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>
375 <parameter name="URL">
376 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
380 <para>This application will place calls to one or more specified channels. As soon
381 as one of the requested channels answers, the originating channel will be
382 answered, if it has not already been answered. These two channels will then
383 be active in a bridged call. All other channels that were requested will then
386 <para>Unless there is a timeout specified, the Dial application will wait
387 indefinitely until one of the called channels answers, the user hangs up, or
388 if all of the called channels are busy or unavailable. Dialplan executing will
389 continue if no requested channels can be called, or if the timeout expires.
390 This application will report normal termination if the originating channel
391 hangs up, or if the call is bridged and either of the parties in the bridge
392 ends the call.</para>
393 <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
394 application will be put into that group (as in Set(GROUP()=...).
395 If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
396 application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
397 however, the variable will be unset after use.</para>
399 <para>This application sets the following channel variables:</para>
401 <variable name="DIALEDTIME">
402 <para>This is the time from dialing a channel until when it is disconnected.</para>
404 <variable name="ANSWEREDTIME">
405 <para>This is the amount of time for actual call.</para>
407 <variable name="DIALSTATUS">
408 <para>This is the status of the call</para>
409 <value name="CHANUNAVAIL" />
410 <value name="CONGESTION" />
411 <value name="NOANSWER" />
412 <value name="BUSY" />
413 <value name="ANSWER" />
414 <value name="CANCEL" />
415 <value name="DONTCALL">
416 For the Privacy and Screening Modes.
417 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
419 <value name="TORTURE">
420 For the Privacy and Screening Modes.
421 Will be set if the called party chooses to send the calling party to the 'torture' script.
423 <value name="INVALIDARGS" />
428 <application name="RetryDial" language="en_US">
430 Place a call, retrying on failure allowing an optional exit extension.
433 <parameter name="announce" required="true">
434 <para>Filename of sound that will be played when no channel can be reached</para>
436 <parameter name="sleep" required="true">
437 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
439 <parameter name="retries" required="true">
440 <para>Number of retries</para>
441 <para>When this is reached flow will continue at the next priority in the dialplan</para>
443 <parameter name="dialargs" required="true">
444 <para>Same format as arguments provided to the Dial application</para>
448 <para>This application will attempt to place a call using the normal Dial application.
449 If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
450 Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
451 After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
452 If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
453 While waiting to retry a call, a 1 digit extension may be dialed. If that
454 extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
455 one, The call will jump to that extension immediately.
456 The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
457 to the Dial application.</para>
462 static char *app = "Dial";
463 static char *rapp = "RetryDial";
466 OPT_ANNOUNCE = (1 << 0),
467 OPT_RESETCDR = (1 << 1),
468 OPT_DTMF_EXIT = (1 << 2),
469 OPT_SENDDTMF = (1 << 3),
470 OPT_FORCECLID = (1 << 4),
471 OPT_GO_ON = (1 << 5),
472 OPT_CALLEE_HANGUP = (1 << 6),
473 OPT_CALLER_HANGUP = (1 << 7),
474 OPT_ORIGINAL_CLID = (1 << 8),
475 OPT_DURATION_LIMIT = (1 << 9),
476 OPT_MUSICBACK = (1 << 10),
477 OPT_CALLEE_MACRO = (1 << 11),
478 OPT_SCREEN_NOINTRO = (1 << 12),
479 OPT_SCREEN_NOCALLERID = (1 << 13),
480 OPT_IGNORE_CONNECTEDLINE = (1 << 14),
481 OPT_SCREENING = (1 << 15),
482 OPT_PRIVACY = (1 << 16),
483 OPT_RINGBACK = (1 << 17),
484 OPT_DURATION_STOP = (1 << 18),
485 OPT_CALLEE_TRANSFER = (1 << 19),
486 OPT_CALLER_TRANSFER = (1 << 20),
487 OPT_CALLEE_MONITOR = (1 << 21),
488 OPT_CALLER_MONITOR = (1 << 22),
489 OPT_GOTO = (1 << 23),
490 OPT_OPERMODE = (1 << 24),
491 OPT_CALLEE_PARK = (1 << 25),
492 OPT_CALLER_PARK = (1 << 26),
493 OPT_IGNORE_FORWARDING = (1 << 27),
494 OPT_CALLEE_GOSUB = (1 << 28),
495 OPT_CALLEE_MIXMONITOR = (1 << 29),
496 OPT_CALLER_MIXMONITOR = (1 << 30),
499 #define DIAL_STILLGOING (1 << 31)
500 #define DIAL_NOFORWARDHTML ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
501 #define DIAL_NOCONNECTEDLINE ((uint64_t)1 << 33)
502 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 34)
503 #define OPT_PEER_H ((uint64_t)1 << 35)
504 #define OPT_CALLEE_GO_ON ((uint64_t)1 << 36)
507 OPT_ARG_ANNOUNCE = 0,
510 OPT_ARG_DURATION_LIMIT,
512 OPT_ARG_CALLEE_MACRO,
513 OPT_ARG_CALLEE_GOSUB,
514 OPT_ARG_CALLEE_GO_ON,
516 OPT_ARG_DURATION_STOP,
518 /* note: this entry _MUST_ be the last one in the enum */
522 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
523 AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
524 AST_APP_OPTION('C', OPT_RESETCDR),
525 AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
526 AST_APP_OPTION('d', OPT_DTMF_EXIT),
527 AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
528 AST_APP_OPTION('e', OPT_PEER_H),
529 AST_APP_OPTION('f', OPT_FORCECLID),
530 AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
531 AST_APP_OPTION('g', OPT_GO_ON),
532 AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
533 AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
534 AST_APP_OPTION('H', OPT_CALLER_HANGUP),
535 AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
536 AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
537 AST_APP_OPTION('k', OPT_CALLEE_PARK),
538 AST_APP_OPTION('K', OPT_CALLER_PARK),
539 AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
540 AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
541 AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
542 AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
543 AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
544 AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
545 AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
546 AST_APP_OPTION('p', OPT_SCREENING),
547 AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
548 AST_APP_OPTION('r', OPT_RINGBACK),
549 AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
550 AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
551 AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
552 AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
553 AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
554 AST_APP_OPTION('W', OPT_CALLER_MONITOR),
555 AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
556 AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
559 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
560 OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
561 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
562 !chan->audiohooks && !peer->audiohooks)
565 * The list of active channels
568 struct chanlist *next;
569 struct ast_channel *chan;
571 struct ast_party_connected_line connected;
574 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode);
576 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
578 /* Hang up a tree of stuff */
581 /* Hangup any existing lines we have open */
582 if (outgoing->chan && (outgoing->chan != exception)) {
583 if (answered_elsewhere) {
584 /* The flag is used for local channel inheritance and stuff */
585 ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
586 /* This is for the channel drivers */
587 outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
589 ast_hangup(outgoing->chan);
592 outgoing = outgoing->next;
597 #define AST_MAX_WATCHERS 256
600 * argument to handle_cause() and other functions.
603 struct ast_channel *chan;
609 static void handle_cause(int cause, struct cause_args *num)
611 struct ast_cdr *cdr = num->chan->cdr;
620 case AST_CAUSE_CONGESTION:
626 case AST_CAUSE_NO_ROUTE_DESTINATION:
627 case AST_CAUSE_UNREGISTERED:
633 case AST_CAUSE_NORMAL_CLEARING:
642 /* free the buffer if allocated, and set the pointer to the second arg */
643 #define S_REPLACE(s, new_val) \
650 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
652 char rexten[2] = { exten, '\0' };
655 if (!ast_goto_if_exists(chan, context, rexten, pri))
658 if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
660 else if (!ast_strlen_zero(chan->macrocontext)) {
661 if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
668 /* do not call with chan lock held */
669 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
674 ast_channel_lock(chan);
675 context = ast_strdupa(S_OR(chan->macrocontext, chan->context));
676 exten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
677 ast_channel_unlock(chan);
679 return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
682 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
684 manager_event(EVENT_FLAG_CALL, "Dial",
685 "SubEvent: Begin\r\n"
687 "Destination: %s\r\n"
688 "CallerIDNum: %s\r\n"
689 "CallerIDName: %s\r\n"
691 "DestUniqueID: %s\r\n"
692 "Dialstring: %s\r\n",
693 src->name, dst->name, S_OR(src->cid.cid_num, "<unknown>"),
694 S_OR(src->cid.cid_name, "<unknown>"), src->uniqueid,
695 dst->uniqueid, dialstring ? dialstring : "");
698 static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
700 manager_event(EVENT_FLAG_CALL, "Dial",
704 "DialStatus: %s\r\n",
705 src->name, src->uniqueid, dialstatus);
709 * helper function for wait_for_answer()
711 * XXX this code is highly suspicious, as it essentially overwrites
712 * the outgoing channel without properly deleting it.
714 static void do_forward(struct chanlist *o,
715 struct cause_args *num, struct ast_flags64 *peerflags, int single)
718 struct ast_channel *original = o->chan;
719 struct ast_channel *c = o->chan; /* the winner */
720 struct ast_channel *in = num->chan; /* the input channel */
721 struct ast_party_redirecting *apr = &o->chan->redirecting;
722 struct ast_party_connected_line *apc = &o->chan->connected;
727 ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
728 if ((stuff = strchr(tmpchan, '/'))) {
732 const char *forward_context;
734 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
735 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
736 ast_channel_unlock(c);
740 /* Before processing channel, go ahead and check for forwarding */
741 ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
742 /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
743 if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
744 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
746 cause = AST_CAUSE_BUSY;
748 /* Setup parameters */
749 c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
752 ast_channel_make_compatible(o->chan, in);
753 ast_channel_inherit_variables(in, o->chan);
754 ast_channel_datastore_inherit(in, o->chan);
756 ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
759 ast_clear_flag64(o, DIAL_STILLGOING);
760 handle_cause(cause, num);
761 ast_hangup(original);
764 ast_rtp_instance_early_bridge_make_compatible(c, in);
767 c->cdrflags = in->cdrflags;
769 ast_channel_set_redirecting(c, apr);
771 while (ast_channel_trylock(in)) {
772 CHANNEL_DEADLOCK_AVOIDANCE(c);
774 S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(original->cid.cid_rdnis, S_OR(in->macroexten, in->exten))));
776 c->cid.cid_tns = in->cid.cid_tns;
778 if (ast_test_flag64(o, OPT_FORCECLID)) {
779 S_REPLACE(c->cid.cid_num, ast_strdupa(S_OR(in->macroexten, in->exten)));
780 S_REPLACE(c->cid.cid_name, NULL);
781 ast_string_field_set(c, accountcode, c->accountcode);
783 ast_party_caller_copy(&c->cid, &in->cid);
784 ast_string_field_set(c, accountcode, in->accountcode);
786 ast_party_connected_line_copy(&c->connected, apc);
788 S_REPLACE(in->cid.cid_rdnis, ast_strdup(c->cid.cid_rdnis));
789 ast_channel_update_redirecting(in, apr);
791 ast_clear_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE);
793 ast_channel_unlock(in);
794 ast_channel_unlock(c);
796 if (ast_call(c, tmpchan, 0)) {
797 ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
798 ast_clear_flag64(o, DIAL_STILLGOING);
799 ast_hangup(original);
805 while (ast_channel_trylock(in)) {
806 CHANNEL_DEADLOCK_AVOIDANCE(c);
808 senddialevent(in, c, stuff);
809 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
810 char cidname[AST_MAX_EXTENSION] = "";
811 const char *tmpexten;
812 tmpexten = ast_strdupa(S_OR(in->macroexten, in->exten));
813 ast_channel_unlock(in);
814 ast_channel_unlock(c);
815 ast_set_callerid(c, tmpexten, get_cid_name(cidname, sizeof(cidname), in), NULL);
817 ast_channel_unlock(in);
818 ast_channel_unlock(c);
820 /* Hangup the original channel now, in case we needed it */
821 ast_hangup(original);
824 ast_indicate(in, -1);
829 /* argument used for some functions. */
830 struct privacy_args {
834 char privintro[1024];
838 static struct ast_channel *wait_for_answer(struct ast_channel *in,
839 struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
840 struct privacy_args *pa,
841 const struct cause_args *num_in, int *result, char *dtmf_progress)
843 struct cause_args num = *num_in;
844 int prestart = num.busy + num.congestion + num.nochan;
846 struct ast_channel *peer = NULL;
847 /* single is set if only one destination is enabled */
848 int single = outgoing && !outgoing->next;
850 struct chanlist *epollo;
852 struct ast_party_connected_line connected_caller;
853 struct ast_str *featurecode = ast_str_alloca(FEATURE_MAX_LEN + 1);
855 /* Turn off hold music, etc */
856 if (!ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK))
857 ast_deactivate_generator(in);
859 /* If we are calling a single channel, make them compatible for in-band tone purpose */
860 ast_channel_make_compatible(outgoing->chan, in);
862 if (!ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE) && !ast_test_flag64(outgoing, DIAL_NOCONNECTEDLINE)) {
863 ast_channel_lock(outgoing->chan);
864 ast_connected_line_copy_from_caller(&connected_caller, &outgoing->chan->cid);
865 ast_channel_unlock(outgoing->chan);
866 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
867 ast_channel_update_connected_line(in, &connected_caller);
868 ast_party_connected_line_free(&connected_caller);
873 for (epollo = outgoing; epollo; epollo = epollo->next)
874 ast_poll_channel_add(in, epollo->chan);
877 while (*to && !peer) {
879 int pos = 0; /* how many channels do we handle */
880 int numlines = prestart;
881 struct ast_channel *winner;
882 struct ast_channel *watchers[AST_MAX_WATCHERS];
884 watchers[pos++] = in;
885 for (o = outgoing; o; o = o->next) {
886 /* Keep track of important channels */
887 if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
888 watchers[pos++] = o->chan;
891 if (pos == 1) { /* only the input channel is available */
892 if (numlines == (num.busy + num.congestion + num.nochan)) {
893 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
895 strcpy(pa->status, "BUSY");
896 else if (num.congestion)
897 strcpy(pa->status, "CONGESTION");
899 strcpy(pa->status, "CHANUNAVAIL");
901 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
906 winner = ast_waitfor_n(watchers, pos, to);
907 for (o = outgoing; o; o = o->next) {
909 struct ast_channel *c = o->chan;
913 if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
915 ast_verb(3, "%s answered %s\n", c->name, in->name);
916 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
917 if (o->connected.id.number) {
918 ast_channel_update_connected_line(in, &o->connected);
919 } else if (!ast_test_flag64(o, DIAL_NOCONNECTEDLINE)) {
921 ast_connected_line_copy_from_caller(&connected_caller, &c->cid);
922 ast_channel_unlock(c);
923 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
924 ast_channel_update_connected_line(in, &connected_caller);
925 ast_party_connected_line_free(&connected_caller);
929 ast_copy_flags64(peerflags, o,
930 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
931 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
932 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
933 OPT_CALLEE_PARK | OPT_CALLER_PARK |
934 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
936 ast_string_field_set(c, dialcontext, "");
937 ast_copy_string(c->exten, "", sizeof(c->exten));
943 /* here, o->chan == c == winner */
944 if (!ast_strlen_zero(c->call_forward)) {
945 do_forward(o, &num, peerflags, single);
948 f = ast_read(winner);
950 in->hangupcause = c->hangupcause;
952 ast_poll_channel_del(in, c);
956 ast_clear_flag64(o, DIAL_STILLGOING);
957 handle_cause(in->hangupcause, &num);
960 if (f->frametype == AST_FRAME_CONTROL) {
961 switch(f->subclass) {
962 case AST_CONTROL_ANSWER:
963 /* This is our guy if someone answered. */
965 ast_verb(3, "%s answered %s\n", c->name, in->name);
966 if (!single && !ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
967 if (o->connected.id.number) {
968 ast_channel_update_connected_line(in, &o->connected);
969 } else if (!ast_test_flag64(o, DIAL_NOCONNECTEDLINE)) {
971 ast_connected_line_copy_from_caller(&connected_caller, &c->cid);
972 ast_channel_unlock(c);
973 connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
974 ast_channel_update_connected_line(in, &connected_caller);
975 ast_party_connected_line_free(&connected_caller);
980 peer->cdr->answer = ast_tvnow();
981 peer->cdr->disposition = AST_CDR_ANSWERED;
983 ast_copy_flags64(peerflags, o,
984 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
985 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
986 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
987 OPT_CALLEE_PARK | OPT_CALLER_PARK |
988 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
990 ast_string_field_set(c, dialcontext, "");
991 ast_copy_string(c->exten, "", sizeof(c->exten));
992 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
993 /* Setup early bridge if appropriate */
994 ast_channel_early_bridge(in, peer);
996 /* If call has been answered, then the eventual hangup is likely to be normal hangup */
997 in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
998 c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
1000 case AST_CONTROL_BUSY:
1001 ast_verb(3, "%s is busy\n", c->name);
1002 in->hangupcause = c->hangupcause;
1005 ast_clear_flag64(o, DIAL_STILLGOING);
1006 handle_cause(AST_CAUSE_BUSY, &num);
1008 case AST_CONTROL_CONGESTION:
1009 ast_verb(3, "%s is circuit-busy\n", c->name);
1010 in->hangupcause = c->hangupcause;
1013 ast_clear_flag64(o, DIAL_STILLGOING);
1014 handle_cause(AST_CAUSE_CONGESTION, &num);
1016 case AST_CONTROL_RINGING:
1017 ast_verb(3, "%s is ringing\n", c->name);
1018 /* Setup early media if appropriate */
1019 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1020 ast_channel_early_bridge(in, c);
1021 if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
1022 ast_indicate(in, AST_CONTROL_RINGING);
1026 case AST_CONTROL_PROGRESS:
1027 ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
1028 /* Setup early media if appropriate */
1029 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1030 ast_channel_early_bridge(in, c);
1031 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1032 ast_indicate(in, AST_CONTROL_PROGRESS);
1033 if(!ast_strlen_zero(dtmf_progress)) {
1034 ast_verb(3, "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n", dtmf_progress);
1035 ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
1038 case AST_CONTROL_VIDUPDATE:
1039 ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
1040 ast_indicate(in, AST_CONTROL_VIDUPDATE);
1042 case AST_CONTROL_SRCUPDATE:
1043 ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
1044 ast_indicate(in, AST_CONTROL_SRCUPDATE);
1046 case AST_CONTROL_CONNECTED_LINE:
1047 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1048 ast_verb(3, "Connected line update to %s prevented.\n", in->name);
1049 } else if (!single) {
1050 struct ast_party_connected_line connected;
1051 ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n", c->name, in->name);
1052 ast_party_connected_line_set_init(&connected, &o->connected);
1053 ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
1054 ast_party_connected_line_set(&o->connected, &connected);
1055 ast_party_connected_line_free(&connected);
1057 ast_verb(3, "%s connected line has changed, passing it to %s\n", c->name, in->name);
1058 ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
1061 case AST_CONTROL_REDIRECTING:
1062 if (ast_test_flag64(peerflags, OPT_IGNORE_CONNECTEDLINE)) {
1063 ast_verb(3, "Redirecting update to %s prevented.\n", in->name);
1065 ast_verb(3, "%s redirecting info has changed, passing it to %s\n", c->name, in->name);
1066 ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
1069 case AST_CONTROL_PROCEEDING:
1070 ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
1071 if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
1072 ast_channel_early_bridge(in, c);
1073 if (!ast_test_flag64(outgoing, OPT_RINGBACK))
1074 ast_indicate(in, AST_CONTROL_PROCEEDING);
1076 case AST_CONTROL_HOLD:
1077 ast_verb(3, "Call on %s placed on hold\n", c->name);
1078 ast_indicate(in, AST_CONTROL_HOLD);
1080 case AST_CONTROL_UNHOLD:
1081 ast_verb(3, "Call on %s left from hold\n", c->name);
1082 ast_indicate(in, AST_CONTROL_UNHOLD);
1084 case AST_CONTROL_OFFHOOK:
1085 case AST_CONTROL_FLASH:
1086 /* Ignore going off hook and flash */
1089 if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
1090 ast_verb(3, "%s stopped sounds\n", c->name);
1091 ast_indicate(in, -1);
1092 pa->sentringing = 0;
1096 ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
1098 } else if (single) {
1099 switch (f->frametype) {
1100 case AST_FRAME_VOICE:
1101 case AST_FRAME_IMAGE:
1102 case AST_FRAME_TEXT:
1103 if (ast_write(in, f)) {
1104 ast_log(LOG_WARNING, "Unable to write frame\n");
1107 case AST_FRAME_HTML:
1108 if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass, f->data.ptr, f->datalen) == -1) {
1109 ast_log(LOG_WARNING, "Unable to send URL\n");
1119 struct ast_frame *f = ast_read(in);
1121 if (f && (f->frametype != AST_FRAME_VOICE))
1122 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1123 else if (!f || (f->frametype != AST_FRAME_VOICE))
1124 printf("Hangup received on %s\n", in->name);
1126 if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
1129 strcpy(pa->status, "CANCEL");
1130 ast_cdr_noanswer(in->cdr);
1132 if (f->data.uint32) {
1133 in->hangupcause = f->data.uint32;
1140 /* now f is guaranteed non-NULL */
1141 if (f->frametype == AST_FRAME_DTMF) {
1142 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1143 const char *context;
1144 ast_channel_lock(in);
1145 context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1146 if (onedigit_goto(in, context, (char) f->subclass, 1)) {
1147 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
1149 ast_cdr_noanswer(in->cdr);
1150 *result = f->subclass;
1151 strcpy(pa->status, "CANCEL");
1153 ast_channel_unlock(in);
1156 ast_channel_unlock(in);
1159 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1160 detect_disconnect(in, f->subclass, featurecode)) {
1161 ast_verb(3, "User requested call disconnect.\n");
1163 strcpy(pa->status, "CANCEL");
1164 ast_cdr_noanswer(in->cdr);
1170 /* Forward HTML stuff */
1171 if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
1172 if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data.ptr, f->datalen) == -1)
1173 ast_log(LOG_WARNING, "Unable to send URL\n");
1175 if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END))) {
1176 if (ast_write(outgoing->chan, f))
1177 ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
1179 if (single && (f->frametype == AST_FRAME_CONTROL) &&
1180 ((f->subclass == AST_CONTROL_HOLD) ||
1181 (f->subclass == AST_CONTROL_UNHOLD) ||
1182 (f->subclass == AST_CONTROL_VIDUPDATE) ||
1183 (f->subclass == AST_CONTROL_SRCUPDATE) ||
1184 (f->subclass == AST_CONTROL_CONNECTED_LINE) ||
1185 (f->subclass == AST_CONTROL_REDIRECTING))) {
1186 ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
1187 ast_indicate_data(outgoing->chan, f->subclass, f->data.ptr, f->datalen);
1192 ast_verb(3, "Nobody picked up in %d ms\n", orig);
1193 if (!*to || ast_check_hangup(in))
1194 ast_cdr_noanswer(in->cdr);
1198 for (epollo = outgoing; epollo; epollo = epollo->next) {
1200 ast_poll_channel_del(in, epollo->chan);
1207 static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str *featurecode)
1209 struct ast_flags features = { AST_FEATURE_DISCONNECT }; /* only concerned with disconnect feature */
1210 struct ast_call_feature feature = { 0, };
1213 ast_str_append(&featurecode, 1, "%c", code);
1215 res = ast_feature_detect(chan, &features, ast_str_buffer(featurecode), &feature);
1217 if (res != AST_FEATURE_RETURN_STOREDIGITS) {
1218 ast_str_reset(featurecode);
1220 if (feature.feature_mask & AST_FEATURE_DISCONNECT) {
1227 static void replace_macro_delimiter(char *s)
1234 /* returns true if there is a valid privacy reply */
1235 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1239 if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1241 if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1246 static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
1247 char *parse, struct timeval *calldurationlimit)
1249 char *stringp = ast_strdupa(parse);
1250 char *limit_str, *warning_str, *warnfreq_str;
1252 int play_to_caller = 0, play_to_callee = 0;
1255 limit_str = strsep(&stringp, ":");
1256 warning_str = strsep(&stringp, ":");
1257 warnfreq_str = strsep(&stringp, ":");
1259 config->timelimit = atol(limit_str);
1261 config->play_warning = atol(warning_str);
1263 config->warning_freq = atol(warnfreq_str);
1265 if (!config->timelimit) {
1266 ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
1267 config->timelimit = config->play_warning = config->warning_freq = 0;
1268 config->warning_sound = NULL;
1269 return -1; /* error */
1270 } else if ( (delta = config->play_warning - config->timelimit) > 0) {
1271 int w = config->warning_freq;
1273 /* If the first warning is requested _after_ the entire call would end,
1274 and no warning frequency is requested, then turn off the warning. If
1275 a warning frequency is requested, reduce the 'first warning' time by
1276 that frequency until it falls within the call's total time limit.
1278 timelim->| delta |<-playwarning
1279 0__________________|_________________|
1282 so the number of intervals to cut is 1+(delta-1)/w
1286 config->play_warning = 0;
1288 config->play_warning -= w * ( 1 + (delta-1)/w );
1289 if (config->play_warning < 1)
1290 config->play_warning = config->warning_freq = 0;
1294 ast_channel_lock(chan);
1296 var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
1298 play_to_caller = var ? ast_true(var) : 1;
1300 var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
1301 play_to_callee = var ? ast_true(var) : 0;
1303 if (!play_to_caller && !play_to_callee)
1306 var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
1307 config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
1309 /* The code looking at config wants a NULL, not just "", to decide
1310 * that the message should not be played, so we replace "" with NULL.
1311 * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
1315 var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
1316 config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1318 var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
1319 config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1321 ast_channel_unlock(chan);
1323 /* undo effect of S(x) in case they are both used */
1324 calldurationlimit->tv_sec = 0;
1325 calldurationlimit->tv_usec = 0;
1327 /* more efficient to do it like S(x) does since no advanced opts */
1328 if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
1329 calldurationlimit->tv_sec = config->timelimit / 1000;
1330 calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
1331 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
1332 calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
1333 config->timelimit = play_to_caller = play_to_callee =
1334 config->play_warning = config->warning_freq = 0;
1336 ast_verb(3, "Limit Data for this call:\n");
1337 ast_verb(4, "timelimit = %ld\n", config->timelimit);
1338 ast_verb(4, "play_warning = %ld\n", config->play_warning);
1339 ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
1340 ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
1341 ast_verb(4, "warning_freq = %ld\n", config->warning_freq);
1342 ast_verb(4, "start_sound = %s\n", S_OR(config->start_sound, ""));
1343 ast_verb(4, "warning_sound = %s\n", config->warning_sound);
1344 ast_verb(4, "end_sound = %s\n", S_OR(config->end_sound, ""));
1347 ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
1349 ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
1353 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1354 struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1360 /* Get the user's intro, store it in priv-callerintros/$CID,
1361 unless it is already there-- this should be done before the
1362 call is actually dialed */
1364 /* all ring indications and moh for the caller has been halted as soon as the
1365 target extension was picked up. We are going to have to kill some
1366 time and make the caller believe the peer hasn't picked up yet */
1368 if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1369 char *original_moh = ast_strdupa(chan->musicclass);
1370 ast_indicate(chan, -1);
1371 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1372 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1373 ast_string_field_set(chan, musicclass, original_moh);
1374 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1375 ast_indicate(chan, AST_CONTROL_RINGING);
1379 /* Start autoservice on the other chan ?? */
1380 res2 = ast_autoservice_start(chan);
1381 /* Now Stream the File */
1382 for (loopcount = 0; loopcount < 3; loopcount++) {
1383 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1385 if (!res2) /* on timeout, play the message again */
1386 res2 = ast_play_and_wait(peer, "priv-callpending");
1387 if (!valid_priv_reply(opts, res2))
1389 /* priv-callpending script:
1390 "I have a caller waiting, who introduces themselves as:"
1393 res2 = ast_play_and_wait(peer, pa->privintro);
1394 if (!valid_priv_reply(opts, res2))
1396 /* now get input from the called party, as to their choice */
1398 /* XXX can we have both, or they are mutually exclusive ? */
1399 if (ast_test_flag64(opts, OPT_PRIVACY))
1400 res2 = ast_play_and_wait(peer, "priv-callee-options");
1401 if (ast_test_flag64(opts, OPT_SCREENING))
1402 res2 = ast_play_and_wait(peer, "screen-callee-options");
1404 /*! \page DialPrivacy Dial Privacy scripts
1405 \par priv-callee-options script:
1406 "Dial 1 if you wish this caller to reach you directly in the future,
1407 and immediately connect to their incoming call
1408 Dial 2 if you wish to send this caller to voicemail now and
1410 Dial 3 to send this caller to the torture menus, now and forevermore.
1411 Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1412 Dial 5 to allow this caller to come straight thru to you in the future,
1413 but right now, just this once, send them to voicemail."
1414 \par screen-callee-options script:
1415 "Dial 1 if you wish to immediately connect to the incoming call
1416 Dial 2 if you wish to send this caller to voicemail.
1417 Dial 3 to send this caller to the torture menus.
1418 Dial 4 to send this caller to a simple "go away" menu.
1420 if (valid_priv_reply(opts, res2))
1422 /* invalid option */
1423 res2 = ast_play_and_wait(peer, "vm-sorry");
1426 if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1428 } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1429 ast_indicate(chan, -1);
1430 pa->sentringing = 0;
1432 ast_autoservice_stop(chan);
1433 if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1434 /* map keypresses to various things, the index is res2 - '1' */
1435 static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1436 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1438 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1439 opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1440 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1446 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1449 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1452 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1455 /* XXX should we set status to DENY ? */
1456 if (ast_test_flag64(opts, OPT_PRIVACY))
1458 /* if not privacy, then 5 is the same as "default" case */
1459 default: /* bad input or -1 if failure to start autoservice */
1460 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
1461 /* well, there seems basically two choices. Just patch the caller thru immediately,
1462 or,... put 'em thru to voicemail. */
1463 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1464 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1465 /* XXX should we set status to DENY ? */
1466 /* XXX what about the privacy flags ? */
1470 if (res2 == '1') { /* the only case where we actually connect */
1471 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1472 just clog things up, and it's not useful information, not being tied to a CID */
1473 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1474 ast_filedelete(pa->privintro, NULL);
1475 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1476 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1478 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1480 return 0; /* the good exit path */
1482 ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
1487 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1488 static int setup_privacy_args(struct privacy_args *pa,
1489 struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1494 int silencethreshold;
1496 if (!ast_strlen_zero(chan->cid.cid_num)) {
1497 l = ast_strdupa(chan->cid.cid_num);
1498 ast_shrink_phone_number(l);
1499 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1500 ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1501 pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1503 ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1504 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1509 tnam = ast_strdupa(chan->name);
1510 /* clean the channel name so slashes don't try to end up in disk file name */
1511 for (tn2 = tnam; *tn2; tn2++) {
1512 if (*tn2 == '/') /* any other chars to be afraid of? */
1515 ast_verb(3, "Privacy-- callerid is empty\n");
1517 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
1519 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1522 ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1524 if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
1525 /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
1526 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1527 pa->privdb_val = AST_PRIVACY_ALLOW;
1528 } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1529 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1532 if (pa->privdb_val == AST_PRIVACY_DENY) {
1533 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1534 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1536 } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1537 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1538 return 0; /* Is this right? */
1539 } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1540 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1541 return 0; /* is this right??? */
1542 } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1543 /* Get the user's intro, store it in priv-callerintros/$CID,
1544 unless it is already there-- this should be done before the
1545 call is actually dialed */
1547 /* make sure the priv-callerintros dir actually exists */
1548 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1549 if ((res = ast_mkdir(pa->privintro, 0755))) {
1550 ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1554 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1555 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1556 /* the DELUX version of this code would allow this caller the
1557 option to hear and retape their previously recorded intro.
1560 int duration; /* for feedback from play_and_wait */
1561 /* the file doesn't exist yet. Let the caller submit his
1562 vocal intro for posterity */
1563 /* priv-recordintro script:
1565 "At the tone, please say your name:"
1568 silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1570 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 */
1571 /* don't think we'll need a lock removed, we took care of
1572 conflicts by naming the pa.privintro file */
1574 /* Delete the file regardless since they hung up during recording */
1575 ast_filedelete(pa->privintro, NULL);
1576 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1577 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1579 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1582 if (!ast_streamfile(chan, "vm-dialout", chan->language) )
1583 ast_waitstream(chan, "");
1586 return 1; /* success */
1589 static void end_bridge_callback(void *data)
1593 struct ast_channel *chan = data;
1601 ast_channel_lock(chan);
1602 if (chan->cdr->answer.tv_sec) {
1603 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
1604 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1607 if (chan->cdr->start.tv_sec) {
1608 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
1609 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1611 ast_channel_unlock(chan);
1614 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
1615 bconfig->end_bridge_callback_data = originator;
1618 static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
1620 int res = -1; /* default: error */
1621 char *rest, *cur; /* scan the list of destinations */
1622 struct chanlist *outgoing = NULL; /* list of destinations */
1623 struct ast_channel *peer;
1624 int to; /* timeout */
1625 struct cause_args num = { chan, 0, 0, 0 };
1628 char cidname[AST_MAX_EXTENSION] = "";
1630 struct ast_bridge_config config = { { 0, } };
1631 struct timeval calldurationlimit = { 0, };
1632 char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
1633 struct privacy_args pa = {
1636 .status = "INVALIDARGS",
1638 int sentringing = 0, moh = 0;
1639 const char *outbound_group = NULL;
1643 AST_DECLARE_APP_ARGS(args,
1645 AST_APP_ARG(timeout);
1646 AST_APP_ARG(options);
1649 struct ast_flags64 opts = { 0, };
1650 char *opt_args[OPT_ARG_ARRAY_SIZE];
1651 struct ast_datastore *datastore = NULL;
1652 int fulldial = 0, num_dialed = 0;
1654 /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
1655 pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
1656 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
1657 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
1658 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
1659 pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
1661 if (ast_strlen_zero(data)) {
1662 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1663 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1667 parse = ast_strdupa(data);
1669 AST_STANDARD_APP_ARGS(args, parse);
1671 if (!ast_strlen_zero(args.options) &&
1672 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
1673 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1677 if (ast_strlen_zero(args.peers)) {
1678 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1679 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1683 if (ast_test_flag64(&opts, OPT_OPERMODE)) {
1684 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
1685 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
1688 if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
1689 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
1690 if (!calldurationlimit.tv_sec) {
1691 ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
1692 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1695 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
1698 if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
1699 dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
1700 dtmfcalled = strsep(&dtmf_progress, ":");
1701 dtmfcalling = strsep(&dtmf_progress, ":");
1704 if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
1705 if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
1709 if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
1710 ast_cdr_reset(chan->cdr, NULL);
1711 if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
1712 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
1714 if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
1715 res = setup_privacy_args(&pa, &opts, opt_args, chan);
1718 res = -1; /* reset default */
1721 if (ast_test_flag64(&opts, OPT_DTMF_EXIT) || ast_test_flag64(&opts, OPT_CALLER_HANGUP)) {
1722 __ast_answer(chan, 0, 0);
1728 /* If a channel group has been specified, get it for use when we create peer channels */
1730 ast_channel_lock(chan);
1731 if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
1732 outbound_group = ast_strdupa(outbound_group);
1733 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
1734 } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
1735 outbound_group = ast_strdupa(outbound_group);
1737 ast_channel_unlock(chan);
1738 ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_IGNORE_CONNECTEDLINE);
1740 /* loop through the list of dial destinations */
1742 while ((cur = strsep(&rest, "&")) ) {
1743 struct chanlist *tmp;
1744 struct ast_channel *tc; /* channel for this destination */
1745 /* Get a technology/[device:]number pair */
1747 char *interface = ast_strdupa(number);
1748 char *tech = strsep(&number, "/");
1749 /* find if we already dialed this interface */
1750 struct ast_dialed_interface *di;
1751 AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
1754 ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
1757 if (!(tmp = ast_calloc(1, sizeof(*tmp))))
1760 ast_copy_flags64(tmp, &opts,
1761 OPT_CANCEL_ELSEWHERE |
1762 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1763 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1764 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1765 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1766 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1767 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
1768 ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
1770 ast_copy_string(numsubst, number, sizeof(numsubst));
1771 /* Request the peer */
1773 ast_channel_lock(chan);
1774 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
1775 /* If the incoming channel has previously had connected line information
1776 * set on it (perhaps through the CONNECTED_LINE dialplan function) then
1777 * seed the calllist's connected line information with this previously
1780 if (chan->connected.id.number) {
1781 ast_party_connected_line_copy(&tmp->connected, &chan->connected);
1783 ast_channel_unlock(chan);
1786 dialed_interfaces = datastore->data;
1788 if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
1789 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
1794 datastore->inheritance = DATASTORE_INHERIT_FOREVER;
1796 if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
1801 datastore->data = dialed_interfaces;
1802 AST_LIST_HEAD_INIT(dialed_interfaces);
1804 ast_channel_lock(chan);
1805 ast_channel_datastore_add(chan, datastore);
1806 ast_channel_unlock(chan);
1809 AST_LIST_LOCK(dialed_interfaces);
1810 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
1811 if (!strcasecmp(di->interface, interface)) {
1812 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
1817 AST_LIST_UNLOCK(dialed_interfaces);
1825 /* It is always ok to dial a Local interface. We only keep track of
1826 * which "real" interfaces have been dialed. The Local channel will
1827 * inherit this list so that if it ends up dialing a real interface,
1828 * it won't call one that has already been called. */
1829 if (strcasecmp(tech, "Local")) {
1830 if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
1831 AST_LIST_UNLOCK(dialed_interfaces);
1835 strcpy(di->interface, interface);
1837 AST_LIST_LOCK(dialed_interfaces);
1838 AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
1839 AST_LIST_UNLOCK(dialed_interfaces);
1842 tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
1844 /* If we can't, just go on to the next call */
1845 ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
1846 tech, cause, ast_cause2str(cause));
1847 handle_cause(cause, &num);
1848 if (!rest) /* we are on the last destination */
1849 chan->hangupcause = cause;
1853 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
1855 ast_channel_lock(tc);
1856 while (ast_channel_trylock(chan)) {
1857 CHANNEL_DEADLOCK_AVOIDANCE(tc);
1859 /* Setup outgoing SDP to match incoming one */
1860 if (!outgoing && !rest) {
1861 ast_rtp_instance_early_bridge_make_compatible(tc, chan);
1864 /* Inherit specially named variables from parent channel */
1865 ast_channel_inherit_variables(chan, tc);
1866 ast_channel_datastore_inherit(chan, tc);
1868 tc->appl = "AppDial";
1869 tc->data = "(Outgoing Line)";
1870 memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
1872 /* If the new channel has no callerid, try to guess what it should be */
1873 if (ast_strlen_zero(tc->cid.cid_num)) {
1874 if (!ast_strlen_zero(chan->connected.id.number)) {
1875 ast_set_callerid(tc, chan->connected.id.number, chan->connected.id.name, chan->connected.ani);
1876 } else if (!ast_strlen_zero(chan->cid.cid_dnid)) {
1877 ast_set_callerid(tc, chan->cid.cid_dnid, NULL, NULL);
1878 } else if (!ast_strlen_zero(S_OR(chan->macroexten, chan->exten))) {
1879 ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), NULL, NULL);
1881 ast_set_flag64(tmp, DIAL_NOCONNECTEDLINE);
1884 ast_connected_line_copy_from_caller(&tc->connected, &chan->cid);
1886 S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
1887 ast_party_redirecting_copy(&tc->redirecting, &chan->redirecting);
1889 tc->cid.cid_tns = chan->cid.cid_tns;
1891 ast_string_field_set(tc, accountcode, chan->accountcode);
1892 tc->cdrflags = chan->cdrflags;
1893 if (ast_strlen_zero(tc->musicclass))
1894 ast_string_field_set(tc, musicclass, chan->musicclass);
1896 /* Pass ADSI CPE and transfer capability */
1897 tc->adsicpe = chan->adsicpe;
1898 tc->transfercapability = chan->transfercapability;
1900 /* If we have an outbound group, set this peer channel to it */
1902 ast_app_group_set_channel(tc, outbound_group);
1903 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
1904 if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
1905 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1907 /* Check if we're forced by configuration */
1908 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
1909 ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1912 /* Inherit context and extension */
1913 ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
1914 if (!ast_strlen_zero(chan->macroexten))
1915 ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
1917 ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
1919 ast_channel_unlock(tc);
1920 res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
1922 /* Save the info in cdr's that we called them */
1924 ast_cdr_setdestchan(chan->cdr, tc->name);
1926 /* check the results of ast_call */
1928 /* Again, keep going even if there's an error */
1929 ast_debug(1, "ast call on peer returned %d\n", res);
1930 ast_verb(3, "Couldn't call %s\n", numsubst);
1931 if (tc->hangupcause) {
1932 chan->hangupcause = tc->hangupcause;
1934 ast_channel_unlock(chan);
1940 const char *tmpexten = ast_strdupa(S_OR(chan->macroexten, chan->exten));
1941 senddialevent(chan, tc, numsubst);
1942 ast_verb(3, "Called %s\n", numsubst);
1943 ast_channel_unlock(chan);
1944 if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
1945 ast_set_callerid(tc, tmpexten, get_cid_name(cidname, sizeof(cidname), chan), NULL);
1948 /* Put them in the list of outgoing thingies... We're ready now.
1949 XXX If we're forcibly removed, these outgoing calls won't get
1951 ast_set_flag64(tmp, DIAL_STILLGOING);
1953 tmp->next = outgoing;
1955 /* If this line is up, don't try anybody else */
1956 if (outgoing->chan->_state == AST_STATE_UP)
1960 if (ast_strlen_zero(args.timeout)) {
1963 to = atoi(args.timeout);
1967 ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
1973 strcpy(pa.status, "CHANUNAVAIL");
1974 if (fulldial == num_dialed) {
1979 /* Our status will at least be NOANSWER */
1980 strcpy(pa.status, "NOANSWER");
1981 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
1983 if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1984 char *original_moh = ast_strdupa(chan->musicclass);
1985 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1986 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1987 ast_string_field_set(chan, musicclass, original_moh);
1989 ast_moh_start(chan, NULL, NULL);
1991 ast_indicate(chan, AST_CONTROL_PROGRESS);
1992 } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
1993 ast_indicate(chan, AST_CONTROL_RINGING);
1998 peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result, dtmf_progress);
2000 /* The ast_channel_datastore_remove() function could fail here if the
2001 * datastore was moved to another channel during a masquerade. If this is
2002 * the case, don't free the datastore here because later, when the channel
2003 * to which the datastore was moved hangs up, it will attempt to free this
2004 * datastore again, causing a crash
2006 if (!ast_channel_datastore_remove(chan, datastore))
2007 ast_datastore_free(datastore);
2011 } else if (to) { /* Musta gotten hung up */
2013 } else { /* Nobody answered, next please? */
2017 /* SIP, in particular, sends back this error code to indicate an
2018 * overlap dialled number needs more digits. */
2019 if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
2020 res = AST_PBX_INCOMPLETE;
2023 /* almost done, although the 'else' block is 400 lines */
2027 strcpy(pa.status, "ANSWER");
2028 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2029 /* Ah ha! Someone answered within the desired timeframe. Of course after this
2030 we will always return with -1 so that it is hung up properly after the
2032 hanguptree(outgoing, peer, 1);
2034 /* If appropriate, log that we have a destination channel */
2036 ast_cdr_setdestchan(chan->cdr, peer->name);
2038 pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
2040 ast_channel_lock(peer);
2041 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
2044 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
2045 ast_channel_unlock(peer);
2047 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
2048 ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
2049 ast_channel_sendurl( peer, args.url );
2051 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
2052 if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
2057 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
2061 /* Start autoservice on the other chan */
2062 res = ast_autoservice_start(chan);
2063 /* Now Stream the File */
2065 res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
2067 digit = ast_waitstream(peer, AST_DIGIT_ANY);
2069 /* Ok, done. stop autoservice */
2070 res = ast_autoservice_stop(chan);
2071 if (digit > 0 && !res)
2072 res = ast_senddigit(chan, digit, 0);
2078 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
2079 replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
2080 ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
2081 /* peer goes to the same context and extension as chan, so just copy info from chan*/
2082 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2083 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2084 peer->priority = chan->priority + 2;
2085 ast_pbx_start(peer);
2086 hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
2093 if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
2094 struct ast_app *theapp;
2095 const char *macro_result;
2097 res = ast_autoservice_start(chan);
2099 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2103 theapp = pbx_findapp("Macro");
2105 if (theapp && !res) { /* XXX why check res here ? */
2106 /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
2107 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
2108 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
2110 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
2111 res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
2112 ast_debug(1, "Macro exited with status %d\n", res);
2115 ast_log(LOG_ERROR, "Could not find application Macro\n");
2119 if (ast_autoservice_stop(chan) < 0) {
2120 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2124 ast_channel_lock(peer);
2126 if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
2127 char *macro_transfer_dest;
2129 if (!strcasecmp(macro_result, "BUSY")) {
2130 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2131 ast_set_flag64(peerflags, OPT_GO_ON);
2133 } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
2134 ast_copy_string(pa.status, macro_result, sizeof(pa.status));
2135 ast_set_flag64(peerflags, OPT_GO_ON);
2137 } else if (!strcasecmp(macro_result, "CONTINUE")) {
2138 /* hangup peer and keep chan alive assuming the macro has changed
2139 the context / exten / priority or perhaps
2140 the next priority in the current exten is desired.
2142 ast_set_flag64(peerflags, OPT_GO_ON);
2144 } else if (!strcasecmp(macro_result, "ABORT")) {
2145 /* Hangup both ends unless the caller has the g flag */
2147 } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
2149 /* perform a transfer to a new extension */
2150 if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
2151 replace_macro_delimiter(macro_transfer_dest);
2152 if (!ast_parseable_goto(chan, macro_transfer_dest))
2153 ast_set_flag64(peerflags, OPT_GO_ON);
2158 ast_channel_unlock(peer);
2161 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
2162 struct ast_app *theapp;
2163 const char *gosub_result;
2164 char *gosub_args, *gosub_argstart;
2167 res9 = ast_autoservice_start(chan);
2169 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2173 theapp = pbx_findapp("Gosub");
2175 if (theapp && !res9) {
2176 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
2178 /* Set where we came from */
2179 ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
2180 ast_copy_string(peer->exten, "s", sizeof(peer->exten));
2183 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
2184 if (gosub_argstart) {
2185 *gosub_argstart = 0;
2186 if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
2187 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2190 *gosub_argstart = ',';
2192 if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
2193 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2199 res9 = pbx_exec(peer, theapp, gosub_args);
2201 struct ast_pbx_args args;
2202 /* A struct initializer fails to compile for this case ... */
2203 memset(&args, 0, sizeof(args));
2204 args.no_hangup_chan = 1;
2205 ast_pbx_run_args(peer, &args);
2207 ast_free(gosub_args);
2208 ast_debug(1, "Gosub exited with status %d\n", res9);
2210 ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
2214 ast_log(LOG_ERROR, "Could not find application Gosub\n");
2218 if (ast_autoservice_stop(chan) < 0) {
2219 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2223 ast_channel_lock(peer);
2225 if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
2226 char *gosub_transfer_dest;
2228 if (!strcasecmp(gosub_result, "BUSY")) {
2229 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2230 ast_set_flag64(peerflags, OPT_GO_ON);
2232 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
2233 ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2234 ast_set_flag64(peerflags, OPT_GO_ON);
2236 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
2237 /* hangup peer and keep chan alive assuming the macro has changed
2238 the context / exten / priority or perhaps
2239 the next priority in the current exten is desired.
2241 ast_set_flag64(peerflags, OPT_GO_ON);
2243 } else if (!strcasecmp(gosub_result, "ABORT")) {
2244 /* Hangup both ends unless the caller has the g flag */
2246 } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
2248 /* perform a transfer to a new extension */
2249 if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
2250 replace_macro_delimiter(gosub_transfer_dest);
2251 if (!ast_parseable_goto(chan, gosub_transfer_dest))
2252 ast_set_flag64(peerflags, OPT_GO_ON);
2257 ast_channel_unlock(peer);
2261 if (!ast_tvzero(calldurationlimit)) {
2262 struct timeval whentohangup = calldurationlimit;
2263 peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
2265 if (!ast_strlen_zero(dtmfcalled)) {
2266 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
2267 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
2269 if (!ast_strlen_zero(dtmfcalling)) {
2270 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
2271 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
2275 if (res) { /* some error */
2278 if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
2279 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
2280 if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
2281 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
2282 if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
2283 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
2284 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
2285 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
2286 if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
2287 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
2288 if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
2289 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
2290 if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
2291 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
2292 if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
2293 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
2294 if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
2295 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
2296 if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
2297 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
2298 if (ast_test_flag64(peerflags, OPT_GO_ON))
2299 ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
2301 config.end_bridge_callback = end_bridge_callback;
2302 config.end_bridge_callback_data = chan;
2303 config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
2308 } else if (sentringing) {
2310 ast_indicate(chan, -1);
2312 /* Be sure no generators are left on it */
2313 ast_deactivate_generator(chan);
2314 /* Make sure channels are compatible */
2315 res = ast_channel_make_compatible(chan, peer);
2317 ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
2323 struct oprmode oprmode;
2325 oprmode.peer = peer;
2326 oprmode.mode = opermode;
2328 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
2330 res = ast_bridge_call(chan, peer, &config);
2333 strcpy(peer->context, chan->context);
2335 if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
2340 strcpy(peer->exten, "h");
2342 autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
2343 ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
2345 while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
2348 if (found && res9) {
2349 /* Something bad happened, or a hangup has been requested. */
2350 ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2351 ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2353 ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP); /* set it back the way it was */
2355 if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON)) {
2356 if(!ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {
2357 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
2358 ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
2361 res = ast_goto_if_exists(peer, chan->context, chan->exten, (chan->priority) + 1);
2362 if (res == AST_PBX_GOTO_FAILED) {
2367 ast_pbx_start(peer);
2369 if (!ast_check_hangup(chan))
2370 chan->hangupcause = peer->hangupcause;
2378 } else if (sentringing) {
2380 ast_indicate(chan, -1);
2382 ast_channel_early_bridge(chan, NULL);
2383 hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
2384 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2385 senddialendevent(chan, pa.status);
2386 ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
2388 if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
2389 if (!ast_tvzero(calldurationlimit))
2390 memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
2395 if (config.warning_sound) {
2396 ast_free((char *)config.warning_sound);
2398 if (config.end_sound) {
2399 ast_free((char *)config.end_sound);
2401 if (config.start_sound) {
2402 ast_free((char *)config.start_sound);
2407 static int dial_exec(struct ast_channel *chan, void *data)
2409 struct ast_flags64 peerflags;
2411 memset(&peerflags, 0, sizeof(peerflags));
2413 return dial_exec_full(chan, data, &peerflags, NULL);
2416 static int retrydial_exec(struct ast_channel *chan, void *data)
2419 const char *context = NULL;
2420 int sleepms = 0, loops = 0, res = -1;
2421 struct ast_flags64 peerflags = { 0, };
2422 AST_DECLARE_APP_ARGS(args,
2423 AST_APP_ARG(announce);
2425 AST_APP_ARG(retries);
2426 AST_APP_ARG(dialdata);
2429 if (ast_strlen_zero(data)) {
2430 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
2434 parse = ast_strdupa(data);
2435 AST_STANDARD_APP_ARGS(args, parse);
2437 if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
2440 if (!ast_strlen_zero(args.retries)) {
2441 loops = atoi(args.retries);
2444 if (!args.dialdata) {
2445 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
2453 loops = -1; /* run forever */
2455 ast_channel_lock(chan);
2456 context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
2457 context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
2458 ast_channel_unlock(chan);
2464 chan->data = "Retrying";
2465 if (ast_test_flag(chan, AST_FLAG_MOH))
2468 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
2473 if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
2474 if (!ast_strlen_zero(args.announce)) {
2475 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2476 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2477 ast_waitstream(chan, AST_DIGIT_ANY);
2479 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2481 if (!res && sleepms) {
2482 if (!ast_test_flag(chan, AST_FLAG_MOH))
2483 ast_moh_start(chan, NULL, NULL);
2484 res = ast_waitfordigit(chan, sleepms);
2487 if (!ast_strlen_zero(args.announce)) {
2488 if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2489 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2490 res = ast_waitstream(chan, "");
2492 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2495 if (!ast_test_flag(chan, AST_FLAG_MOH))
2496 ast_moh_start(chan, NULL, NULL);
2498 res = ast_waitfordigit(chan, sleepms);
2503 if (res < 0 || res == AST_PBX_INCOMPLETE) {
2505 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
2506 if (onedigit_goto(chan, context, (char) res, 1)) {
2518 if (ast_test_flag(chan, AST_FLAG_MOH))
2524 static int unload_module(void)
2527 struct ast_context *con;
2529 res = ast_unregister_application(app);
2530 res |= ast_unregister_application(rapp);
2532 if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
2533 ast_context_remove_extension2(con, "s", 1, NULL, 0);
2534 ast_context_destroy(con, "app_dial"); /* leave nothing behind */
2540 static int load_module(void)
2543 struct ast_context *con;
2545 con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
2547 ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
2549 ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
2551 res = ast_register_application_xml(app, dial_exec);
2552 res |= ast_register_application_xml(rapp, retrydial_exec);
2557 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");
projects / asterisk/asterisk.git / blob