate(in, -1);
1100 pa->sentringing = 0;
1104 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
1106 } else if (single) {
1107 switch (f->frametype) {
1108 case AST_FRAME_VOICE:
1109 case AST_FRAME_IMAGE:
1110 case AST_FRAME_TEXT:
1111 if (ast_write(in, f)) {
1112 ast_log(LOG_WARNING, "Unable to write frame\n");
1115 case AST_FRAME_HTML:
1116 if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass, f->data.ptr, f->datalen) == -1) {
1117 ast_log(LOG_WARNING, "Unable to send URL\n");
1127 struct ast_frame *f = ast_read(in);
1129 if (f && (f->frametype != AST_FRAME_VOICE))
1130 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1131 else if (!f || (f->frametype != AST_FRAME_VOICE))
1132 printf("Hangup received on %s\n", in->name);
1134 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
1137 strcpy(pa->status, "CANCEL");
1138 ast_cdr_noanswer(in->cdr);
1140 if (f->data.uint32) {
1141 in->hangupcause = f->data.uint32;
1148 /* now f is guaranteed non-NULL */
1149 if (f->frametype == AST_FRAME_DTMF) {
1150 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1151 const char *context;
1152 ast_channel_lock(in);
1153 context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1154 if (onedigit_goto(in, context, (char) f->subclass, 1)) {
1155 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
1157 ast_cdr_noanswer(in->cdr);
1158 *result = f->subclass;
1159 strcpy(pa->status, "CANCEL");
1161 ast_channel_unlock(in);
1164 ast_channel_unlock(in);
1167 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1168 detect_disconnect(in, f->subclass, featurecode)) {
1169 ast_verb(3, "User requested call disconnect.\n");
1171 strcpy(pa->status, "CANCEL");
1172 ast_cdr_noanswer(in->cdr);
1178 /* Forward HTML stuff */
1179 if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
1180 if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data.ptr, f->datalen) == -1)
1181 ast_log(LOG_WARNING, "Unable to send URL\n");
1183 if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
1184 if (ast_write(outgoing->chan, f))
1185 ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
1187 if (single && (f->frametype == AST_FRAME_CONTROL) &&
1188 ((f->subclass == AST_CONTROL_HOLD) ||
1189 (f->subclass == AST_CONTROL_UNHOLD) ||
1190 (f->subclass == AST_CONTROL_VIDUPDATE) ||
1191 (f->subclass == AST_CONTROL_SRCUPDATE) ||
1192 (f->subclass == AST_CONTROL_CONNECTED_LINE) ||
1193 (f->subclass == AST_CONTROL_REDIRECTING))) {
1194 ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
1195 ast_indicate_data(outgoing->chan, f->subclass, f->data.ptr, f->datalen);
1200 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1201 if (!*to || ast_check_hangup(in))
1202 ast_cdr_noanswer(in->cdr);
1206 for (epollo = outgoing; epollo; epollo = epollo->next) {
1208 ast_poll_channel_del(in, epollo->chan);
1215 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
1217 struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
1218 struct ast_call_feature feature = { 0, };
1221 ast_str_append(&featurecode, 1, "%c", code);
1223 res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
1225 if (res != AST_FEATURE_RETURN_STOREDIGITS) {
1226 ast_str_reset(featurecode);
1228 if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
1235 static void replace_macro_delimiter(char *s)
1242 /* returns true if there is a valid privacy reply */
1243 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1247 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1249 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1254 static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
1255 char *parse, struct timeval *calldurationlimit)
1257 char *stringp = ast_strdupa(parse);
1258 char *limit_str, *warning_str, *warnfreq_str;
1260 int play_to_caller = 0, play_to_callee = 0;
1263 limit_str = strsep(&stringp, ":");
1264 warning_str = strsep(&stringp, ":");
1265 warnfreq_str = strsep(&stringp, ":");
1267 config->timelimit = atol(limit_str);
1269 config->play_warning = atol(warning_str);
1271 config->warning_freq = atol(warnfreq_str);
1273 if (!config->timelimit) {
1274 ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
1275 config->timelimit = config->play_warning = config->warning_freq = 0;
1276 config->warning_sound = NULL;
1277 return -1; /* error */
1278 } else if ( (delta = config->play_warning - config->timelimit) > 0) {
1279 int w = config->warning_freq;
1281 /* If the first warning is requested _after_ the entire call would end,
1282 and no warning frequency is requested, then turn off the warning. If
1283 a warning frequency is requested, reduce the 'first warning' time by
1284 that frequency until it falls within the call's total time limit.
1286 timelim->| delta |<-playwarning
1287 0__________________|_________________|
1290 so the number of intervals to cut is 1+(delta-1)/w
1294 config->play_warning = 0;
1296 config->play_warning -= w * ( 1 + (delta-1)/w );
1297 if (config->play_warning < 1)
1298 config->play_warning = config->warning_freq = 0;
1302 ast_channel_lock(chan);
1304 var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
1306 play_to_caller = var ? ast_true(var) : 1;
1308 var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
1309 play_to_callee = var ? ast_true(var) : 0;
1311 if (!play_to_caller && !play_to_callee)
1314 var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
1315 config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
1317 /* The code looking at config wants a NULL, not just "", to decide
1318 * that the message should not be played, so we replace "" with NULL.
1319 * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
1323 var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
1324 config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1326 var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
1327 config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1329 ast_channel_unlock(chan);
1331 /* undo effect of S(x) in case they are both used */
1332 calldurationlimit->tv_sec = 0;
1333 calldurationlimit->tv_usec = 0;
1335 /* more efficient to do it like S(x) does since no advanced opts */
1336 if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
1337 calldurationlimit->tv_sec = config->timelimit / 1000;
1338 calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
1339 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
1340 calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
1341 config->timelimit = play_to_caller = play_to_callee =
1342 config->play_warning = config->warning_freq = 0;
1344 ast_verb(3, "Limit Data for this call:\n");
1345 ast_verb(4, "timelimit = %ld\n", config->timelimit);
1346 ast_verb(4, "play_warning = %ld\n", config->play_warning);
1347 ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
1348 ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
1349 ast_verb(4, "warning_freq = %ld\n", config->warning_freq);
1350 ast_verb(4, "start_sound = %s\n", S_OR(config->start_sound, ""));
1351 ast_verb(4, "warning_sound = %s\n", config->warning_sound);
1352 ast_verb(4, "end_sound = %s\n", S_OR(config->end_sound, ""));
1355 ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
1357 ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
1361 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1362 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1368 /* Get the user's intro, store it in priv-callerintros/$CID,
1369 unless it is already there-- this should be done before the
1370 call is actually dialed */
1372 /* all ring indications and moh for the caller has been halted as soon as the
1373 target extension was picked up. We are going to have to kill some
1374 time and make the caller believe the peer hasn't picked up yet */
1376 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1377 char *original_moh = ast_strdupa(chan->musicclass);
1378 ast_indicate(chan, -1);
1379 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1380 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1381 ast_string_field_set(chan, musicclass, original_moh);
1382 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1383 ast_indicate(chan, AST_CONTROL_RINGING);
1387 /* Start autoservice on the other chan ?? */
1388 res2 = ast_autoservice_start(chan);
1389 /* Now Stream the File */
1390 for (loopcount = 0; loopcount < 3; loopcount++) {
1391 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1393 if (!res2) /* on timeout, play the message again */
1394 res2 = ast_play_and_wait(peer, "priv-callpending");
1395 if (!valid_priv_reply(opts, res2))
1397 /* priv-callpending script:
1398 "I have a caller waiting, who introduces themselves as:"
1401 res2 = ast_play_and_wait(peer, pa->privintro);
1402 if (!valid_priv_reply(opts, res2))
1404 /* now get input from the called party, as to their choice */
1406 /* XXX can we have both, or they are mutually exclusive ? */
1407 if (ast_test_flag64(opts, OPT_PRIVACY))
1408 res2 = ast_play_and_wait(peer, "priv-callee-options");
1409 if (ast_test_flag64(opts, OPT_SCREENING))
1410 res2 = ast_play_and_wait(peer, "screen-callee-options");
1412 /*! \page DialPrivacy Dial Privacy scripts
1413 \par priv-callee-options script:
1414 "Dial 1 if you wish this caller to reach you directly in the future,
1415 and immediately connect to their incoming call
1416 Dial 2 if you wish to send this caller to voicemail now and
1418 Dial 3 to send this caller to the torture menus, now and forevermore.
1419 Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1420 Dial 5 to allow this caller to come straight thru to you in the future,
1421 but right now, just this once, send them to voicemail."
1422 \par screen-callee-options script:
1423 "Dial 1 if you wish to immediately connect to the incoming call
1424 Dial 2 if you wish to send this caller to voicemail.
1425 Dial 3 to send this caller to the torture menus.
1426 Dial 4 to send this caller to a simple "go away" menu.
1428 if (valid_priv_reply(opts, res2))
1430 /* invalid option */
1431 res2 = ast_play_and_wait(peer, "vm-sorry");
1434 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1436 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1437 ast_indicate(chan, -1);
1438 pa->sentringing = 0;
1440 ast_autoservice_stop(chan);
1441 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1442 /* map keypresses to various things, the index is res2 - '1' */
1443 static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1444 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1446 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1447 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1448 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1454 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1457 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1460 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1463 /* XXX should we set status to DENY ? */
1464 if (ast_test_flag64(opts, OPT_PRIVACY))
1466 /* if not privacy, then 5 is the same as "default" case */
1467 default: /* bad input or -1 if failure to start autoservice */
1468 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
1469 /* well, there seems basically two choices. Just patch the caller thru immediately,
1470 or,... put 'em thru to voicemail. */
1471 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1472 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1473 /* XXX should we set status to DENY ? */
1474 /* XXX what about the privacy flags ? */
1478 if (res2 == '1') { /* the only case where we actually connect */
1479 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1480 just clog things up, and it's not useful information, not being tied to a CID */
1481 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1482 ast_filedelete(pa->privintro, NULL);
1483 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1484 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1486 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1488 return 0; /* the good exit path */
1490 ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
1495 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1496 static int setup_privacy_args(struct privacy_args *pa,
1497 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1502 int silencethreshold;
1504 if (!ast_strlen_zero(chan->cid.cid_num)) {
1505 l = ast_strdupa(chan->cid.cid_num);
1506 ast_shrink_phone_number(l);
1507 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1508 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1509 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1511 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1512 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1517 tnam = ast_strdupa(chan->name);
1518 /* clean the channel name so slashes don't try to end up in disk file name */
1519 for (tn2 = tnam; *tn2; tn2++) {
1520 if (*tn2 == '/') /* any other chars to be afraid of? */
1523 ast_verb(3, "Privacy-- callerid is empty\n");
1525 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
1527 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1530 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1532 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1533 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1534 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1535 pa->privdb_val = AST_PRIVACY_ALLOW;
1536 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1537 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1540 if (pa->privdb_val == AST_PRIVACY_DENY) {
1541 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1542 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1544 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1545 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1546 return 0; /* Is this right? */
1547 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1548 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1549 return 0; /* is this right??? */
1550 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1551 /* Get the user's intro, store it in priv-callerintros/$CID,
1552 unless it is already there-- this should be done before the
1553 call is actually dialed */
1555 /* make sure the priv-callerintros dir actually exists */
1556 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1557 if ((res = ast_mkdir(pa->privintro, 0755))) {
1558 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1562 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1563 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1564 /* the DELUX version of this code would allow this caller the
1565 option to hear and retape their previously recorded intro.
1568 int duration; /* for feedback from play_and_wait */
1569 /* the file doesn't exist yet. Let the caller submit his
1570 vocal intro for posterity */
1571 /* priv-recordintro script:
1573 "At the tone, please say your name:"
1576 silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1578 res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "gsm", &duration, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
1579 /* don't think we'll need a lock removed, we took care of
1580 conflicts by naming the pa.privintro file */
1582 /* Delete the file regardless since they hung up during recording */
1583 ast_filedelete(pa->privintro, NULL);
1584 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1585 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1587 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1590 if (!ast_streamfile(chan, "vm-dialout", chan->language) )
1591 ast_waitstream(chan, "");
1594 return 1; /* success */
1597 static void end_bridge_callback(void *data)
1601 struct ast_channel *chan = data;
1609 ast_channel_lock(chan);
1610 if (chan->cdr->answer.tv_sec) {
1611 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
1612 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1615 if (chan->cdr->start.tv_sec) {
1616 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
1617 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1619 ast_channel_unlock(chan);
1622 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
1623 bconfig->end_bridge_callback_data = originator;
1626 static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
1628 int res = -1; /* default: error */
1629 char *rest, *cur; /* scan the list of destinations */
1630 struct chanlist *outgoing = NULL; /* list of destinations */
1631 struct ast_channel *peer;
1632 int to; /* timeout */
1633 struct cause_args num = { chan, 0, 0, 0 };
1636 char cidname[AST_MAX_EXTENSION] = "";
1638 struct ast_bridge_config config = { { 0, } };
1639 struct timeval calldurationlimit = { 0, };
1640 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
1641 struct privacy_args pa = {
1644 .status = "INVALIDARGS",
1646 int sentringing = 0, moh = 0;
1647 const char *outbound_group = NULL;
1651 AST_DECLARE_APP_ARGS(args,
1653 AST_APP_ARG(timeout);
1654 AST_APP_ARG(options);
1657 struct ast_flags64 opts = { 0, };
1658 char *opt_args[OPT_ARG_ARRAY_SIZE];
1659 struct ast_datastore *datastore = NULL;
1660 int fulldial = 0, num_dialed = 0;
1662 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
1663 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
1664 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
1665 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
1666 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
1667 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
1669 if (ast_strlen_zero(data)) {
1670 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1671 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1675 parse = ast_strdupa(data);
1677 AST_STANDARD_APP_ARGS(args, parse);
1679 if (!ast_strlen_zero(args.options) &&
1680 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
1681 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1685 if (ast_strlen_zero(args.peers)) {
1686 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1687 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1691 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
1692 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
1693 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
1696 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
1697 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
1698 if (!calldurationlimit.tv_sec) {
1699 ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
1700 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1703 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
1706 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
1707 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
1708 dtmfcalled = strsep(&dtmf_progress, ":");
1709 dtmfcalling = strsep(&dtmf_progress, ":");
1712 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
1713 if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
1717 if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
1718 ast_cdr_reset(chan->cdr, NULL);
1719 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
1720 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
1722 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
1723 res = setup_privacy_args(&pa, &opts, opt_args, chan);
1726 res = -1; /* reset default */
1729 if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
1730 __ast_answer(chan, 0, 0);
1736 /* If a channel group has been specified, get it for use when we create peer channels */
1738 ast_channel_lock(chan);
1739 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
1740 outbound_group = ast_strdupa(outbound_group);
1741 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
1742 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
1743 outbound_group = ast_strdupa(outbound_group);
1745 ast_channel_unlock(chan);
1746 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE | OPT_CANCEL_TIMEOUT);
1748 /* loop through the list of dial destinations */
1750 while ((cur = strsep(&rest, "&")) ) {
1751 struct chanlist *tmp;
1752 struct ast_channel *tc; /* channel for this destination */
1753 /* Get a technology/[device:]number pair */
1755 char *interface = ast_strdupa(number);
1756 char *tech = strsep(&number, "/");
1757 /* find if we already dialed this interface */
1758 struct ast_dialed_interface *di;
1759 AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
1762 ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
1765 if (!(tmp = ast_calloc(1, sizeof(*tmp))))
1768 ast_copy_flags64(tmp, &opts,
1769 OPT_CANCEL_ELSEWHERE |
1770 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1771 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1772 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1773 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1774 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1775 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
1776 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
1778 ast_copy_string(numsubst, number, sizeof(numsubst));
1779 /* Request the peer */
1781 ast_channel_lock(chan);
1782 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
1783 /* If the incoming channel has previously had connected line information
1784 * set on it (perhaps through the CONNECTED_LINE dialplan function) then
1785 * seed the calllist's connected line information with this previously
1788 if (chan->connected.id.number) {
1789 ast_party_connected_line_copy(&tmp->connected, &chan->connected);
1791 ast_channel_unlock(chan);
1794 dialed_interfaces = datastore->data;
1796 if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
1797 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
1802 datastore->inheritance = DATASTORE_INHERIT_FOREVER;
1804 if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
1809 datastore->data = dialed_interfaces;
1810 AST_LIST_HEAD_INIT(dialed_interfaces);
1812 ast_channel_lock(chan);
1813 ast_channel_datastore_add(chan, datastore);
1814 ast_channel_unlock(chan);
1817 AST_LIST_LOCK(dialed_interfaces);
1818 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
1819 if (!strcasecmp(di->interface, interface)) {
1820 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
1825 AST_LIST_UNLOCK(dialed_interfaces);
1833 /* It is always ok to dial a Local interface. We only keep track of
1834 * which "real" interfaces have been dialed. The Local channel will
1835 * inherit this list so that if it ends up dialing a real interface,
1836 * it won't call one that has already been called. */
1837 if (strcasecmp(tech, "Local")) {
1838 if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
1839 AST_LIST_UNLOCK(dialed_interfaces);
1843 strcpy(di->interface, interface);
1845 AST_LIST_LOCK(dialed_interfaces);
1846 AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
1847 AST_LIST_UNLOCK(dialed_interfaces);
1850 tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
1852 /* If we can't, just go on to the next call */
1853 ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
1854 tech, cause, ast_cause2str(cause));
1855 handle_cause(cause, &num);
1856 if (!rest) /* we are on the last destination */
1857 chan->hangupcause = cause;
1861 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
1863 ast_channel_lock(tc);
1864 while (ast_channel_trylock(chan)) {
1865 CHANNEL_DEADLOCK_AVOIDANCE(tc);
1867 /* Setup outgoing SDP to match incoming one */
1868 if (!outgoing && !rest) {
1869 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
1872 /* Inherit specially named variables from parent channel */
1873 ast_channel_inherit_variables(chan, tc);
1874 ast_channel_datastore_inherit(chan, tc);
1876 tc->appl = "AppDial";
1877 tc->data = "(Outgoing Line)";
1878 memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
1880 /* If the new channel has no callerid, try to guess what it should be */
1881 if (ast_strlen_zero(tc->cid.cid_num)) {
1882 if (!ast_strlen_zero(chan->connected.id.number)) {
1883 ast_set_callerid(tc, chan->connected.id.number, chan->connected.id.name, chan->connected.ani);
1884 } else if (!ast_strlen_zero(chan->cid.cid_dnid)) {
1885 ast_set_callerid(tc, chan->cid.cid_dnid, NULL, NULL);
1886 } else if (!ast_strlen_zero(S_OR(chan->macroexten, chan->exten))) {
1887 ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), NULL, NULL);
1889 ast_set_flag64(tmp, DIAL_NOCONNECTEDLINE);
1892 ast_connected_line_copy_from_caller(&tc->connected, &chan->cid);
1894 S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
1895 ast_party_redirecting_copy(&tc->redirecting, &chan->redirecting);
1897 tc->cid.cid_tns = chan->cid.cid_tns;
1899 ast_string_field_set(tc, accountcode, chan->accountcode);
1900 tc->cdrflags = chan->cdrflags;
1901 if (ast_strlen_zero(tc->musicclass))
1902 ast_string_field_set(tc, musicclass, chan->musicclass);
1904 /* Pass ADSI CPE and transfer capability */
1905 tc->adsicpe = chan->adsicpe;
1906 tc->transfercapability = chan->transfercapability;
1908 /* If we have an outbound group, set this peer channel to it */
1910 ast_app_group_set_channel(tc, outbound_group);
1911 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
1912 if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
1913 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1915 /* Check if we're forced by configuration */
1916 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
1917 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1920 /* Inherit context and extension */
1921 ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
1922 if (!ast_strlen_zero(chan->macroexten))
1923 ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
1925 ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
1927 ast_channel_unlock(tc);
1928 res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
1930 /* Save the info in cdr's that we called them */
1932 ast_cdr_setdestchan(chan->cdr, tc->name);
1934 /* check the results of ast_call */
1936 /* Again, keep going even if there's an error */
1937 ast_debug(1, "ast call on peer returned %d\n", res);
1938 ast_verb(3, "Couldn't call %s\n", numsubst);
1939 if (tc->hangupcause) {
1940 chan->hangupcause = tc->hangupcause;
1942 ast_channel_unlock(chan);
1948 const char *tmpexten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
1949 senddialevent(chan, tc, numsubst);
1950 ast_verb(3, "Called %s\n", numsubst);
1951 ast_channel_unlock(chan);
1952 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1953 ast_set_callerid(tc, tmpexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
1956 /* Put them in the list of outgoing thingies... We're ready now.
1957 XXX If we're forcibly removed, these outgoing calls won't get
1959 ast_set_flag64(tmp, DIAL_STILLGOING);
1961 tmp->next = outgoing;
1963 /* If this line is up, don't try anybody else */
1964 if (outgoing->chan->_state == AST_STATE_UP)
1968 if (ast_strlen_zero(args.timeout)) {
1971 to = atoi(args.timeout);
1975 ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
1981 strcpy(pa.status, "CHANUNAVAIL");
1982 if (fulldial == num_dialed) {
1987 /* Our status will at least be NOANSWER */
1988 strcpy(pa.status, "NOANSWER");
1989 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
1991 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1992 char *original_moh = ast_strdupa(chan->musicclass);
1993 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1994 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1995 ast_string_field_set(chan, musicclass, original_moh);
1997 ast_moh_start(chan, NULL, NULL);
1999 ast_indicate(chan, AST_CONTROL_PROGRESS);
2000 } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
2001 ast_indicate(chan, AST_CONTROL_RINGING);
2006 peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result, dtmf_progress);
2008 /* The ast_channel_datastore_remove() function could fail here if the
2009 * datastore was moved to another channel during a masquerade. If this is
2010 * the case, don't free the datastore here because later, when the channel
2011 * to which the datastore was moved hangs up, it will attempt to free this
2012 * datastore again, causing a crash
2014 if (!ast_channel_datastore_remove(chan, datastore))
2015 ast_datastore_free(datastore);
2019 } else if (to) { /* Musta gotten hung up */
2021 } else { /* Nobody answered, next please? */
2025 /* SIP, in particular, sends back this error code to indicate an
2026 * overlap dialled number needs more digits. */
2027 if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
2028 res = AST_PBX_INCOMPLETE;
2031 /* almost done, although the 'else' block is 400 lines */
2035 strcpy(pa.status, "ANSWER");
2036 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2037 /* Ah ha! Someone answered within the desired timeframe. Of course after this
2038 we will always return with -1 so that it is hung up properly after the
2040 hanguptree(outgoing, peer, 1);
2042 /* If appropriate, log that we have a destination channel */
2044 ast_cdr_setdestchan(chan->cdr, peer->name);
2046 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
2048 ast_channel_lock(peer);
2049 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
2052 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
2053 ast_channel_unlock(peer);
2055 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
2056 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
2057 ast_channel_sendurl( peer, args.url );
2059 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
2060 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
2065 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
2069 /* Start autoservice on the other chan */
2070 res = ast_autoservice_start(chan);
2071 /* Now Stream the File */
2073 res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
2075 digit = ast_waitstream(peer, AST_DIGIT_ANY);
2077 /* Ok, done. stop autoservice */
2078 res = ast_autoservice_stop(chan);
2079 if (digit > 0 && !res)
2080 res = ast_senddigit(chan, digit, 0);
2086 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
2087 replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
2088 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
2089 /* peer goes to the same context and extension as chan, so just copy info from chan*/
2090 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2091 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2092 peer->priority = chan->priority + 2;
2093 ast_pbx_start(peer);
2094 hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
2101 if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
2102 struct ast_app *theapp;
2103 const char *macro_result;
2105 res = ast_autoservice_start(chan);
2107 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2111 theapp = pbx_findapp("Macro");
2113 if (theapp && !res) { /* XXX why check res here ? */
2114 /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
2115 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2116 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2118 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
2119 res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
2120 ast_debug(1, "Macro exited with status %d\n", res);
2123 ast_log(LOG_ERROR, "Could not find application Macro\n");
2127 if (ast_autoservice_stop(chan) < 0) {
2128 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2132 ast_channel_lock(peer);
2134 if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
2135 char *macro_transfer_dest;
2137 if (!strcasecmp(macro_result, "BUSY")) {
2138 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2139 ast_set_flag64(peerflags, OPT_GO_ON);
2141 } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
2142 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2143 ast_set_flag64(peerflags, OPT_GO_ON);
2145 } else if (!strcasecmp(macro_result, "CONTINUE")) {
2146 /* hangup peer and keep chan alive assuming the macro has changed
2147 the context / exten / priority or perhaps
2148 the next priority in the current exten is desired.
2150 ast_set_flag64(peerflags, OPT_GO_ON);
2152 } else if (!strcasecmp(macro_result, "ABORT")) {
2153 /* Hangup both ends unless the caller has the g flag */
2155 } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
2157 /* perform a transfer to a new extension */
2158 if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
2159 replace_macro_delimiter(macro_transfer_dest);
2160 if (!ast_parseable_goto(chan, macro_transfer_dest))
2161 ast_set_flag64(peerflags, OPT_GO_ON);
2166 ast_channel_unlock(peer);
2169 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
2170 struct ast_app *theapp;
2171 const char *gosub_result;
2172 char *gosub_args, *gosub_argstart;
2175 res9 = ast_autoservice_start(chan);
2177 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2181 theapp = pbx_findapp("Gosub");
2183 if (theapp && !res9) {
2184 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
2186 /* Set where we came from */
2187 ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
2188 ast_copy_string(peer->exten, "s", sizeof(peer->exten));
2191 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
2192 if (gosub_argstart) {
2193 *gosub_argstart = 0;
2194 if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
2195 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2198 *gosub_argstart = ',';
2200 if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
2201 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2207 res9 = pbx_exec(peer, theapp, gosub_args);
2209 struct ast_pbx_args args;
2210 /* A struct initializer fails to compile for this case ... */
2211 memset(&args, 0, sizeof(args));
2212 args.no_hangup_chan = 1;
2213 ast_pbx_run_args(peer, &args);
2215 ast_free(gosub_args);
2216 ast_debug(1, "Gosub exited with status %d\n", res9);
2218 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
2222 ast_log(LOG_ERROR, "Could not find application Gosub\n");
2226 if (ast_autoservice_stop(chan) < 0) {
2227 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2231 ast_channel_lock(peer);
2233 if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
2234 char *gosub_transfer_dest;
2236 if (!strcasecmp(gosub_result, "BUSY")) {
2237 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2238 ast_set_flag64(peerflags, OPT_GO_ON);
2240 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
2241 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2242 ast_set_flag64(peerflags, OPT_GO_ON);
2244 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
2245 /* hangup peer and keep chan alive assuming the macro has changed
2246 the context / exten / priority or perhaps
2247 the next priority in the current exten is desired.
2249 ast_set_flag64(peerflags, OPT_GO_ON);
2251 } else if (!strcasecmp(gosub_result, "ABORT")) {
2252 /* Hangup both ends unless the caller has the g flag */
2254 } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
2256 /* perform a transfer to a new extension */
2257 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
2258 replace_macro_delimiter(gosub_transfer_dest);
2259 if (!ast_parseable_goto(chan, gosub_transfer_dest))
2260 ast_set_flag64(peerflags, OPT_GO_ON);
2265 ast_channel_unlock(peer);
2269 if (!ast_tvzero(calldurationlimit)) {
2270 struct timeval whentohangup = calldurationlimit;
2271 peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
2273 if (!ast_strlen_zero(dtmfcalled)) {
2274 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
2275 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
2277 if (!ast_strlen_zero(dtmfcalling)) {
2278 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
2279 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
2283 if (res) { /* some error */
2286 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
2287 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
2288 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
2289 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
2290 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
2291 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
2292 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
2293 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
2294 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
2295 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
2296 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
2297 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
2298 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
2299 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
2300 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
2301 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
2302 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
2303 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
2304 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
2305 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
2306 if (ast_test_flag64(peerflags, OPT_GO_ON))
2307 ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
2309 config.end_bridge_callback = end_bridge_callback;
2310 config.end_bridge_callback_data = chan;
2311 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
2316 } else if (sentringing) {
2318 ast_indicate(chan, -1);
2320 /* Be sure no generators are left on it */
2321 ast_deactivate_generator(chan);
2322 /* Make sure channels are compatible */
2323 res = ast_channel_make_compatible(chan, peer);
2325 ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
2331 struct oprmode oprmode;
2333 oprmode.peer = peer;
2334 oprmode.mode = opermode;
2336 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
2338 res = ast_bridge_call(chan, peer, &config);
2341 strcpy(peer->context, chan->context);
2343 if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
2348 strcpy(peer->exten, "h");
2350 autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
2351 ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
2353 while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
2356 if (found && res9) {
2357 /* Something bad happened, or a hangup has been requested. */
2358 ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2359 ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2361 ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP); /* set it back the way it was */
2363 if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
2364 if(!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
2365 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
2366 ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
2369 res = ast_goto_if_exists(peer, chan->context, chan->exten, (chan->priority) + 1);
2370 if (res == AST_PBX_GOTO_FAILED) {
2375 ast_pbx_start(peer);
2377 if (!ast_check_hangup(chan))
2378 chan->hangupcause = peer->hangupcause;
2386 } else if (sentringing) {
2388 ast_indicate(chan, -1);
2390 ast_channel_early_bridge(chan, NULL);
2391 hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
2392 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2393 senddialendevent(chan, pa.status);
2394 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
2396 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
2397 if (!ast_tvzero(calldurationlimit))
2398 memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
2403 if (config.warning_sound) {
2404 ast_free((char *)config.warning_sound);
2406 if (config.end_sound) {
2407 ast_free((char *)config.end_sound);
2409 if (config.start_sound) {
2410 ast_free((char *)config.start_sound);
2415 static int dial_exec(struct ast_channel *chan, void *data)
2417 struct ast_flags64 peerflags;
2419 memset(&peerflags, 0, sizeof(peerflags));
2421 return dial_exec_full(chan, data, &peerflags, NULL);
2424 static int retrydial_exec(struct ast_channel *chan, void *data)
2427 const char *context = NULL;
2428 int sleepms = 0, loops = 0, res = -1;
2429 struct ast_flags64 peerflags = { 0, };
2430 AST_DECLARE_APP_ARGS(args,
2431 AST_APP_ARG(announce);
2433 AST_APP_ARG(retries);
2434 AST_APP_ARG(dialdata);
2437 if (ast_strlen_zero(data)) {
2438 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
2442 parse = ast_strdupa(data);
2443 AST_STANDARD_APP_ARGS(args, parse);
2445 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
2448 if (!ast_strlen_zero(args.retries)) {
2449 loops = atoi(args.retries);
2452 if (!args.dialdata) {
2453 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
2461 loops = -1; /* run forever */
2463 ast_channel_lock(chan);
2464 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
2465 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
2466 ast_channel_unlock(chan);
2472 chan->data = "Retrying";
2473 if (ast_test_flag(chan, AST_FLAG_MOH))
2476 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
2481 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
2482 if (!ast_strlen_zero(args.announce)) {
2483 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2484 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2485 ast_waitstream(chan, AST_DIGIT_ANY);
2487 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2489 if (!res && sleepms) {
2490 if (!ast_test_flag(chan, AST_FLAG_MOH))
2491 ast_moh_start(chan, NULL, NULL);
2492 res = ast_waitfordigit(chan, sleepms);
2495 if (!ast_strlen_zero(args.announce)) {
2496 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2497 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2498 res = ast_waitstream(chan, "");
2500 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2503 if (!ast_test_flag(chan, AST_FLAG_MOH))
2504 ast_moh_start(chan, NULL, NULL);
2506 res = ast_waitfordigit(chan, sleepms);
2511 if (res < 0 || res == AST_PBX_INCOMPLETE) {
2513 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
2514 if (onedigit_goto(chan, context, (char) res, 1)) {
2526 if (ast_test_flag(chan, AST_FLAG_MOH))
2532 static int unload_module(void)
2535 struct ast_context *con;
2537 res = ast_unregister_application(app);
2538 res |= ast_unregister_application(rapp);
2540 if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
2541 ast_context_remove_extension2(con, "s", 1, NULL, 0);
2542 ast_context_destroy(con, "app_dial"); /* leave nothing behind */
2548 static int load_module(void)
2551 struct ast_context *con;
2553 con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
2555 ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
2557 ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
2559 res = ast_register_application_xml(app, dial_exec);
2560 res |= ast_register_application_xml(rapp, retrydial_exec);
2565 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");
projects / asterisk/asterisk.git / blob