Merged revisions 172517 via svnmerge from
[asterisk/asterisk.git] / apps / app_dial.c
1 /*
2  * Asterisk -- An open source telephony toolkit.
3  *
4  * Copyright (C) 1999 - 2008, Digium, Inc.
5  *
6  * Mark Spencer <markster@digium.com>
7  *
8  * See http://www.asterisk.org for more information about
9  * the Asterisk project. Please do not directly contact
10  * any of the maintainers of this project for assistance;
11  * the project provides a web site, mailing lists and IRC
12  * channels for your use.
13  *
14  * This program is free software, distributed under the terms of
15  * the GNU General Public License Version 2. See the LICENSE file
16  * at the top of the source tree.
17  */
18
19 /*! \file
20  *
21  * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
22  *
23  * \author Mark Spencer <markster@digium.com>
24  *
25  * \ingroup applications
26  */
27
28 /*** MODULEINFO
29         <depend>chan_local</depend>
30  ***/
31
32
33 #include "asterisk.h"
34
35 ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
36
37 #include <sys/time.h>
38 #include <sys/signal.h>
39 #include <sys/stat.h>
40 #include <netinet/in.h>
41
42 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
43 #include "asterisk/lock.h"
44 #include "asterisk/file.h"
45 #include "asterisk/channel.h"
46 #include "asterisk/pbx.h"
47 #include "asterisk/module.h"
48 #include "asterisk/translate.h"
49 #include "asterisk/say.h"
50 #include "asterisk/config.h"
51 #include "asterisk/features.h"
52 #include "asterisk/musiconhold.h"
53 #include "asterisk/callerid.h"
54 #include "asterisk/utils.h"
55 #include "asterisk/app.h"
56 #include "asterisk/causes.h"
57 #include "asterisk/rtp.h"
58 #include "asterisk/cdr.h"
59 #include "asterisk/manager.h"
60 #include "asterisk/privacy.h"
61 #include "asterisk/stringfields.h"
62 #include "asterisk/global_datastores.h"
63 #include "asterisk/dsp.h"
64
65 /*** DOCUMENTATION
66         <application name="Dial" language="en_US">
67                 <synopsis>
68                         Attempt to connect to another device or endpoint and bridge the call.
69                 </synopsis>
70                 <syntax>
71                         <parameter name="Technology/Resource" required="true" argsep="&amp;">
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>
77                                 </argument>
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&amp;Technology3/Resourse3&amp;.....</para>
82                                 </argument>
83                         </parameter>
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>
87                         </parameter>
88                         <parameter name="options" required="false">
89                            <optionlist>
90                                 <option name="A">
91                                         <argument name="x" required="true">
92                                                 <para>The file to play to the called party</para>
93                                         </argument>
94                                         <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
95                                 </option>
96                                 <option name="C">
97                                         <para>Reset the call detail record (CDR) for this call.</para>
98                                 </option>
99                                 <option name="c">
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>
102                                 </option>
103                                 <option name="d">
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,
107                                         if it exists.</para>
108                                 </option>
109                                 <option name="D" argsep=":">
110                                         <argument name="called" />
111                                         <argument name="calling" />
112                                         <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
113                                         party has answered, but before the call gets bridged. The 
114                                         <replaceable>called</replaceable> DTMF string is sent to the called party, and the 
115                                         <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments 
116                                         can be used alone.</para>
117                                 </option>
118                                 <option name="e">
119                                         <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
120                                 </option>
121                                 <option name="f">
122                                         <para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
123                                         extension associated with the channel using a dialplan <literal>hint</literal>.
124                                         For example, some PSTNs do not allow CallerID to be set to anything
125                                         other than the number assigned to the caller.</para>
126                                 </option>
127                                 <option name="F" argsep="^">
128                                         <argument name="context" required="false" />
129                                         <argument name="exten" required="false" />
130                                         <argument name="priority" required="true" />
131                                         <para>When the caller hangs up, transfer the called party
132                                         to the specified destination and continue execution at that location.</para>
133                                 </option>
134                                 <option name="g">
135                                         <para>Proceed with dialplan execution at the next priority in the current extension if the
136                                         destination channel hangs up.</para>
137                                 </option>
138                                 <option name="G" argsep="^">
139                                         <argument name="context" required="false" />
140                                         <argument name="exten" required="false" />
141                                         <argument name="priority" required="true" />
142                                         <para>If the call is answered, transfer the calling party to
143                                         the specified <replaceable>priority</replaceable> and the called party to the specified 
144                                         <replaceable>priority</replaceable> plus one.</para>
145                                         <note>
146                                                 <para>You cannot use any additional action post answer options in conjunction with this option.</para>
147                                         </note>
148                                 </option>
149                                 <option name="h">
150                                         <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
151                                 </option>
152                                 <option name="H">
153                                         <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
154                                 </option>
155                                 <option name="i">
156                                         <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
157                                 </option>
158                                 <option name="k">
159                                         <para>Allow the called party to enable parking of the call by sending
160                                         the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
161                                 </option>
162                                 <option name="K">
163                                         <para>Allow the calling party to enable parking of the call by sending
164                                         the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
165                                 </option>
166                                 <option name="L" argsep=":">
167                                         <argument name="x" required="true">
168                                                 <para>Maximum call time, in milliseconds</para>
169                                         </argument>
170                                         <argument name="y">
171                                                 <para>Warning time, in milliseconds</para>
172                                         </argument>
173                                         <argument name="z">
174                                                 <para>Repeat time, in milliseconds</para>
175                                         </argument>
176                                         <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
177                                         left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
178                                         <para>This option is affected by the following variables:</para>
179                                         <variablelist>
180                                                 <variable name="LIMIT_PLAYAUDIO_CALLER">
181                                                         <value name="yes" default="true" />
182                                                         <value name="no" />
183                                                         <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
184                                                 </variable>
185                                                 <variable name="LIMIT_PLAYAUDIO_CALLEE">
186                                                         <value name="yes" />
187                                                         <value name="no" default="true"/>
188                                                         <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
189                                                 </variable>
190                                                 <variable name="LIMIT_TIMEOUT_FILE">
191                                                         <value name="filename"/>
192                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
193                                                         If not set, the time remaining will be announced.</para>
194                                                 </variable>
195                                                 <variable name="LIMIT_CONNECT_FILE">
196                                                         <value name="filename"/>
197                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
198                                                         If not set, the time remaining will be announced.</para>
199                                                 </variable>
200                                                 <variable name="LIMIT_WARNING_FILE">
201                                                         <value name="filename"/>
202                                                         <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
203                                                         a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
204                                                 </variable>
205                                         </variablelist>
206                                 </option>
207                                 <option name="m">
208                                         <argument name="class" required="false"/>
209                                         <para>Provide hold music to the calling party until a requested
210                                         channel answers. A specific music on hold <replaceable>class</replaceable>
211                                         (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
212                                 </option>
213                                 <option name="M" argsep="^">
214                                         <argument name="macro" required="true">
215                                                 <para>Name of the macro that should be executed.</para>
216                                         </argument>
217                                         <argument name="arg" multiple="true">
218                                                 <para>Macro arguments</para>
219                                         </argument>
220                                         <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel 
221                                         before connecting to the calling channel. Arguments can be specified to the Macro
222                                         using <literal>^</literal> as a delimiter. The macro can set the variable
223                                         <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
224                                         finished executing:</para>
225                                         <variablelist>
226                                                 <variable name="MACRO_RESULT">
227                                                         <para>If set, this action will be taken after the macro finished executing.</para>
228                                                         <value name="ABORT">
229                                                                 Hangup both legs of the call
230                                                         </value>
231                                                         <value name="CONGESTION">
232                                                                 Behave as if line congestion was encountered
233                                                         </value>
234                                                         <value name="BUSY">
235                                                                 Behave as if a busy signal was encountered
236                                                         </value>
237                                                         <value name="CONTINUE">
238                                                                 Hangup the called party and allow the calling party to continue dialplan execution at the next priority
239                                                         </value>
240                                                         <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
241                                                         <value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
242                                                                 Transfer the call to the specified destination.
243                                                         </value>
244                                                 </variable>
245                                         </variablelist>
246                                         <note>
247                                                 <para>You cannot use any additional action post answer options in conjunction
248                                                 with this option. Also, pbx services are not run on the peer (called) channel,
249                                                 so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
250                                         </note>
251                                 </option>
252                                 <option name="n">
253                                         <para>This option is a modifier for the call screening/privacy mode. (See the 
254                                         <literal>p</literal> and <literal>P</literal> options.) It specifies
255                                         that no introductions are to be saved in the <directory>priv-callerintros</directory>
256                                         directory.</para>
257                                 </option>
258                                 <option name="N">
259                                         <para>This option is a modifier for the call screening/privacy mode. It specifies
260                                         that if Caller*ID is present, do not screen the call.</para>
261                                 </option>
262                                 <option name="o">
263                                         <para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
264                                         be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
265                                         behavior of Asterisk 1.0 and earlier.</para>
266                                 </option>
267                                 <option name="O">
268                                         <argument name="mode">
269                                                 <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
270                                                 the originator hanging up will cause the phone to ring back immediately.</para>
271                                                 <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator 
272                                                 flashes the trunk, it will ring their phone back.</para>
273                                         </argument>
274                                         <para>Enables <emphasis>operator services</emphasis> mode.  This option only
275                                         works when bridging a DAHDI channel to another DAHDI channel
276                                         only. if specified on non-DAHDI interfaces, it will be ignored.
277                                         When the destination answers (presumably an operator services
278                                         station), the originator no longer has control of their line.
279                                         They may hang up, but the switch will not release their line
280                                         until the destination party (the operator) hangs up.</para>
281                                 </option>
282                                 <option name="p">
283                                         <para>This option enables screening mode. This is basically Privacy mode
284                                         without memory.</para>
285                                 </option>
286                                 <option name="P">
287                                         <argument name="x" />
288                                         <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
289                                         it is provided. The current extension is used if a database family/key is not specified.</para>
290                                 </option>
291                                 <option name="r">
292                                         <para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
293                                         party until the called channel has answered.</para>
294                                 </option>
295                                 <option name="S">
296                                         <argument name="x" required="true" />
297                                         <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
298                                         answered the call.</para>
299                                 </option>
300                                 <option name="t">
301                                         <para>Allow the called party to transfer the calling party by sending the
302                                         DTMF sequence defined in <filename>features.conf</filename>.</para>
303                                 </option>
304                                 <option name="T">
305                                         <para>Allow the calling party to transfer the called party by sending the
306                                         DTMF sequence defined in <filename>features.conf</filename>.</para>
307                                 </option>
308                                 <option name="U" argsep="^">
309                                         <argument name="x" required="true">
310                                                 <para>Name of the subroutine to execute via Gosub</para>
311                                         </argument>
312                                         <argument name="arg" multiple="true" required="false">
313                                                 <para>Arguments for the Gosub routine</para>
314                                         </argument>
315                                         <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
316                                         to the calling channel. Arguments can be specified to the Gosub
317                                         using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
318                                         <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
319                                         <variablelist>
320                                                 <variable name="GOSUB_RESULT">
321                                                         <value name="ABORT">
322                                                                 Hangup both legs of the call.
323                                                         </value>
324                                                         <value name="CONGESTION">
325                                                                 Behave as if line congestion was encountered.
326                                                         </value>
327                                                         <value name="BUSY">
328                                                                 Behave as if a busy signal was encountered.
329                                                         </value>
330                                                         <value name="CONTINUE">
331                                                                 Hangup the called party and allow the calling party
332                                                                 to continue dialplan execution at the next priority.
333                                                         </value>
334                                                         <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
335                                                         <value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
336                                                                 Transfer the call to the specified priority. Optionally, an extension, or
337                                                                 extension and priority can be specified.
338                                                         </value>
339                                                 </variable>
340                                         </variablelist>
341                                         <note>
342                                                 <para>You cannot use any additional action post answer options in conjunction
343                                                 with this option. Also, pbx services are not run on the peer (called) channel,
344                                                 so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
345                                         </note>
346                                 </option>
347                                 <option name="w">
348                                         <para>Allow the called party to enable recording of the call by sending
349                                         the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
350                                 </option>
351                                 <option name="W">
352                                         <para>Allow the calling party to enable recording of the call by sending
353                                         the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
354                                 </option>
355                                 <option name="x">
356                                         <para>Allow the called party to enable recording of the call by sending
357                                         the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
358                                 </option>
359                                 <option name="X">
360                                         <para>Allow the calling party to enable recording of the call by sending
361                                         the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
362                                 </option>
363                                 </optionlist>
364                         </parameter>
365                         <parameter name="URL">
366                                 <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
367                         </parameter>
368                 </syntax>
369                 <description>
370                         <para>This application will place calls to one or more specified channels. As soon
371                         as one of the requested channels answers, the originating channel will be
372                         answered, if it has not already been answered. These two channels will then
373                         be active in a bridged call. All other channels that were requested will then
374                         be hung up.</para>
375
376                         <para>Unless there is a timeout specified, the Dial application will wait
377                         indefinitely until one of the called channels answers, the user hangs up, or
378                         if all of the called channels are busy or unavailable. Dialplan executing will
379                         continue if no requested channels can be called, or if the timeout expires.
380                         This application will report normal termination if the originating channel
381                         hangs up, or if the call is bridged and either of the parties in the bridge
382                         ends the call.</para>
383
384                         <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
385                         application will be put into that group (as in Set(GROUP()=...).
386                         If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
387                         application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
388                         however, the variable will be unset after use.</para>
389
390                         <para>This application sets the following channel variables:</para>
391                         <variablelist>
392                                 <variable name="DIALEDTIME">
393                                         <para>This is the time from dialing a channel until when it is disconnected.</para>
394                                 </variable>
395                                 <variable name="ANSWEREDTIME">
396                                         <para>This is the amount of time for actual call.</para>
397                                 </variable>
398                                 <variable name="DIALSTATUS">
399                                         <para>This is the status of the call</para>
400                                         <value name="CHANUNAVAIL" />
401                                         <value name="CONGESTION" />
402                                         <value name="NOANSWER" />
403                                         <value name="BUSY" />
404                                         <value name="ANSWER" />
405                                         <value name="CANCEL" />
406                                         <value name="DONTCALL">
407                                                 For the Privacy and Screening Modes.
408                                                 Will be set if the called party chooses to send the calling party to the 'Go Away' script.
409                                         </value>
410                                         <value name="TORTURE">
411                                                 For the Privacy and Screening Modes.
412                                                 Will be set if the called party chooses to send the calling party to the 'torture' script.
413                                         </value>
414                                         <value name="INVALIDARGS" />
415                                 </variable>
416                         </variablelist>
417                 </description>
418         </application>
419         <application name="RetryDial" language="en_US">
420                 <synopsis>
421                         Place a call, retrying on failure allowing an optional exit extension.
422                 </synopsis>
423                 <syntax>
424                         <parameter name="announce" required="true">
425                                 <para>Filename of sound that will be played when no channel can be reached</para>
426                         </parameter>
427                         <parameter name="sleep" required="true">
428                                 <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
429                         </parameter>
430                         <parameter name="retries" required="true">
431                                 <para>Number of retries</para>
432                                 <para>When this is reached flow will continue at the next priority in the dialplan</para>
433                         </parameter>
434                         <parameter name="dialargs" required="true">
435                                 <para>Same format as arguments provided to the Dial application</para>
436                         </parameter>
437                 </syntax>
438                 <description>
439                         <para>This application will attempt to place a call using the normal Dial application.
440                         If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
441                         Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
442                         After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
443                         If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
444                         While waiting to retry a call, a 1 digit extension may be dialed. If that
445                         extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
446                         one, The call will jump to that extension immediately.
447                         The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
448                         to the Dial application.</para>
449                 </description>
450         </application>
451  ***/
452
453 static char *app = "Dial";
454 static char *rapp = "RetryDial";
455
456 enum {
457         OPT_ANNOUNCE =          (1 << 0),
458         OPT_RESETCDR =          (1 << 1),
459         OPT_DTMF_EXIT =         (1 << 2),
460         OPT_SENDDTMF =          (1 << 3),
461         OPT_FORCECLID =         (1 << 4),
462         OPT_GO_ON =             (1 << 5),
463         OPT_CALLEE_HANGUP =     (1 << 6),
464         OPT_CALLER_HANGUP =     (1 << 7),
465         OPT_DURATION_LIMIT =    (1 << 9),
466         OPT_MUSICBACK =         (1 << 10),
467         OPT_CALLEE_MACRO =      (1 << 11),
468         OPT_SCREEN_NOINTRO =    (1 << 12),
469         OPT_SCREEN_NOCLID =     (1 << 13),
470         OPT_ORIGINAL_CLID =     (1 << 14),
471         OPT_SCREENING =         (1 << 15),
472         OPT_PRIVACY =           (1 << 16),
473         OPT_RINGBACK =          (1 << 17),
474         OPT_DURATION_STOP =     (1 << 18),
475         OPT_CALLEE_TRANSFER =   (1 << 19),
476         OPT_CALLER_TRANSFER =   (1 << 20),
477         OPT_CALLEE_MONITOR =    (1 << 21),
478         OPT_CALLER_MONITOR =    (1 << 22),
479         OPT_GOTO =              (1 << 23),
480         OPT_OPERMODE =          (1 << 24),
481         OPT_CALLEE_PARK =       (1 << 25),
482         OPT_CALLER_PARK =       (1 << 26),
483         OPT_IGNORE_FORWARDING = (1 << 27),
484         OPT_CALLEE_GOSUB =      (1 << 28),
485         OPT_CALLEE_MIXMONITOR = (1 << 29),
486         OPT_CALLER_MIXMONITOR = (1 << 30),
487 };
488
489 #define DIAL_STILLGOING      (1 << 31)
490 #define DIAL_NOFORWARDHTML   ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
491 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 33)
492 #define OPT_PEER_H           ((uint64_t)1 << 34)
493 #define OPT_CALLEE_GO_ON     ((uint64_t)1 << 35)
494
495 enum {
496         OPT_ARG_ANNOUNCE = 0,
497         OPT_ARG_SENDDTMF,
498         OPT_ARG_GOTO,
499         OPT_ARG_DURATION_LIMIT,
500         OPT_ARG_MUSICBACK,
501         OPT_ARG_CALLEE_MACRO,
502         OPT_ARG_CALLEE_GOSUB,
503         OPT_ARG_CALLEE_GO_ON,
504         OPT_ARG_PRIVACY,
505         OPT_ARG_DURATION_STOP,
506         OPT_ARG_OPERMODE,
507         /* note: this entry _MUST_ be the last one in the enum */
508         OPT_ARG_ARRAY_SIZE,
509 };
510
511 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
512         AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
513         AST_APP_OPTION('C', OPT_RESETCDR),
514         AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
515         AST_APP_OPTION('d', OPT_DTMF_EXIT),
516         AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
517         AST_APP_OPTION('e', OPT_PEER_H),
518         AST_APP_OPTION('f', OPT_FORCECLID),
519         AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
520         AST_APP_OPTION('g', OPT_GO_ON),
521         AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
522         AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
523         AST_APP_OPTION('H', OPT_CALLER_HANGUP),
524         AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
525         AST_APP_OPTION('k', OPT_CALLEE_PARK),
526         AST_APP_OPTION('K', OPT_CALLER_PARK),
527         AST_APP_OPTION('k', OPT_CALLEE_PARK),
528         AST_APP_OPTION('K', OPT_CALLER_PARK),
529         AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
530         AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
531         AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
532         AST_APP_OPTION('n', OPT_SCREEN_NOINTRO),
533         AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
534         AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
535         AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
536         AST_APP_OPTION('p', OPT_SCREENING),
537         AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
538         AST_APP_OPTION('r', OPT_RINGBACK),
539         AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
540         AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
541         AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
542         AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
543         AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
544         AST_APP_OPTION('W', OPT_CALLER_MONITOR),
545         AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
546         AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
547 END_OPTIONS );
548
549 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
550         OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
551         OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | OPT_CALLER_PARK) && \
552         !chan->audiohooks && !peer->audiohooks)
553
554 /*
555  * The list of active channels
556  */
557 struct chanlist {
558         struct chanlist *next;
559         struct ast_channel *chan;
560         uint64_t flags;
561 };
562
563
564 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
565 {
566         /* Hang up a tree of stuff */
567         struct chanlist *oo;
568         while (outgoing) {
569                 /* Hangup any existing lines we have open */
570                 if (outgoing->chan && (outgoing->chan != exception)) {
571                         if (answered_elsewhere) {
572                                 /* The flag is used for local channel inheritance and stuff */
573                                 ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
574                                 /* This is for the channel drivers */
575                                 outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
576                         }
577                         ast_hangup(outgoing->chan);
578                 }
579                 oo = outgoing;
580                 outgoing = outgoing->next;
581                 ast_free(oo);
582         }
583 }
584
585 #define AST_MAX_WATCHERS 256
586
587 /*
588  * argument to handle_cause() and other functions.
589  */
590 struct cause_args {
591         struct ast_channel *chan;
592         int busy;
593         int congestion;
594         int nochan;
595 };
596
597 static void handle_cause(int cause, struct cause_args *num)
598 {
599         struct ast_cdr *cdr = num->chan->cdr;
600
601         switch(cause) {
602         case AST_CAUSE_BUSY:
603                 if (cdr)
604                         ast_cdr_busy(cdr);
605                 num->busy++;
606                 break;
607
608         case AST_CAUSE_CONGESTION:
609                 if (cdr)
610                         ast_cdr_failed(cdr);
611                 num->congestion++;
612                 break;
613
614         case AST_CAUSE_NO_ROUTE_DESTINATION:
615         case AST_CAUSE_UNREGISTERED:
616                 if (cdr)
617                         ast_cdr_failed(cdr);
618                 num->nochan++;
619                 break;
620
621         case AST_CAUSE_NORMAL_CLEARING:
622                 break;
623
624         default:
625                 num->nochan++;
626                 break;
627         }
628 }
629
630 /* free the buffer if allocated, and set the pointer to the second arg */
631 #define S_REPLACE(s, new_val)           \
632         do {                            \
633                 if (s)                  \
634                         ast_free(s);    \
635                 s = (new_val);          \
636         } while (0)
637
638 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
639 {
640         char rexten[2] = { exten, '\0' };
641
642         if (context) {
643                 if (!ast_goto_if_exists(chan, context, rexten, pri))
644                         return 1;
645         } else {
646                 if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
647                         return 1;
648                 else if (!ast_strlen_zero(chan->macrocontext)) {
649                         if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
650                                 return 1;
651                 }
652         }
653         return 0;
654 }
655
656
657 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
658 {
659         const char *context = S_OR(chan->macrocontext, chan->context);
660         const char *exten = S_OR(chan->macroexten, chan->exten);
661
662         return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
663 }
664
665 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
666 {
667         manager_event(EVENT_FLAG_CALL, "Dial",
668                 "SubEvent: Begin\r\n"
669                 "Channel: %s\r\n"
670                 "Destination: %s\r\n"
671                 "CallerIDNum: %s\r\n"
672                 "CallerIDName: %s\r\n"
673                 "UniqueID: %s\r\n"
674                 "DestUniqueID: %s\r\n"
675                 "Dialstring: %s\r\n",
676                 src->name, dst->name, S_OR(src->cid.cid_num, "<unknown>"),
677                 S_OR(src->cid.cid_name, "<unknown>"), src->uniqueid,
678                 dst->uniqueid, dialstring ? dialstring : "");
679 }
680
681 static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
682 {
683         manager_event(EVENT_FLAG_CALL, "Dial",
684                 "SubEvent: End\r\n"
685                 "Channel: %s\r\n"
686                 "UniqueID: %s\r\n"
687                 "DialStatus: %s\r\n",
688                 src->name, src->uniqueid, dialstatus);
689 }
690
691 /*!
692  * helper function for wait_for_answer()
693  *
694  * XXX this code is highly suspicious, as it essentially overwrites
695  * the outgoing channel without properly deleting it.
696  */
697 static void do_forward(struct chanlist *o,
698         struct cause_args *num, struct ast_flags64 *peerflags, int single)
699 {
700         char tmpchan[256];
701         struct ast_channel *original = o->chan;
702         struct ast_channel *c = o->chan; /* the winner */
703         struct ast_channel *in = num->chan; /* the input channel */
704         char *stuff;
705         char *tech;
706         int cause;
707
708         ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
709         if ((stuff = strchr(tmpchan, '/'))) {
710                 *stuff++ = '\0';
711                 tech = tmpchan;
712         } else {
713                 const char *forward_context;
714                 ast_channel_lock(c);
715                 forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
716                 snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
717                 ast_channel_unlock(c);
718                 stuff = tmpchan;
719                 tech = "Local";
720         }
721         /* Before processing channel, go ahead and check for forwarding */
722         ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
723         /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
724         if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
725                 ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
726                 c = o->chan = NULL;
727                 cause = AST_CAUSE_BUSY;
728         } else {
729                 /* Setup parameters */
730                 c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
731                 if (c) {
732                         if (single)
733                                 ast_channel_make_compatible(o->chan, in);
734                         ast_channel_inherit_variables(in, o->chan);
735                         ast_channel_datastore_inherit(in, o->chan);
736                 } else
737                         ast_log(LOG_NOTICE, "Unable to create local channel for call forward to '%s/%s' (cause = %d)\n", tech, stuff, cause);
738         }
739         if (!c) {
740                 ast_clear_flag64(o, DIAL_STILLGOING);
741                 handle_cause(cause, num);
742                 ast_hangup(original);
743         } else {
744                 char *new_cid_num, *new_cid_name;
745                 struct ast_channel *src;
746
747                 ast_rtp_make_compatible(c, in, single);
748                 if (ast_test_flag64(o, OPT_FORCECLID)) {
749                         new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
750                         new_cid_name = NULL; /* XXX no name ? */
751                         src = c; /* XXX possible bug in previous code, which used 'winner' ? it may have changed */
752                 } else {
753                         new_cid_num = ast_strdup(in->cid.cid_num);
754                         new_cid_name = ast_strdup(in->cid.cid_name);
755                         src = in;
756                 }
757                 ast_string_field_set(c, accountcode, src->accountcode);
758                 c->cdrflags = src->cdrflags;
759                 S_REPLACE(c->cid.cid_num, new_cid_num);
760                 S_REPLACE(c->cid.cid_name, new_cid_name);
761
762                 if (in->cid.cid_ani) { /* XXX or maybe unconditional ? */
763                         S_REPLACE(c->cid.cid_ani, ast_strdup(in->cid.cid_ani));
764                 }
765                 S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(in->macroexten, in->exten)));
766                 if (ast_call(c, tmpchan, 0)) {
767                         ast_log(LOG_NOTICE, "Failed to dial on local channel for call forward to '%s'\n", tmpchan);
768                         ast_clear_flag64(o, DIAL_STILLGOING);
769                         ast_hangup(original);
770                         ast_hangup(c);
771                         c = o->chan = NULL;
772                         num->nochan++;
773                 } else {
774                         senddialevent(in, c, stuff);
775                         /* After calling, set callerid to extension */
776                         if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
777                                 char cidname[AST_MAX_EXTENSION] = "";
778                                 ast_set_callerid(c, S_OR(in->macroexten, in->exten), get_cid_name(cidname, sizeof(cidname), in), NULL);
779                         }
780                         /* Hangup the original channel now, in case we needed it */
781                         ast_hangup(original);
782                 }
783                 if (single) {
784                         ast_indicate(in, -1);
785                 }
786         }
787 }
788
789 /* argument used for some functions. */
790 struct privacy_args {
791         int sentringing;
792         int privdb_val;
793         char privcid[256];
794         char privintro[1024];
795         char status[256];
796 };
797
798 static struct ast_channel *wait_for_answer(struct ast_channel *in,
799         struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
800         struct privacy_args *pa,
801         const struct cause_args *num_in, int *result)
802 {
803         struct cause_args num = *num_in;
804         int prestart = num.busy + num.congestion + num.nochan;
805         int orig = *to;
806         struct ast_channel *peer = NULL;
807         /* single is set if only one destination is enabled */
808         int single = outgoing && !outgoing->next && !ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
809 #ifdef HAVE_EPOLL
810         struct chanlist *epollo;
811 #endif
812
813         if (single) {
814                 /* Turn off hold music, etc */
815                 ast_deactivate_generator(in);
816                 /* If we are calling a single channel, make them compatible for in-band tone purpose */
817                 ast_channel_make_compatible(outgoing->chan, in);
818         }
819
820 #ifdef HAVE_EPOLL
821         for (epollo = outgoing; epollo; epollo = epollo->next)
822                 ast_poll_channel_add(in, epollo->chan);
823 #endif
824
825         while (*to && !peer) {
826                 struct chanlist *o;
827                 int pos = 0; /* how many channels do we handle */
828                 int numlines = prestart;
829                 struct ast_channel *winner;
830                 struct ast_channel *watchers[AST_MAX_WATCHERS];
831
832                 watchers[pos++] = in;
833                 for (o = outgoing; o; o = o->next) {
834                         /* Keep track of important channels */
835                         if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
836                                 watchers[pos++] = o->chan;
837                         numlines++;
838                 }
839                 if (pos == 1) { /* only the input channel is available */
840                         if (numlines == (num.busy + num.congestion + num.nochan)) {
841                                 ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
842                                 if (num.busy)
843                                         strcpy(pa->status, "BUSY");
844                                 else if (num.congestion)
845                                         strcpy(pa->status, "CONGESTION");
846                                 else if (num.nochan)
847                                         strcpy(pa->status, "CHANUNAVAIL");
848                         } else {
849                                 ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
850                         }
851                         *to = 0;
852                         return NULL;
853                 }
854                 winner = ast_waitfor_n(watchers, pos, to);
855                 for (o = outgoing; o; o = o->next) {
856                         struct ast_frame *f;
857                         struct ast_channel *c = o->chan;
858
859                         if (c == NULL)
860                                 continue;
861                         if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
862                                 if (!peer) {
863                                         ast_verb(3, "%s answered %s\n", c->name, in->name);
864                                         peer = c;
865                                         ast_copy_flags64(peerflags, o,
866                                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
867                                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
868                                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
869                                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
870                                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
871                                                 DIAL_NOFORWARDHTML);
872                                         ast_string_field_set(c, dialcontext, "");
873                                         ast_copy_string(c->exten, "", sizeof(c->exten));
874                                 }
875                                 continue;
876                         }
877                         if (c != winner)
878                                 continue;
879                         /* here, o->chan == c == winner */
880                         if (!ast_strlen_zero(c->call_forward)) {
881                                 do_forward(o, &num, peerflags, single);
882                                 continue;
883                         }
884                         f = ast_read(winner);
885                         if (!f) {
886                                 in->hangupcause = c->hangupcause;
887 #ifdef HAVE_EPOLL
888                                 ast_poll_channel_del(in, c);
889 #endif
890                                 ast_hangup(c);
891                                 c = o->chan = NULL;
892                                 ast_clear_flag64(o, DIAL_STILLGOING);
893                                 handle_cause(in->hangupcause, &num);
894                                 continue;
895                         }
896                         if (f->frametype == AST_FRAME_CONTROL) {
897                                 switch(f->subclass) {
898                                 case AST_CONTROL_ANSWER:
899                                         /* This is our guy if someone answered. */
900                                         if (!peer) {
901                                                 ast_verb(3, "%s answered %s\n", c->name, in->name);
902                                                 peer = c;
903                                                 if (peer->cdr) {
904                                                         peer->cdr->answer = ast_tvnow();
905                                                         peer->cdr->disposition = AST_CDR_ANSWERED;
906                                                 }
907                                                 ast_copy_flags64(peerflags, o,
908                                                         OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
909                                                         OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
910                                                         OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
911                                                         OPT_CALLEE_PARK | OPT_CALLER_PARK |
912                                                         OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
913                                                         DIAL_NOFORWARDHTML);
914                                                 ast_string_field_set(c, dialcontext, "");
915                                                 ast_copy_string(c->exten, "", sizeof(c->exten));
916                                                 if (CAN_EARLY_BRIDGE(peerflags, in, peer))
917                                                         /* Setup early bridge if appropriate */
918                                                         ast_channel_early_bridge(in, peer);
919                                         }
920                                         /* If call has been answered, then the eventual hangup is likely to be normal hangup */
921                                         in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
922                                         c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
923                                         break;
924                                 case AST_CONTROL_BUSY:
925                                         ast_verb(3, "%s is busy\n", c->name);
926                                         in->hangupcause = c->hangupcause;
927                                         ast_hangup(c);
928                                         c = o->chan = NULL;
929                                         ast_clear_flag64(o, DIAL_STILLGOING);
930                                         handle_cause(AST_CAUSE_BUSY, &num);
931                                         break;
932                                 case AST_CONTROL_CONGESTION:
933                                         ast_verb(3, "%s is circuit-busy\n", c->name);
934                                         in->hangupcause = c->hangupcause;
935                                         ast_hangup(c);
936                                         c = o->chan = NULL;
937                                         ast_clear_flag64(o, DIAL_STILLGOING);
938                                         handle_cause(AST_CAUSE_CONGESTION, &num);
939                                         break;
940                                 case AST_CONTROL_RINGING:
941                                         ast_verb(3, "%s is ringing\n", c->name);
942                                         /* Setup early media if appropriate */
943                                         if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
944                                                 ast_channel_early_bridge(in, c);
945                                         if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
946                                                 ast_indicate(in, AST_CONTROL_RINGING);
947                                                 pa->sentringing++;
948                                         }
949                                         break;
950                                 case AST_CONTROL_PROGRESS:
951                                         ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
952                                         /* Setup early media if appropriate */
953                                         if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
954                                                 ast_channel_early_bridge(in, c);
955                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK))
956                                                 ast_indicate(in, AST_CONTROL_PROGRESS);
957                                         break;
958                                 case AST_CONTROL_VIDUPDATE:
959                                         ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
960                                         ast_indicate(in, AST_CONTROL_VIDUPDATE);
961                                         break;
962                                 case AST_CONTROL_SRCUPDATE:
963                                         ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
964                                         ast_indicate(in, AST_CONTROL_SRCUPDATE);
965                                         break;
966                                 case AST_CONTROL_PROCEEDING:
967                                         ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
968                                         if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
969                                                 ast_channel_early_bridge(in, c);
970                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK))
971                                                 ast_indicate(in, AST_CONTROL_PROCEEDING);
972                                         break;
973                                 case AST_CONTROL_HOLD:
974                                         ast_verb(3, "Call on %s placed on hold\n", c->name);
975                                         ast_indicate(in, AST_CONTROL_HOLD);
976                                         break;
977                                 case AST_CONTROL_UNHOLD:
978                                         ast_verb(3, "Call on %s left from hold\n", c->name);
979                                         ast_indicate(in, AST_CONTROL_UNHOLD);
980                                         break;
981                                 case AST_CONTROL_OFFHOOK:
982                                 case AST_CONTROL_FLASH:
983                                         /* Ignore going off hook and flash */
984                                         break;
985                                 case -1:
986                                         if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
987                                                 ast_verb(3, "%s stopped sounds\n", c->name);
988                                                 ast_indicate(in, -1);
989                                                 pa->sentringing = 0;
990                                         }
991                                         break;
992                                 default:
993                                         ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
994                                 }
995                         } else if (single) {
996                                 switch (f->frametype) {
997                                         case AST_FRAME_VOICE:
998                                         case AST_FRAME_IMAGE:
999                                         case AST_FRAME_TEXT:
1000                                                 if (ast_write(in, f)) {
1001                                                         ast_log(LOG_WARNING, "Unable to write frame\n");
1002                                                 }
1003                                                 break;
1004                                         case AST_FRAME_HTML:
1005                                                 if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass, f->data.ptr, f->datalen) == -1) {
1006                                                         ast_log(LOG_WARNING, "Unable to send URL\n");
1007                                                 }
1008                                                 break;
1009                                         default:
1010                                                 break;
1011                                 }
1012                         }
1013                         ast_frfree(f);
1014                 } /* end for */
1015                 if (winner == in) {
1016                         struct ast_frame *f = ast_read(in);
1017 #if 0
1018                         if (f && (f->frametype != AST_FRAME_VOICE))
1019                                 printf("Frame type: %d, %d\n", f->frametype, f->subclass);
1020                         else if (!f || (f->frametype != AST_FRAME_VOICE))
1021                                 printf("Hangup received on %s\n", in->name);
1022 #endif
1023                         if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
1024                                 /* Got hung up */
1025                                 *to = -1;
1026                                 strcpy(pa->status, "CANCEL");
1027                                 ast_cdr_noanswer(in->cdr);
1028                                 if (f) {
1029                                         if (f->data.uint32) {
1030                                                 in->hangupcause = f->data.uint32;
1031                                         }
1032                                         ast_frfree(f);
1033                                 }
1034                                 return NULL;
1035                         }
1036
1037                         /* now f is guaranteed non-NULL */
1038                         if (f->frametype == AST_FRAME_DTMF) {
1039                                 if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
1040                                         const char *context;
1041                                         ast_channel_lock(in);
1042                                         context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
1043                                         if (onedigit_goto(in, context, (char) f->subclass, 1)) {
1044                                                 ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
1045                                                 *to = 0;
1046                                                 ast_cdr_noanswer(in->cdr);
1047                                                 *result = f->subclass;
1048                                                 strcpy(pa->status, "CANCEL");
1049                                                 ast_frfree(f);
1050                                                 ast_channel_unlock(in);
1051                                                 return NULL;
1052                                         }
1053                                         ast_channel_unlock(in);
1054                                 }
1055
1056                                 if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
1057                                                 (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
1058                                         ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
1059                                         *to = 0;
1060                                         strcpy(pa->status, "CANCEL");
1061                                         ast_cdr_noanswer(in->cdr);
1062                                         ast_frfree(f);
1063                                         return NULL;
1064                                 }
1065                         }
1066
1067                         /* Forward HTML stuff */
1068                         if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
1069                                 if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data.ptr, f->datalen) == -1)
1070                                         ast_log(LOG_WARNING, "Unable to send URL\n");
1071
1072                         if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END)))  {
1073                                 if (ast_write(outgoing->chan, f))
1074                                         ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
1075                         }
1076                         if (single && (f->frametype == AST_FRAME_CONTROL) &&
1077                                 ((f->subclass == AST_CONTROL_HOLD) ||
1078                                 (f->subclass == AST_CONTROL_UNHOLD) ||
1079                                 (f->subclass == AST_CONTROL_VIDUPDATE) ||
1080                                  (f->subclass == AST_CONTROL_SRCUPDATE))) {
1081                                 ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
1082                                 ast_indicate_data(outgoing->chan, f->subclass, f->data.ptr, f->datalen);
1083                         }
1084                         ast_frfree(f);
1085                 }
1086                 if (!*to)
1087                         ast_verb(3, "Nobody picked up in %d ms\n", orig);
1088                 if (!*to || ast_check_hangup(in))
1089                         ast_cdr_noanswer(in->cdr);
1090         }
1091
1092 #ifdef HAVE_EPOLL
1093         for (epollo = outgoing; epollo; epollo = epollo->next) {
1094                 if (epollo->chan)
1095                         ast_poll_channel_del(in, epollo->chan);
1096         }
1097 #endif
1098
1099         return peer;
1100 }
1101
1102 static void replace_macro_delimiter(char *s)
1103 {
1104         for (; *s; s++)
1105                 if (*s == '^')
1106                         *s = ',';
1107 }
1108
1109 /* returns true if there is a valid privacy reply */
1110 static int valid_priv_reply(struct ast_flags64 *opts, int res)
1111 {
1112         if (res < '1')
1113                 return 0;
1114         if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
1115                 return 1;
1116         if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
1117                 return 1;
1118         return 0;
1119 }
1120
1121 static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
1122         char *parse, struct timeval *calldurationlimit)
1123 {
1124         char *stringp = ast_strdupa(parse);
1125         char *limit_str, *warning_str, *warnfreq_str;
1126         const char *var;
1127         int play_to_caller = 0, play_to_callee = 0;
1128         int delta;
1129
1130         limit_str = strsep(&stringp, ":");
1131         warning_str = strsep(&stringp, ":");
1132         warnfreq_str = strsep(&stringp, ":");
1133
1134         config->timelimit = atol(limit_str);
1135         if (warning_str)
1136                 config->play_warning = atol(warning_str);
1137         if (warnfreq_str)
1138                 config->warning_freq = atol(warnfreq_str);
1139
1140         if (!config->timelimit) {
1141                 ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
1142                 config->timelimit = config->play_warning = config->warning_freq = 0;
1143                 config->warning_sound = NULL;
1144                 return -1; /* error */
1145         } else if ( (delta = config->play_warning - config->timelimit) > 0) {
1146                 int w = config->warning_freq;
1147
1148                 /* If the first warning is requested _after_ the entire call would end,
1149                    and no warning frequency is requested, then turn off the warning. If
1150                    a warning frequency is requested, reduce the 'first warning' time by
1151                    that frequency until it falls within the call's total time limit.
1152                    Graphically:
1153                                   timelim->|    delta        |<-playwarning
1154                         0__________________|_________________|
1155                                          | w  |    |    |    |
1156
1157                    so the number of intervals to cut is 1+(delta-1)/w
1158                 */
1159
1160                 if (w == 0) {
1161                         config->play_warning = 0;
1162                 } else {
1163                         config->play_warning -= w * ( 1 + (delta-1)/w );
1164                         if (config->play_warning < 1)
1165                                 config->play_warning = config->warning_freq = 0;
1166                 }
1167         }
1168         
1169         ast_channel_lock(chan);
1170
1171         var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
1172
1173         play_to_caller = var ? ast_true(var) : 1;
1174
1175         var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
1176         play_to_callee = var ? ast_true(var) : 0;
1177
1178         if (!play_to_caller && !play_to_callee)
1179                 play_to_caller = 1;
1180
1181         var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
1182         config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
1183
1184         /* The code looking at config wants a NULL, not just "", to decide
1185          * that the message should not be played, so we replace "" with NULL.
1186          * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
1187          * not found.
1188          */
1189
1190         var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
1191         config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1192
1193         var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
1194         config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
1195
1196         ast_channel_unlock(chan);
1197
1198         /* undo effect of S(x) in case they are both used */
1199         calldurationlimit->tv_sec = 0;
1200         calldurationlimit->tv_usec = 0;
1201
1202         /* more efficient to do it like S(x) does since no advanced opts */
1203         if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
1204                 calldurationlimit->tv_sec = config->timelimit / 1000;
1205                 calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
1206                 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
1207                         calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
1208                 config->timelimit = play_to_caller = play_to_callee =
1209                 config->play_warning = config->warning_freq = 0;
1210         } else {
1211                 ast_verb(3, "Limit Data for this call:\n");
1212                 ast_verb(4, "timelimit      = %ld\n", config->timelimit);
1213                 ast_verb(4, "play_warning   = %ld\n", config->play_warning);
1214                 ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
1215                 ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
1216                 ast_verb(4, "warning_freq   = %ld\n", config->warning_freq);
1217                 ast_verb(4, "start_sound    = %s\n", S_OR(config->start_sound, ""));
1218                 ast_verb(4, "warning_sound  = %s\n", config->warning_sound);
1219                 ast_verb(4, "end_sound      = %s\n", S_OR(config->end_sound, ""));
1220         }
1221         if (play_to_caller)
1222                 ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
1223         if (play_to_callee)
1224                 ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
1225         return 0;
1226 }
1227
1228 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
1229         struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
1230 {
1231
1232         int res2;
1233         int loopcount = 0;
1234
1235         /* Get the user's intro, store it in priv-callerintros/$CID,
1236            unless it is already there-- this should be done before the
1237            call is actually dialed  */
1238
1239         /* all ring indications and moh for the caller has been halted as soon as the
1240            target extension was picked up. We are going to have to kill some
1241            time and make the caller believe the peer hasn't picked up yet */
1242
1243         if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1244                 char *original_moh = ast_strdupa(chan->musicclass);
1245                 ast_indicate(chan, -1);
1246                 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1247                 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1248                 ast_string_field_set(chan, musicclass, original_moh);
1249         } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1250                 ast_indicate(chan, AST_CONTROL_RINGING);
1251                 pa->sentringing++;
1252         }
1253
1254         /* Start autoservice on the other chan ?? */
1255         res2 = ast_autoservice_start(chan);
1256         /* Now Stream the File */
1257         for (loopcount = 0; loopcount < 3; loopcount++) {
1258                 if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
1259                         break;
1260                 if (!res2) /* on timeout, play the message again */
1261                         res2 = ast_play_and_wait(peer, "priv-callpending");
1262                 if (!valid_priv_reply(opts, res2))
1263                         res2 = 0;
1264                 /* priv-callpending script:
1265                    "I have a caller waiting, who introduces themselves as:"
1266                 */
1267                 if (!res2)
1268                         res2 = ast_play_and_wait(peer, pa->privintro);
1269                 if (!valid_priv_reply(opts, res2))
1270                         res2 = 0;
1271                 /* now get input from the called party, as to their choice */
1272                 if (!res2) {
1273                         /* XXX can we have both, or they are mutually exclusive ? */
1274                         if (ast_test_flag64(opts, OPT_PRIVACY))
1275                                 res2 = ast_play_and_wait(peer, "priv-callee-options");
1276                         if (ast_test_flag64(opts, OPT_SCREENING))
1277                                 res2 = ast_play_and_wait(peer, "screen-callee-options");
1278                 }
1279                 /*! \page DialPrivacy Dial Privacy scripts
1280                 \par priv-callee-options script:
1281                         "Dial 1 if you wish this caller to reach you directly in the future,
1282                                 and immediately connect to their incoming call
1283                          Dial 2 if you wish to send this caller to voicemail now and
1284                                 forevermore.
1285                          Dial 3 to send this caller to the torture menus, now and forevermore.
1286                          Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
1287                          Dial 5 to allow this caller to come straight thru to you in the future,
1288                                 but right now, just this once, send them to voicemail."
1289                 \par screen-callee-options script:
1290                         "Dial 1 if you wish to immediately connect to the incoming call
1291                          Dial 2 if you wish to send this caller to voicemail.
1292                          Dial 3 to send this caller to the torture menus.
1293                          Dial 4 to send this caller to a simple "go away" menu.
1294                 */
1295                 if (valid_priv_reply(opts, res2))
1296                         break;
1297                 /* invalid option */
1298                 res2 = ast_play_and_wait(peer, "vm-sorry");
1299         }
1300
1301         if (ast_test_flag64(opts, OPT_MUSICBACK)) {
1302                 ast_moh_stop(chan);
1303         } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
1304                 ast_indicate(chan, -1);
1305                 pa->sentringing = 0;
1306         }
1307         ast_autoservice_stop(chan);
1308         if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
1309                 /* map keypresses to various things, the index is res2 - '1' */
1310                 static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
1311                 static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
1312                 int i = res2 - '1';
1313                 ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
1314                         opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
1315                 ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
1316         }
1317         switch (res2) {
1318         case '1':
1319                 break;
1320         case '2':
1321                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1322                 break;
1323         case '3':
1324                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1325                 break;
1326         case '4':
1327                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1328                 break;
1329         case '5':
1330                 /* XXX should we set status to DENY ? */
1331                 if (ast_test_flag64(opts, OPT_PRIVACY))
1332                         break;
1333                 /* if not privacy, then 5 is the same as "default" case */
1334         default: /* bad input or -1 if failure to start autoservice */
1335                 /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do?  */
1336                 /* well, there seems basically two choices. Just patch the caller thru immediately,
1337                           or,... put 'em thru to voicemail. */
1338                 /* since the callee may have hung up, let's do the voicemail thing, no database decision */
1339                 ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
1340                 /* XXX should we set status to DENY ? */
1341                 /* XXX what about the privacy flags ? */
1342                 break;
1343         }
1344
1345         if (res2 == '1') { /* the only case where we actually connect */
1346                 /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
1347                    just clog things up, and it's not useful information, not being tied to a CID */
1348                 if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
1349                         ast_filedelete(pa->privintro, NULL);
1350                         if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1351                                 ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1352                         else
1353                                 ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1354                 }
1355                 return 0; /* the good exit path */
1356         } else {
1357                 ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
1358                 return -1;
1359         }
1360 }
1361
1362 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
1363 static int setup_privacy_args(struct privacy_args *pa,
1364         struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
1365 {
1366         char callerid[60];
1367         int res;
1368         char *l;
1369         int silencethreshold;
1370
1371         if (!ast_strlen_zero(chan->cid.cid_num)) {
1372                 l = ast_strdupa(chan->cid.cid_num);
1373                 ast_shrink_phone_number(l);
1374                 if (ast_test_flag64(opts, OPT_PRIVACY) ) {
1375                         ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
1376                         pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
1377                 } else {
1378                         ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
1379                         pa->privdb_val = AST_PRIVACY_UNKNOWN;
1380                 }
1381         } else {
1382                 char *tnam, *tn2;
1383
1384                 tnam = ast_strdupa(chan->name);
1385                 /* clean the channel name so slashes don't try to end up in disk file name */
1386                 for (tn2 = tnam; *tn2; tn2++) {
1387                         if (*tn2 == '/')  /* any other chars to be afraid of? */
1388                                 *tn2 = '=';
1389                 }
1390                 ast_verb(3, "Privacy-- callerid is empty\n");
1391
1392                 snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
1393                 l = callerid;
1394                 pa->privdb_val = AST_PRIVACY_UNKNOWN;
1395         }
1396
1397         ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
1398
1399         if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCLID)) {
1400                 /* if callerid is set and OPT_SCREEN_NOCLID is set also */
1401                 ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
1402                 pa->privdb_val = AST_PRIVACY_ALLOW;
1403         } else if (ast_test_flag64(opts, OPT_SCREEN_NOCLID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
1404                 ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
1405         }
1406         
1407         if (pa->privdb_val == AST_PRIVACY_DENY) {
1408                 ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
1409                 ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
1410                 return 0;
1411         } else if (pa->privdb_val == AST_PRIVACY_KILL) {
1412                 ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
1413                 return 0; /* Is this right? */
1414         } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
1415                 ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
1416                 return 0; /* is this right??? */
1417         } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
1418                 /* Get the user's intro, store it in priv-callerintros/$CID,
1419                    unless it is already there-- this should be done before the
1420                    call is actually dialed  */
1421
1422                 /* make sure the priv-callerintros dir actually exists */
1423                 snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
1424                 if ((res = ast_mkdir(pa->privintro, 0755))) {
1425                         ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
1426                         return -1;
1427                 }
1428
1429                 snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
1430                 if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
1431                         /* the DELUX version of this code would allow this caller the
1432                            option to hear and retape their previously recorded intro.
1433                         */
1434                 } else {
1435                         int duration; /* for feedback from play_and_wait */
1436                         /* the file doesn't exist yet. Let the caller submit his
1437                            vocal intro for posterity */
1438                         /* priv-recordintro script:
1439
1440                            "At the tone, please say your name:"
1441
1442                         */
1443                         silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
1444                         ast_answer(chan);
1445                         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 */
1446                                                                         /* don't think we'll need a lock removed, we took care of
1447                                                                            conflicts by naming the pa.privintro file */
1448                         if (res == -1) {
1449                                 /* Delete the file regardless since they hung up during recording */
1450                                 ast_filedelete(pa->privintro, NULL);
1451                                 if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
1452                                         ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
1453                                 else
1454                                         ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
1455                                 return -1;
1456                         }
1457                         if (!ast_streamfile(chan, "vm-dialout", chan->language) )
1458                                 ast_waitstream(chan, "");
1459                 }
1460         }
1461         return 1; /* success */
1462 }
1463
1464 static void end_bridge_callback(void *data)
1465 {
1466         char buf[80];
1467         time_t end;
1468         struct ast_channel *chan = data;
1469
1470         if (!chan->cdr) {
1471                 return;
1472         }
1473
1474         time(&end);
1475
1476         ast_channel_lock(chan);
1477         if (chan->cdr->answer.tv_sec) {
1478                 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->answer.tv_sec);
1479                 pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
1480         }
1481
1482         if (chan->cdr->start.tv_sec) {
1483                 snprintf(buf, sizeof(buf), "%ld", end - chan->cdr->start.tv_sec);
1484                 pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
1485         }
1486         ast_channel_unlock(chan);
1487 }
1488
1489 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
1490         bconfig->end_bridge_callback_data = originator;
1491 }
1492
1493 static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
1494 {
1495         int res = -1; /* default: error */
1496         char *rest, *cur; /* scan the list of destinations */
1497         struct chanlist *outgoing = NULL; /* list of destinations */
1498         struct ast_channel *peer;
1499         int to; /* timeout */
1500         struct cause_args num = { chan, 0, 0, 0 };
1501         int cause;
1502         char numsubst[256];
1503         char cidname[AST_MAX_EXTENSION] = "";
1504
1505         struct ast_bridge_config config = { { 0, } };
1506         struct timeval calldurationlimit = { 0, };
1507         char *dtmfcalled = NULL, *dtmfcalling = NULL;
1508         struct privacy_args pa = {
1509                 .sentringing = 0,
1510                 .privdb_val = 0,
1511                 .status = "INVALIDARGS",
1512         };
1513         int sentringing = 0, moh = 0;
1514         const char *outbound_group = NULL;
1515         int result = 0;
1516         char *parse;
1517         int opermode = 0;
1518         AST_DECLARE_APP_ARGS(args,
1519                 AST_APP_ARG(peers);
1520                 AST_APP_ARG(timeout);
1521                 AST_APP_ARG(options);
1522                 AST_APP_ARG(url);
1523         );
1524         struct ast_flags64 opts = { 0, };
1525         char *opt_args[OPT_ARG_ARRAY_SIZE];
1526         struct ast_datastore *datastore = NULL;
1527         int fulldial = 0, num_dialed = 0;
1528
1529         /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
1530         pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
1531         pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
1532         pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
1533         pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
1534         pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
1535
1536         if (ast_strlen_zero(data)) {
1537                 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1538                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1539                 return -1;
1540         }
1541
1542         parse = ast_strdupa(data);
1543
1544         AST_STANDARD_APP_ARGS(args, parse);
1545
1546         if (!ast_strlen_zero(args.options) &&
1547                 ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
1548                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1549                 goto done;
1550         }
1551
1552         if (ast_strlen_zero(args.peers)) {
1553                 ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
1554                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1555                 goto done;
1556         }
1557
1558
1559         if (ast_test_flag64(&opts, OPT_OPERMODE)) {
1560                 opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
1561                 ast_verb(3, "Setting operator services mode to %d.\n", opermode);
1562         }
1563         
1564         if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
1565                 calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
1566                 if (!calldurationlimit.tv_sec) {
1567                         ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
1568                         pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1569                         goto done;
1570                 }
1571                 ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
1572         }
1573
1574         if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
1575                 dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
1576                 dtmfcalled = strsep(&dtmfcalling, ":");
1577         }
1578
1579         if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
1580                 if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
1581                         goto done;
1582         }
1583
1584         if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
1585                 ast_cdr_reset(chan->cdr, NULL);
1586         if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
1587                 opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
1588
1589         if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
1590                 res = setup_privacy_args(&pa, &opts, opt_args, chan);
1591                 if (res <= 0)
1592                         goto out;
1593                 res = -1; /* reset default */
1594         }
1595
1596         if (continue_exec)
1597                 *continue_exec = 0;
1598
1599         /* If a channel group has been specified, get it for use when we create peer channels */
1600
1601         ast_channel_lock(chan);
1602         if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
1603                 outbound_group = ast_strdupa(outbound_group);   
1604                 pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
1605         } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
1606                 outbound_group = ast_strdupa(outbound_group);
1607         }
1608         ast_channel_unlock(chan);       
1609         ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING);
1610
1611         /* loop through the list of dial destinations */
1612         rest = args.peers;
1613         while ((cur = strsep(&rest, "&")) ) {
1614                 struct chanlist *tmp;
1615                 struct ast_channel *tc; /* channel for this destination */
1616                 /* Get a technology/[device:]number pair */
1617                 char *number = cur;
1618                 char *interface = ast_strdupa(number);
1619                 char *tech = strsep(&number, "/");
1620                 /* find if we already dialed this interface */
1621                 struct ast_dialed_interface *di;
1622                 AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
1623                 num_dialed++;
1624                 if (!number) {
1625                         ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
1626                         goto out;
1627                 }
1628                 if (!(tmp = ast_calloc(1, sizeof(*tmp))))
1629                         goto out;
1630                 if (opts.flags) {
1631                         ast_copy_flags64(tmp, &opts,
1632                                 OPT_CANCEL_ELSEWHERE |
1633                                 OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
1634                                 OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
1635                                 OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
1636                                 OPT_CALLEE_PARK | OPT_CALLER_PARK |
1637                                 OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
1638                                 OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
1639                         ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
1640                 }
1641                 ast_copy_string(numsubst, number, sizeof(numsubst));
1642                 /* Request the peer */
1643
1644                 ast_channel_lock(chan);
1645                 datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
1646                 ast_channel_unlock(chan);
1647
1648                 if (datastore)
1649                         dialed_interfaces = datastore->data;
1650                 else {
1651                         if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
1652                                 ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
1653                                 ast_free(tmp);
1654                                 goto out;
1655                         }
1656
1657                         datastore->inheritance = DATASTORE_INHERIT_FOREVER;
1658
1659                         if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
1660                                 ast_free(tmp);
1661                                 goto out;
1662                         }
1663
1664                         datastore->data = dialed_interfaces;
1665                         AST_LIST_HEAD_INIT(dialed_interfaces);
1666
1667                         ast_channel_lock(chan);
1668                         ast_channel_datastore_add(chan, datastore);
1669                         ast_channel_unlock(chan);
1670                 }
1671
1672                 AST_LIST_LOCK(dialed_interfaces);
1673                 AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
1674                         if (!strcasecmp(di->interface, interface)) {
1675                                 ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
1676                                         di->interface);
1677                                 break;
1678                         }
1679                 }
1680                 AST_LIST_UNLOCK(dialed_interfaces);
1681
1682                 if (di) {
1683                         fulldial++;
1684                         ast_free(tmp);
1685                         continue;
1686                 }
1687
1688                 /* It is always ok to dial a Local interface.  We only keep track of
1689                  * which "real" interfaces have been dialed.  The Local channel will
1690                  * inherit this list so that if it ends up dialing a real interface,
1691                  * it won't call one that has already been called. */
1692                 if (strcasecmp(tech, "Local")) {
1693                         if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
1694                                 AST_LIST_UNLOCK(dialed_interfaces);
1695                                 ast_free(tmp);
1696                                 goto out;
1697                         }
1698                         strcpy(di->interface, interface);
1699
1700                         AST_LIST_LOCK(dialed_interfaces);
1701                         AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
1702                         AST_LIST_UNLOCK(dialed_interfaces);
1703                 }
1704
1705                 tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
1706                 if (!tc) {
1707                         /* If we can't, just go on to the next call */
1708                         ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
1709                                 tech, cause, ast_cause2str(cause));
1710                         handle_cause(cause, &num);
1711                         if (!rest) /* we are on the last destination */
1712                                 chan->hangupcause = cause;
1713                         ast_free(tmp);
1714                         continue;
1715                 }
1716                 pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
1717
1718                 /* Setup outgoing SDP to match incoming one */
1719                 ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
1720                 
1721                 /* Inherit specially named variables from parent channel */
1722                 ast_channel_inherit_variables(chan, tc);
1723                 ast_channel_datastore_inherit(chan, tc);
1724
1725                 tc->appl = "AppDial";
1726                 tc->data = "(Outgoing Line)";
1727                 memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
1728
1729                 S_REPLACE(tc->cid.cid_num, ast_strdup(chan->cid.cid_num));
1730                 S_REPLACE(tc->cid.cid_name, ast_strdup(chan->cid.cid_name));
1731                 S_REPLACE(tc->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
1732                 S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
1733                 
1734                 ast_string_field_set(tc, accountcode, chan->accountcode);
1735                 tc->cdrflags = chan->cdrflags;
1736                 if (ast_strlen_zero(tc->musicclass))
1737                         ast_string_field_set(tc, musicclass, chan->musicclass);
1738                 /* Pass callingpres, type of number, tns, ADSI CPE, transfer capability */
1739                 tc->cid.cid_pres = chan->cid.cid_pres;
1740                 tc->cid.cid_ton = chan->cid.cid_ton;
1741                 tc->cid.cid_tns = chan->cid.cid_tns;
1742                 tc->cid.cid_ani2 = chan->cid.cid_ani2;
1743                 tc->adsicpe = chan->adsicpe;
1744                 tc->transfercapability = chan->transfercapability;
1745
1746                 /* If we have an outbound group, set this peer channel to it */
1747                 if (outbound_group)
1748                         ast_app_group_set_channel(tc, outbound_group);
1749                 /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
1750                 if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
1751                         ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1752
1753                 /* Check if we're forced by configuration */
1754                 if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
1755                          ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
1756
1757
1758                 /* Inherit context and extension */
1759                 ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
1760                 if (!ast_strlen_zero(chan->macroexten))
1761                         ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
1762                 else
1763                         ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
1764
1765                 res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
1766
1767                 /* Save the info in cdr's that we called them */
1768                 if (chan->cdr)
1769                         ast_cdr_setdestchan(chan->cdr, tc->name);
1770
1771                 /* check the results of ast_call */
1772                 if (res) {
1773                         /* Again, keep going even if there's an error */
1774                         ast_debug(1, "ast call on peer returned %d\n", res);
1775                         ast_verb(3, "Couldn't call %s\n", numsubst);
1776                         if (tc->hangupcause) {
1777                                 chan->hangupcause = tc->hangupcause;
1778                         }
1779                         ast_hangup(tc);
1780                         tc = NULL;
1781                         ast_free(tmp);
1782                         continue;
1783                 } else {
1784                         senddialevent(chan, tc, numsubst);
1785                         ast_verb(3, "Called %s\n", numsubst);
1786                         if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID))
1787                                 ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), get_cid_name(cidname, sizeof(cidname), chan), NULL);
1788                 }
1789                 /* Put them in the list of outgoing thingies...  We're ready now.
1790                    XXX If we're forcibly removed, these outgoing calls won't get
1791                    hung up XXX */
1792                 ast_set_flag64(tmp, DIAL_STILLGOING);
1793                 tmp->chan = tc;
1794                 tmp->next = outgoing;
1795                 outgoing = tmp;
1796                 /* If this line is up, don't try anybody else */
1797                 if (outgoing->chan->_state == AST_STATE_UP)
1798                         break;
1799         }
1800         
1801         if (ast_strlen_zero(args.timeout)) {
1802                 to = -1;
1803         } else {
1804                 to = atoi(args.timeout);
1805                 if (to > 0)
1806                         to *= 1000;
1807                 else {
1808                         ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
1809                         to = -1;
1810                 }
1811         }
1812
1813         if (!outgoing) {
1814                 strcpy(pa.status, "CHANUNAVAIL");
1815                 if (fulldial == num_dialed) {
1816                         res = -1;
1817                         goto out;
1818                 }
1819         } else {
1820                 /* Our status will at least be NOANSWER */
1821                 strcpy(pa.status, "NOANSWER");
1822                 if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
1823                         moh = 1;
1824                         if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
1825                                 char *original_moh = ast_strdupa(chan->musicclass);
1826                                 ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
1827                                 ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
1828                                 ast_string_field_set(chan, musicclass, original_moh);
1829                         } else {
1830                                 ast_moh_start(chan, NULL, NULL);
1831                         }
1832                         ast_indicate(chan, AST_CONTROL_PROGRESS);
1833                 } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
1834                         ast_indicate(chan, AST_CONTROL_RINGING);
1835                         sentringing++;
1836                 }
1837         }
1838
1839         peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result);
1840
1841         /* The ast_channel_datastore_remove() function could fail here if the
1842          * datastore was moved to another channel during a masquerade. If this is
1843          * the case, don't free the datastore here because later, when the channel
1844          * to which the datastore was moved hangs up, it will attempt to free this
1845          * datastore again, causing a crash
1846          */
1847         if (!ast_channel_datastore_remove(chan, datastore))
1848                 ast_datastore_free(datastore);
1849         if (!peer) {
1850                 if (result) {
1851                         res = result;
1852                 } else if (to) { /* Musta gotten hung up */
1853                         res = -1;
1854                 } else { /* Nobody answered, next please? */
1855                         res = 0;
1856                 }
1857
1858                 /* SIP, in particular, sends back this error code to indicate an
1859                  * overlap dialled number needs more digits. */
1860                 if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
1861                         res = AST_PBX_INCOMPLETE;
1862                 }
1863
1864                 /* almost done, although the 'else' block is 400 lines */
1865         } else {
1866                 const char *number;
1867
1868                 strcpy(pa.status, "ANSWER");
1869                 pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
1870                 /* Ah ha!  Someone answered within the desired timeframe.  Of course after this
1871                    we will always return with -1 so that it is hung up properly after the
1872                    conversation.  */
1873                 hanguptree(outgoing, peer, 1);
1874                 outgoing = NULL;
1875                 /* If appropriate, log that we have a destination channel */
1876                 if (chan->cdr)
1877                         ast_cdr_setdestchan(chan->cdr, peer->name);
1878                 if (peer->name)
1879                         pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
1880                 
1881                 ast_channel_lock(peer);
1882                 number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER"); 
1883                 if (!number)
1884                         number = numsubst;
1885                 pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
1886                 ast_channel_unlock(peer);
1887
1888                 if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
1889                         ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
1890                         ast_channel_sendurl( peer, args.url );
1891                 }
1892                 if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
1893                         if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
1894                                 res = 0;
1895                                 goto out;
1896                         }
1897                 }
1898                 if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
1899                         res = 0;
1900                 } else {
1901                         int digit = 0;
1902                         /* Start autoservice on the other chan */
1903                         res = ast_autoservice_start(chan);
1904                         /* Now Stream the File */
1905                         if (!res)
1906                                 res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
1907                         if (!res) {
1908                                 digit = ast_waitstream(peer, AST_DIGIT_ANY);
1909                         }
1910                         /* Ok, done. stop autoservice */
1911                         res = ast_autoservice_stop(chan);
1912                         if (digit > 0 && !res)
1913                                 res = ast_senddigit(chan, digit, 0);
1914                         else
1915                                 res = digit;
1916
1917                 }
1918
1919                 if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
1920                         replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
1921                         ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
1922                         /* peer goes to the same context and extension as chan, so just copy info from chan*/
1923                         ast_copy_string(peer->context, chan->context, sizeof(peer->context));
1924                         ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
1925                         peer->priority = chan->priority + 2;
1926                         ast_pbx_start(peer);
1927                         hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
1928                         if (continue_exec)
1929                                 *continue_exec = 1;
1930                         res = 0;
1931                         goto done;
1932                 }
1933
1934                 if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
1935                         struct ast_app *theapp;
1936                         const char *macro_result;
1937
1938                         res = ast_autoservice_start(chan);
1939                         if (res) {
1940                                 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
1941                                 res = -1;
1942                         }
1943
1944                         theapp = pbx_findapp("Macro");
1945
1946                         if (theapp && !res) { /* XXX why check res here ? */
1947                                 /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
1948                                 ast_copy_string(peer->context, chan->context, sizeof(peer->context));
1949                                 ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
1950
1951                                 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
1952                                 res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
1953                                 ast_debug(1, "Macro exited with status %d\n", res);
1954                                 res = 0;
1955                         } else {
1956                                 ast_log(LOG_ERROR, "Could not find application Macro\n");
1957                                 res = -1;
1958                         }
1959
1960                         if (ast_autoservice_stop(chan) < 0) {
1961                                 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
1962                                 res = -1;
1963                         }
1964
1965                         ast_channel_lock(peer);
1966
1967                         if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
1968                                 char *macro_transfer_dest;
1969
1970                                 if (!strcasecmp(macro_result, "BUSY")) {
1971                                         ast_copy_string(pa.status, macro_result, sizeof(pa.status));
1972                                         ast_set_flag64(peerflags, OPT_GO_ON);
1973                                         res = -1;
1974                                 } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
1975                                         ast_copy_string(pa.status, macro_result, sizeof(pa.status));
1976                                         ast_set_flag64(peerflags, OPT_GO_ON);
1977                                         res = -1;
1978                                 } else if (!strcasecmp(macro_result, "CONTINUE")) {
1979                                         /* hangup peer and keep chan alive assuming the macro has changed
1980                                            the context / exten / priority or perhaps
1981                                            the next priority in the current exten is desired.
1982                                         */
1983                                         ast_set_flag64(peerflags, OPT_GO_ON);
1984                                         res = -1;
1985                                 } else if (!strcasecmp(macro_result, "ABORT")) {
1986                                         /* Hangup both ends unless the caller has the g flag */
1987                                         res = -1;
1988                                 } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
1989                                         res = -1;
1990                                         /* perform a transfer to a new extension */
1991                                         if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
1992                                                 replace_macro_delimiter(macro_transfer_dest);
1993                                                 if (!ast_parseable_goto(chan, macro_transfer_dest))
1994                                                         ast_set_flag64(peerflags, OPT_GO_ON);
1995                                         }
1996                                 }
1997                         }
1998
1999                         ast_channel_unlock(peer);
2000                 }
2001
2002                 if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
2003                         struct ast_app *theapp;
2004                         const char *gosub_result;
2005                         char *gosub_args, *gosub_argstart;
2006                         int res9 = -1;
2007
2008                         res9 = ast_autoservice_start(chan);
2009                         if (res9) {
2010                                 ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
2011                                 res9 = -1;
2012                         }
2013
2014                         theapp = pbx_findapp("Gosub");
2015
2016                         if (theapp && !res9) {
2017                                 replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
2018
2019                                 /* Set where we came from */
2020                                 ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
2021                                 ast_copy_string(peer->exten, "s", sizeof(peer->exten));
2022                                 peer->priority = 0;
2023
2024                                 gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
2025                                 if (gosub_argstart) {
2026                                         *gosub_argstart = 0;
2027                                         if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
2028                                                 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2029                                                 gosub_args = NULL;
2030                                         }
2031                                         *gosub_argstart = ',';
2032                                 } else {
2033                                         if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
2034                                                 ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
2035                                                 gosub_args = NULL;
2036                                         }
2037                                 }
2038
2039                                 if (gosub_args) {
2040                                         res9 = pbx_exec(peer, theapp, gosub_args);
2041                                         if (!res9) {
2042                                                 struct ast_pbx_args args;
2043                                                 /* A struct initializer fails to compile for this case ... */
2044                                                 memset(&args, 0, sizeof(args));
2045                                                 args.no_hangup_chan = 1;
2046                                                 ast_pbx_run_args(peer, &args);
2047                                         }
2048                                         ast_free(gosub_args);
2049                                         ast_debug(1, "Gosub exited with status %d\n", res9);
2050                                 } else {
2051                                         ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
2052                                 }
2053
2054                         } else if (!res9) {
2055                                 ast_log(LOG_ERROR, "Could not find application Gosub\n");
2056                                 res9 = -1;
2057                         }
2058
2059                         if (ast_autoservice_stop(chan) < 0) {
2060                                 ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
2061                                 res9 = -1;
2062                         }
2063                         
2064                         ast_channel_lock(peer);
2065
2066                         if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
2067                                 char *gosub_transfer_dest;
2068
2069                                 if (!strcasecmp(gosub_result, "BUSY")) {
2070                                         ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2071                                         ast_set_flag64(peerflags, OPT_GO_ON);
2072                                         res9 = -1;
2073                                 } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
2074                                         ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
2075                                         ast_set_flag64(peerflags, OPT_GO_ON);
2076                                         res9 = -1;
2077                                 } else if (!strcasecmp(gosub_result, "CONTINUE")) {
2078                                         /* hangup peer and keep chan alive assuming the macro has changed
2079                                            the context / exten / priority or perhaps
2080                                            the next priority in the current exten is desired.
2081                                         */
2082                                         ast_set_flag64(peerflags, OPT_GO_ON);
2083                                         res9 = -1;
2084                                 } else if (!strcasecmp(gosub_result, "ABORT")) {
2085                                         /* Hangup both ends unless the caller has the g flag */
2086                                         res9 = -1;
2087                                 } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
2088                                         res9 = -1;
2089                                         /* perform a transfer to a new extension */
2090                                         if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
2091                                                 replace_macro_delimiter(gosub_transfer_dest);
2092                                                 if (!ast_parseable_goto(chan, gosub_transfer_dest))
2093                                                         ast_set_flag64(peerflags, OPT_GO_ON);
2094                                         }
2095                                 }
2096                         }
2097
2098                         ast_channel_unlock(peer);       
2099                 }
2100
2101                 if (!res) {
2102                         if (!ast_tvzero(calldurationlimit)) {
2103                                 struct timeval whentohangup = calldurationlimit;
2104                                 peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
2105                         }
2106                         if (!ast_strlen_zero(dtmfcalled)) {
2107                                 ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
2108                                 res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
2109                         }
2110                         if (!ast_strlen_zero(dtmfcalling)) {
2111                                 ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
2112                                 res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
2113                         }
2114                 }
2115
2116                 if (res) { /* some error */
2117                         res = -1;
2118                 } else {
2119                         if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
2120                                 ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
2121                         if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
2122                                 ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
2123                         if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
2124                                 ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
2125                         if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
2126                                 ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
2127                         if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
2128                                 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
2129                         if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
2130                                 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
2131                         if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
2132                                 ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
2133                         if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
2134                                 ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
2135                         if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
2136                                 ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
2137                         if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
2138                                 ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
2139                         if (ast_test_flag64(peerflags, OPT_GO_ON))
2140                                 ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
2141
2142                         config.end_bridge_callback = end_bridge_callback;
2143                         config.end_bridge_callback_data = chan;
2144                         config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
2145                         
2146                         if (moh) {
2147                                 moh = 0;
2148                                 ast_moh_stop(chan);
2149                         } else if (sentringing) {
2150                                 sentringing = 0;
2151                                 ast_indicate(chan, -1);
2152                         }
2153                         /* Be sure no generators are left on it */
2154                         ast_deactivate_generator(chan);
2155                         /* Make sure channels are compatible */
2156                         res = ast_channel_make_compatible(chan, peer);
2157                         if (res < 0) {
2158                                 ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
2159                                 ast_hangup(peer);
2160                                 res = -1;
2161                                 goto done;
2162                         }
2163                         if (opermode) {
2164                                 struct oprmode oprmode;
2165
2166                                 oprmode.peer = peer;
2167                                 oprmode.mode = opermode;
2168
2169                                 ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
2170                         }
2171                         res = ast_bridge_call(chan, peer, &config);
2172                 }
2173
2174                 strcpy(peer->context, chan->context);
2175
2176                 if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
2177                         int autoloopflag;
2178                         int found;
2179                         int res9;
2180                         
2181                         strcpy(peer->exten, "h");
2182                         peer->priority = 1;
2183                         autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
2184                         ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
2185
2186                         while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
2187                                 peer->priority++;
2188
2189                         if (found && res9) {
2190                                 /* Something bad happened, or a hangup has been requested. */
2191                                 ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2192                                 ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
2193                         }
2194                         ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP);  /* set it back the way it was */
2195                 }
2196                 if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {          
2197                         replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
2198                         ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
2199                         ast_pbx_start(peer);
2200                 } else {
2201                         if (!ast_check_hangup(chan))
2202                                 chan->hangupcause = peer->hangupcause;
2203                         ast_hangup(peer);
2204                 }
2205         }
2206 out:
2207         if (moh) {
2208                 moh = 0;
2209                 ast_moh_stop(chan);
2210         } else if (sentringing) {
2211                 sentringing = 0;
2212                 ast_indicate(chan, -1);
2213         }
2214         ast_channel_early_bridge(chan, NULL);
2215         hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
2216         pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
2217         senddialendevent(chan, pa.status);
2218         ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
2219         
2220         if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
2221                 if (!ast_tvzero(calldurationlimit))
2222                         memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
2223                 res = 0;
2224         }
2225
2226 done:
2227         if (config.warning_sound) {
2228                 ast_free((char *)config.warning_sound);
2229         }
2230         if (config.end_sound) {
2231                 ast_free((char *)config.end_sound);
2232         }
2233         if (config.start_sound) {
2234                 ast_free((char *)config.start_sound);
2235         }
2236         return res;
2237 }
2238
2239 static int dial_exec(struct ast_channel *chan, void *data)
2240 {
2241         struct ast_flags64 peerflags;
2242
2243         memset(&peerflags, 0, sizeof(peerflags));
2244
2245         return dial_exec_full(chan, data, &peerflags, NULL);
2246 }
2247
2248 static int retrydial_exec(struct ast_channel *chan, void *data)
2249 {
2250         char *parse;
2251         const char *context = NULL;
2252         int sleepms = 0, loops = 0, res = -1;
2253         struct ast_flags64 peerflags = { 0, };
2254         AST_DECLARE_APP_ARGS(args,
2255                 AST_APP_ARG(announce);
2256                 AST_APP_ARG(sleep);
2257                 AST_APP_ARG(retries);
2258                 AST_APP_ARG(dialdata);
2259         );
2260
2261         if (ast_strlen_zero(data)) {
2262                 ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
2263                 return -1;
2264         }
2265
2266         parse = ast_strdupa(data);
2267         AST_STANDARD_APP_ARGS(args, parse);
2268
2269         if ((sleepms = atoi(args.sleep)))
2270                 sleepms *= 1000;
2271
2272         loops = atoi(args.retries);
2273
2274         if (!args.dialdata) {
2275                 ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
2276                 goto done;
2277         }
2278
2279         if (sleepms < 1000)
2280                 sleepms = 10000;
2281
2282         if (!loops)
2283                 loops = -1; /* run forever */
2284
2285         ast_channel_lock(chan);
2286         context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
2287         context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
2288         ast_channel_unlock(chan);
2289
2290         res = 0;
2291         while (loops) {
2292                 int continue_exec;
2293
2294                 chan->data = "Retrying";
2295                 if (ast_test_flag(chan, AST_FLAG_MOH))
2296                         ast_moh_stop(chan);
2297
2298                 res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
2299                 if (continue_exec)
2300                         break;
2301
2302                 if (res == 0) {
2303                         if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
2304                                 if (!ast_strlen_zero(args.announce)) {
2305                                         if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2306                                                 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2307                                                         ast_waitstream(chan, AST_DIGIT_ANY);
2308                                         } else
2309                                                 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2310                                 }
2311                                 if (!res && sleepms) {
2312                                         if (!ast_test_flag(chan, AST_FLAG_MOH))
2313                                                 ast_moh_start(chan, NULL, NULL);
2314                                         res = ast_waitfordigit(chan, sleepms);
2315                                 }
2316                         } else {
2317                                 if (!ast_strlen_zero(args.announce)) {
2318                                         if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
2319                                                 if (!(res = ast_streamfile(chan, args.announce, chan->language)))
2320                                                         res = ast_waitstream(chan, "");
2321                                         } else
2322                                                 ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
2323                                 }
2324                                 if (sleepms) {
2325                                         if (!ast_test_flag(chan, AST_FLAG_MOH))
2326                                                 ast_moh_start(chan, NULL, NULL);
2327                                         if (!res)
2328                                                 res = ast_waitfordigit(chan, sleepms);
2329                                 }
2330                         }
2331                 }
2332
2333                 if (res < 0 || res == AST_PBX_INCOMPLETE) {
2334                         break;
2335                 } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
2336                         if (onedigit_goto(chan, context, (char) res, 1)) {
2337                                 res = 0;
2338                                 break;
2339                         }
2340                 }
2341                 loops--;
2342         }
2343         if (loops == 0)
2344                 res = 0;
2345         else if (res == 1)
2346                 res = 0;
2347
2348         if (ast_test_flag(chan, AST_FLAG_MOH))
2349                 ast_moh_stop(chan);
2350  done:
2351         return res;
2352 }
2353
2354 static int unload_module(void)
2355 {
2356         int res;
2357         struct ast_context *con;
2358
2359         res = ast_unregister_application(app);
2360         res |= ast_unregister_application(rapp);
2361
2362         if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
2363                 ast_context_remove_extension2(con, "s", 1, NULL, 0);
2364                 ast_context_destroy(con, "app_dial"); /* leave nothing behind */
2365         }
2366
2367         return res;
2368 }
2369
2370 static int load_module(void)
2371 {
2372         int res;
2373         struct ast_context *con;
2374
2375         con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
2376         if (!con)
2377                 ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
2378         else
2379                 ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
2380
2381         res = ast_register_application_xml(app, dial_exec);
2382         res |= ast_register_application_xml(rapp, retrydial_exec);
2383
2384         return res;
2385 }
2386
2387 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");