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Asterisk developer's documentation


app_dial.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2008, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*! \file
00020  *
00021  * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
00022  *
00023  * \author Mark Spencer <markster@digium.com>
00024  *
00025  * \ingroup applications
00026  */
00027 
00028 /*** MODULEINFO
00029    <depend>chan_local</depend>
00030  ***/
00031 
00032 
00033 #include "asterisk.h"
00034 
00035 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 305253 $")
00036 
00037 #include <sys/time.h>
00038 #include <sys/signal.h>
00039 #include <sys/stat.h>
00040 #include <netinet/in.h>
00041 
00042 #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
00043 #include "asterisk/lock.h"
00044 #include "asterisk/file.h"
00045 #include "asterisk/channel.h"
00046 #include "asterisk/pbx.h"
00047 #include "asterisk/module.h"
00048 #include "asterisk/translate.h"
00049 #include "asterisk/say.h"
00050 #include "asterisk/config.h"
00051 #include "asterisk/features.h"
00052 #include "asterisk/musiconhold.h"
00053 #include "asterisk/callerid.h"
00054 #include "asterisk/utils.h"
00055 #include "asterisk/app.h"
00056 #include "asterisk/causes.h"
00057 #include "asterisk/rtp.h"
00058 #include "asterisk/cdr.h"
00059 #include "asterisk/manager.h"
00060 #include "asterisk/privacy.h"
00061 #include "asterisk/stringfields.h"
00062 #include "asterisk/global_datastores.h"
00063 #include "asterisk/dsp.h"
00064 
00065 /*** DOCUMENTATION
00066    <application name="Dial" language="en_US">
00067       <synopsis>
00068          Attempt to connect to another device or endpoint and bridge the call.
00069       </synopsis>
00070       <syntax>
00071          <parameter name="Technology/Resource" required="true" argsep="&amp;">
00072             <argument name="Technology/Resource" required="true">
00073                <para>Specification of the device(s) to dial.  These must be in the format of
00074                <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
00075                represents a particular channel driver, and <replaceable>Resource</replaceable>
00076                represents a resource available to that particular channel driver.</para>
00077             </argument>
00078             <argument name="Technology2/Resource2" required="false" multiple="true">
00079                <para>Optional extra devices to dial in parallel</para>
00080                <para>If you need more then one enter them as
00081                Technology2/Resource2&amp;Technology3/Resourse3&amp;.....</para>
00082             </argument>
00083          </parameter>
00084          <parameter name="timeout" required="false">
00085             <para>Specifies the number of seconds we attempt to dial the specified devices</para>
00086             <para>If not specified, this defaults to 136 years.</para>
00087          </parameter>
00088          <parameter name="options" required="false">
00089             <optionlist>
00090             <option name="A">
00091                <argument name="x" required="true">
00092                   <para>The file to play to the called party</para>
00093                </argument>
00094                <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
00095             </option>
00096             <option name="C">
00097                <para>Reset the call detail record (CDR) for this call.</para>
00098             </option>
00099             <option name="c">
00100                <para>If the Dial() application cancels this call, always set the flag to tell the channel
00101                driver that the call is answered elsewhere.</para>
00102             </option>
00103             <option name="d">
00104                <para>Allow the calling user to dial a 1 digit extension while waiting for
00105                a call to be answered. Exit to that extension if it exists in the
00106                current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
00107                if it exists.</para>
00108             </option>
00109             <option name="D" argsep=":">
00110                <argument name="called" />
00111                <argument name="calling" />
00112                <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
00113                party has answered, but before the call gets bridged. The 
00114                <replaceable>called</replaceable> DTMF string is sent to the called party, and the 
00115                <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments 
00116                can be used alone.</para>
00117             </option>
00118             <option name="e">
00119                <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
00120             </option>
00121             <option name="f">
00122                <para>Force the callerid of the <emphasis>calling</emphasis> channel to be set as the
00123                extension associated with the channel using a dialplan <literal>hint</literal>.
00124                For example, some PSTNs do not allow CallerID to be set to anything
00125                other than the number assigned to the caller.</para>
00126             </option>
00127             <option name="F" argsep="^">
00128                <argument name="context" required="false" />
00129                <argument name="exten" required="false" />
00130                <argument name="priority" required="true" />
00131                <para>When the caller hangs up, transfer the called party
00132                to the specified destination and continue execution at that location.</para>
00133             </option>
00134             <option name="g">
00135                <para>Proceed with dialplan execution at the next priority in the current extension if the
00136                destination channel hangs up.</para>
00137             </option>
00138             <option name="G" argsep="^">
00139                <argument name="context" required="false" />
00140                <argument name="exten" required="false" />
00141                <argument name="priority" required="true" />
00142                <para>If the call is answered, transfer the calling party to
00143                the specified <replaceable>priority</replaceable> and the called party to the specified 
00144                <replaceable>priority</replaceable> plus one.</para>
00145                <note>
00146                   <para>You cannot use any additional action post answer options in conjunction with this option.</para>
00147                </note>
00148             </option>
00149             <option name="h">
00150                <para>Allow the called party to hang up by sending the <literal>*</literal> DTMF digit.</para>
00151             </option>
00152             <option name="H">
00153                <para>Allow the calling party to hang up by hitting the <literal>*</literal> DTMF digit.</para>
00154             </option>
00155             <option name="i">
00156                <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
00157             </option>
00158             <option name="k">
00159                <para>Allow the called party to enable parking of the call by sending
00160                the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
00161             </option>
00162             <option name="K">
00163                <para>Allow the calling party to enable parking of the call by sending
00164                the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
00165             </option>
00166             <option name="L" argsep=":">
00167                <argument name="x" required="true">
00168                   <para>Maximum call time, in milliseconds</para>
00169                </argument>
00170                <argument name="y">
00171                   <para>Warning time, in milliseconds</para>
00172                </argument>
00173                <argument name="z">
00174                   <para>Repeat time, in milliseconds</para>
00175                </argument>
00176                <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
00177                left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
00178                <para>This option is affected by the following variables:</para>
00179                <variablelist>
00180                   <variable name="LIMIT_PLAYAUDIO_CALLER">
00181                      <value name="yes" default="true" />
00182                      <value name="no" />
00183                      <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
00184                   </variable>
00185                   <variable name="LIMIT_PLAYAUDIO_CALLEE">
00186                      <value name="yes" />
00187                      <value name="no" default="true"/>
00188                      <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
00189                   </variable>
00190                   <variable name="LIMIT_TIMEOUT_FILE">
00191                      <value name="filename"/>
00192                      <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
00193                      If not set, the time remaining will be announced.</para>
00194                   </variable>
00195                   <variable name="LIMIT_CONNECT_FILE">
00196                      <value name="filename"/>
00197                      <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
00198                      If not set, the time remaining will be announced.</para>
00199                   </variable>
00200                   <variable name="LIMIT_WARNING_FILE">
00201                      <value name="filename"/>
00202                      <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
00203                      a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
00204                   </variable>
00205                </variablelist>
00206             </option>
00207             <option name="m">
00208                <argument name="class" required="false"/>
00209                <para>Provide hold music to the calling party until a requested
00210                channel answers. A specific music on hold <replaceable>class</replaceable>
00211                (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
00212             </option>
00213             <option name="M" argsep="^">
00214                <argument name="macro" required="true">
00215                   <para>Name of the macro that should be executed.</para>
00216                </argument>
00217                <argument name="arg" multiple="true">
00218                   <para>Macro arguments</para>
00219                </argument>
00220                <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel 
00221                before connecting to the calling channel. Arguments can be specified to the Macro
00222                using <literal>^</literal> as a delimiter. The macro can set the variable
00223                <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
00224                finished executing:</para>
00225                <variablelist>
00226                   <variable name="MACRO_RESULT">
00227                      <para>If set, this action will be taken after the macro finished executing.</para>
00228                      <value name="ABORT">
00229                         Hangup both legs of the call
00230                      </value>
00231                      <value name="CONGESTION">
00232                         Behave as if line congestion was encountered
00233                      </value>
00234                      <value name="BUSY">
00235                         Behave as if a busy signal was encountered
00236                      </value>
00237                      <value name="CONTINUE">
00238                         Hangup the called party and allow the calling party to continue dialplan execution at the next priority
00239                      </value>
00240                      <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
00241                      <value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
00242                         Transfer the call to the specified destination.
00243                      </value>
00244                   </variable>
00245                </variablelist>
00246                <note>
00247                   <para>You cannot use any additional action post answer options in conjunction
00248                   with this option. Also, pbx services are not run on the peer (called) channel,
00249                   so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
00250                </note>
00251                <warning><para>Be aware of the limitations that macros have, specifically with regards to use of
00252                the <literal>WaitExten</literal> application. For more information, see the documentation for
00253                Macro()</para></warning>
00254             </option>
00255             <option name="n">
00256                     <argument name="delete">
00257                        <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
00258                   the recorded introduction will not be deleted if the caller hangs up while the remote party has not
00259                   yet answered.</para>
00260                   <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
00261                   always be deleted.</para>
00262                </argument>
00263                <para>This option is a modifier for the call screening/privacy mode. (See the 
00264                <literal>p</literal> and <literal>P</literal> options.) It specifies
00265                that no introductions are to be saved in the <directory>priv-callerintros</directory>
00266                directory.</para>
00267             </option>
00268             <option name="N">
00269                <para>This option is a modifier for the call screening/privacy mode. It specifies
00270                that if Caller*ID is present, do not screen the call.</para>
00271             </option>
00272             <option name="o">
00273                <para>Specify that the Caller*ID that was present on the <emphasis>calling</emphasis> channel
00274                be set as the Caller*ID on the <emphasis>called</emphasis> channel. This was the
00275                behavior of Asterisk 1.0 and earlier.</para>
00276             </option>
00277             <option name="O">
00278                <argument name="mode">
00279                   <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
00280                   the originator hanging up will cause the phone to ring back immediately.</para>
00281                   <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator 
00282                   flashes the trunk, it will ring their phone back.</para>
00283                </argument>
00284                <para>Enables <emphasis>operator services</emphasis> mode.  This option only
00285                works when bridging a DAHDI channel to another DAHDI channel
00286                only. if specified on non-DAHDI interfaces, it will be ignored.
00287                When the destination answers (presumably an operator services
00288                station), the originator no longer has control of their line.
00289                They may hang up, but the switch will not release their line
00290                until the destination party (the operator) hangs up.</para>
00291             </option>
00292             <option name="p">
00293                <para>This option enables screening mode. This is basically Privacy mode
00294                without memory.</para>
00295             </option>
00296             <option name="P">
00297                <argument name="x" />
00298                <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
00299                it is provided. The current extension is used if a database family/key is not specified.</para>
00300             </option>
00301             <option name="r">
00302                <para>Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
00303                party until the called channel has answered.</para>
00304             </option>
00305             <option name="S">
00306                <argument name="x" required="true" />
00307                <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
00308                answered the call.</para>
00309             </option>
00310             <option name="t">
00311                <para>Allow the called party to transfer the calling party by sending the
00312                DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
00313                transfers initiated by other methods.</para>
00314             </option>
00315             <option name="T">
00316                <para>Allow the calling party to transfer the called party by sending the
00317                DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
00318                transfers initiated by other methods.</para>
00319             </option>
00320             <option name="U" argsep="^">
00321                <argument name="x" required="true">
00322                   <para>Name of the subroutine to execute via Gosub</para>
00323                </argument>
00324                <argument name="arg" multiple="true" required="false">
00325                   <para>Arguments for the Gosub routine</para>
00326                </argument>
00327                <para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
00328                to the calling channel. Arguments can be specified to the Gosub
00329                using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
00330                <variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
00331                <variablelist>
00332                   <variable name="GOSUB_RESULT">
00333                      <value name="ABORT">
00334                         Hangup both legs of the call.
00335                      </value>
00336                      <value name="CONGESTION">
00337                         Behave as if line congestion was encountered.
00338                      </value>
00339                      <value name="BUSY">
00340                         Behave as if a busy signal was encountered.
00341                      </value>
00342                      <value name="CONTINUE">
00343                         Hangup the called party and allow the calling party
00344                         to continue dialplan execution at the next priority.
00345                      </value>
00346                      <!-- TODO: Fix this syntax up, once we've figured out how to specify the GOTO syntax -->
00347                      <value name="GOTO:&lt;context&gt;^&lt;exten&gt;^&lt;priority&gt;">
00348                         Transfer the call to the specified priority. Optionally, an extension, or
00349                         extension and priority can be specified.
00350                      </value>
00351                   </variable>
00352                </variablelist>
00353                <note>
00354                   <para>You cannot use any additional action post answer options in conjunction
00355                   with this option. Also, pbx services are not run on the peer (called) channel,
00356                   so you will not be able to set timeouts via the TIMEOUT() function in this routine.</para>
00357                </note>
00358             </option>
00359             <option name="w">
00360                <para>Allow the called party to enable recording of the call by sending
00361                the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
00362             </option>
00363             <option name="W">
00364                <para>Allow the calling party to enable recording of the call by sending
00365                the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
00366             </option>
00367             <option name="x">
00368                <para>Allow the called party to enable recording of the call by sending
00369                the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
00370             </option>
00371             <option name="X">
00372                <para>Allow the calling party to enable recording of the call by sending
00373                the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
00374             </option>
00375             </optionlist>
00376          </parameter>
00377          <parameter name="URL">
00378             <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
00379          </parameter>
00380       </syntax>
00381       <description>
00382          <para>This application will place calls to one or more specified channels. As soon
00383          as one of the requested channels answers, the originating channel will be
00384          answered, if it has not already been answered. These two channels will then
00385          be active in a bridged call. All other channels that were requested will then
00386          be hung up.</para>
00387 
00388          <para>Unless there is a timeout specified, the Dial application will wait
00389          indefinitely until one of the called channels answers, the user hangs up, or
00390          if all of the called channels are busy or unavailable. Dialplan executing will
00391          continue if no requested channels can be called, or if the timeout expires.
00392          This application will report normal termination if the originating channel
00393          hangs up, or if the call is bridged and either of the parties in the bridge
00394          ends the call.</para>
00395 
00396          <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
00397          application will be put into that group (as in Set(GROUP()=...).
00398          If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
00399          application will be put into that group (as in Set(GROUP()=...). Unlike OUTBOUND_GROUP,
00400          however, the variable will be unset after use.</para>
00401 
00402          <para>This application sets the following channel variables:</para>
00403          <variablelist>
00404             <variable name="DIALEDTIME">
00405                <para>This is the time from dialing a channel until when it is disconnected.</para>
00406             </variable>
00407             <variable name="ANSWEREDTIME">
00408                <para>This is the amount of time for actual call.</para>
00409             </variable>
00410             <variable name="DIALSTATUS">
00411                <para>This is the status of the call</para>
00412                <value name="CHANUNAVAIL" />
00413                <value name="CONGESTION" />
00414                <value name="NOANSWER" />
00415                <value name="BUSY" />
00416                <value name="ANSWER" />
00417                <value name="CANCEL" />
00418                <value name="DONTCALL">
00419                   For the Privacy and Screening Modes.
00420                   Will be set if the called party chooses to send the calling party to the 'Go Away' script.
00421                </value>
00422                <value name="TORTURE">
00423                   For the Privacy and Screening Modes.
00424                   Will be set if the called party chooses to send the calling party to the 'torture' script.
00425                </value>
00426                <value name="INVALIDARGS" />
00427             </variable>
00428          </variablelist>
00429       </description>
00430    </application>
00431    <application name="RetryDial" language="en_US">
00432       <synopsis>
00433          Place a call, retrying on failure allowing an optional exit extension.
00434       </synopsis>
00435       <syntax>
00436          <parameter name="announce" required="true">
00437             <para>Filename of sound that will be played when no channel can be reached</para>
00438          </parameter>
00439          <parameter name="sleep" required="true">
00440             <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
00441          </parameter>
00442          <parameter name="retries" required="true">
00443             <para>Number of retries</para>
00444             <para>When this is reached flow will continue at the next priority in the dialplan</para>
00445          </parameter>
00446          <parameter name="dialargs" required="true">
00447             <para>Same format as arguments provided to the Dial application</para>
00448          </parameter>
00449       </syntax>
00450       <description>
00451          <para>This application will attempt to place a call using the normal Dial application.
00452          If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
00453          Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
00454          After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
00455          If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
00456          While waiting to retry a call, a 1 digit extension may be dialed. If that
00457          extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
00458          one, The call will jump to that extension immediately.
00459          The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
00460          to the Dial application.</para>
00461       </description>
00462    </application>
00463  ***/
00464 
00465 static char *app = "Dial";
00466 static char *rapp = "RetryDial";
00467 
00468 enum {
00469    OPT_ANNOUNCE =          (1 << 0),
00470    OPT_RESETCDR =          (1 << 1),
00471    OPT_DTMF_EXIT =         (1 << 2),
00472    OPT_SENDDTMF =          (1 << 3),
00473    OPT_FORCECLID =         (1 << 4),
00474    OPT_GO_ON =             (1 << 5),
00475    OPT_CALLEE_HANGUP =     (1 << 6),
00476    OPT_CALLER_HANGUP =     (1 << 7),
00477    OPT_DURATION_LIMIT =    (1 << 9),
00478    OPT_MUSICBACK =         (1 << 10),
00479    OPT_CALLEE_MACRO =      (1 << 11),
00480    OPT_SCREEN_NOINTRO =    (1 << 12),
00481    OPT_SCREEN_NOCLID =     (1 << 13),
00482    OPT_ORIGINAL_CLID =     (1 << 14),
00483    OPT_SCREENING =         (1 << 15),
00484    OPT_PRIVACY =           (1 << 16),
00485    OPT_RINGBACK =          (1 << 17),
00486    OPT_DURATION_STOP =     (1 << 18),
00487    OPT_CALLEE_TRANSFER =   (1 << 19),
00488    OPT_CALLER_TRANSFER =   (1 << 20),
00489    OPT_CALLEE_MONITOR =    (1 << 21),
00490    OPT_CALLER_MONITOR =    (1 << 22),
00491    OPT_GOTO =              (1 << 23),
00492    OPT_OPERMODE =          (1 << 24),
00493    OPT_CALLEE_PARK =       (1 << 25),
00494    OPT_CALLER_PARK =       (1 << 26),
00495    OPT_IGNORE_FORWARDING = (1 << 27),
00496    OPT_CALLEE_GOSUB =      (1 << 28),
00497    OPT_CALLEE_MIXMONITOR = (1 << 29),
00498    OPT_CALLER_MIXMONITOR = (1 << 30),
00499 };
00500 
00501 #define DIAL_STILLGOING      (1 << 31)
00502 #define DIAL_NOFORWARDHTML   ((uint64_t)1 << 32) /* flags are now 64 bits, so keep it up! */
00503 #define OPT_CANCEL_ELSEWHERE ((uint64_t)1 << 33)
00504 #define OPT_PEER_H           ((uint64_t)1 << 34)
00505 #define OPT_CALLEE_GO_ON     ((uint64_t)1 << 35)
00506 
00507 enum {
00508    OPT_ARG_ANNOUNCE = 0,
00509    OPT_ARG_SENDDTMF,
00510    OPT_ARG_GOTO,
00511    OPT_ARG_DURATION_LIMIT,
00512    OPT_ARG_MUSICBACK,
00513    OPT_ARG_CALLEE_MACRO,
00514    OPT_ARG_CALLEE_GOSUB,
00515    OPT_ARG_CALLEE_GO_ON,
00516    OPT_ARG_PRIVACY,
00517    OPT_ARG_DURATION_STOP,
00518    OPT_ARG_OPERMODE,
00519    OPT_ARG_SCREEN_NOINTRO,
00520    /* note: this entry _MUST_ be the last one in the enum */
00521    OPT_ARG_ARRAY_SIZE,
00522 };
00523 
00524 AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
00525    AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
00526    AST_APP_OPTION('C', OPT_RESETCDR),
00527    AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
00528    AST_APP_OPTION('d', OPT_DTMF_EXIT),
00529    AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
00530    AST_APP_OPTION('e', OPT_PEER_H),
00531    AST_APP_OPTION('f', OPT_FORCECLID),
00532    AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
00533    AST_APP_OPTION('g', OPT_GO_ON),
00534    AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
00535    AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
00536    AST_APP_OPTION('H', OPT_CALLER_HANGUP),
00537    AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
00538    AST_APP_OPTION('k', OPT_CALLEE_PARK),
00539    AST_APP_OPTION('K', OPT_CALLER_PARK),
00540    AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
00541    AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
00542    AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
00543    AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
00544    AST_APP_OPTION('N', OPT_SCREEN_NOCLID),
00545    AST_APP_OPTION('o', OPT_ORIGINAL_CLID),
00546    AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
00547    AST_APP_OPTION('p', OPT_SCREENING),
00548    AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
00549    AST_APP_OPTION('r', OPT_RINGBACK),
00550    AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
00551    AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
00552    AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
00553    AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
00554    AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
00555    AST_APP_OPTION('W', OPT_CALLER_MONITOR),
00556    AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
00557    AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
00558 END_OPTIONS );
00559 
00560 #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
00561    OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
00562    OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK |  \
00563    OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
00564    !chan->audiohooks && !peer->audiohooks)
00565 
00566 /*
00567  * The list of active channels
00568  */
00569 struct chanlist {
00570    struct chanlist *next;
00571    struct ast_channel *chan;
00572    uint64_t flags;
00573 };
00574 
00575 
00576 static void hanguptree(struct chanlist *outgoing, struct ast_channel *exception, int answered_elsewhere)
00577 {
00578    /* Hang up a tree of stuff */
00579    struct chanlist *oo;
00580    while (outgoing) {
00581       /* Hangup any existing lines we have open */
00582       if (outgoing->chan && (outgoing->chan != exception)) {
00583          if (answered_elsewhere) {
00584             /* The flag is used for local channel inheritance and stuff */
00585             ast_set_flag(outgoing->chan, AST_FLAG_ANSWERED_ELSEWHERE);
00586             /* This is for the channel drivers */
00587             outgoing->chan->hangupcause = AST_CAUSE_ANSWERED_ELSEWHERE;
00588          }
00589          ast_hangup(outgoing->chan);
00590       }
00591       oo = outgoing;
00592       outgoing = outgoing->next;
00593       ast_free(oo);
00594    }
00595 }
00596 
00597 #define AST_MAX_WATCHERS 256
00598 
00599 /*
00600  * argument to handle_cause() and other functions.
00601  */
00602 struct cause_args {
00603    struct ast_channel *chan;
00604    int busy;
00605    int congestion;
00606    int nochan;
00607 };
00608 
00609 static void handle_cause(int cause, struct cause_args *num)
00610 {
00611    struct ast_cdr *cdr = num->chan->cdr;
00612 
00613    switch(cause) {
00614    case AST_CAUSE_BUSY:
00615       if (cdr)
00616          ast_cdr_busy(cdr);
00617       num->busy++;
00618       break;
00619 
00620    case AST_CAUSE_CONGESTION:
00621       if (cdr)
00622          ast_cdr_failed(cdr);
00623       num->congestion++;
00624       break;
00625 
00626    case AST_CAUSE_NO_ROUTE_DESTINATION:
00627    case AST_CAUSE_UNREGISTERED:
00628       if (cdr)
00629          ast_cdr_failed(cdr);
00630       num->nochan++;
00631       break;
00632 
00633    case AST_CAUSE_NO_ANSWER:
00634       if (cdr) {
00635          ast_cdr_noanswer(cdr);
00636       }
00637       break;
00638    case AST_CAUSE_NORMAL_CLEARING:
00639       break;
00640 
00641    default:
00642       num->nochan++;
00643       break;
00644    }
00645 }
00646 
00647 /* free the buffer if allocated, and set the pointer to the second arg */
00648 #define S_REPLACE(s, new_val)    \
00649    do {           \
00650       if (s)         \
00651          ast_free(s);   \
00652       s = (new_val);    \
00653    } while (0)
00654 
00655 static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
00656 {
00657    char rexten[2] = { exten, '\0' };
00658 
00659    if (context) {
00660       if (!ast_goto_if_exists(chan, context, rexten, pri))
00661          return 1;
00662    } else {
00663       if (!ast_goto_if_exists(chan, chan->context, rexten, pri))
00664          return 1;
00665       else if (!ast_strlen_zero(chan->macrocontext)) {
00666          if (!ast_goto_if_exists(chan, chan->macrocontext, rexten, pri))
00667             return 1;
00668       }
00669    }
00670    return 0;
00671 }
00672 
00673 
00674 static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
00675 {
00676    const char *context = S_OR(chan->macrocontext, chan->context);
00677    const char *exten = S_OR(chan->macroexten, chan->exten);
00678 
00679    return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
00680 }
00681 
00682 static void senddialevent(struct ast_channel *src, struct ast_channel *dst, const char *dialstring)
00683 {
00684    manager_event(EVENT_FLAG_CALL, "Dial",
00685       "SubEvent: Begin\r\n"
00686       "Channel: %s\r\n"
00687       "Destination: %s\r\n"
00688       "CallerIDNum: %s\r\n"
00689       "CallerIDName: %s\r\n"
00690       "UniqueID: %s\r\n"
00691       "DestUniqueID: %s\r\n"
00692       "Dialstring: %s\r\n",
00693       src->name, dst->name, S_OR(src->cid.cid_num, "<unknown>"),
00694       S_OR(src->cid.cid_name, "<unknown>"), src->uniqueid,
00695       dst->uniqueid, dialstring ? dialstring : "");
00696 }
00697 
00698 static void senddialendevent(const struct ast_channel *src, const char *dialstatus)
00699 {
00700    manager_event(EVENT_FLAG_CALL, "Dial",
00701       "SubEvent: End\r\n"
00702       "Channel: %s\r\n"
00703       "UniqueID: %s\r\n"
00704       "DialStatus: %s\r\n",
00705       src->name, src->uniqueid, dialstatus);
00706 }
00707 
00708 /*!
00709  * helper function for wait_for_answer()
00710  *
00711  * XXX this code is highly suspicious, as it essentially overwrites
00712  * the outgoing channel without properly deleting it.
00713  */
00714 static void do_forward(struct chanlist *o,
00715    struct cause_args *num, struct ast_flags64 *peerflags, int single)
00716 {
00717    char tmpchan[256];
00718    struct ast_channel *original = o->chan;
00719    struct ast_channel *c = o->chan; /* the winner */
00720    struct ast_channel *in = num->chan; /* the input channel */
00721    char *stuff;
00722    char *tech;
00723    int cause;
00724 
00725    ast_copy_string(tmpchan, c->call_forward, sizeof(tmpchan));
00726    if ((stuff = strchr(tmpchan, '/'))) {
00727       *stuff++ = '\0';
00728       tech = tmpchan;
00729    } else {
00730       const char *forward_context;
00731       ast_channel_lock(c);
00732       forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
00733       if (ast_strlen_zero(forward_context)) {
00734          forward_context = NULL;
00735       }
00736       snprintf(tmpchan, sizeof(tmpchan), "%s@%s", c->call_forward, forward_context ? forward_context : c->context);
00737       ast_channel_unlock(c);
00738       stuff = tmpchan;
00739       tech = "Local";
00740    }
00741    /* Before processing channel, go ahead and check for forwarding */
00742    ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", in->name, tech, stuff, c->name);
00743    /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
00744    if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
00745       ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", in->name, tech, stuff);
00746       c = o->chan = NULL;
00747       cause = AST_CAUSE_BUSY;
00748    } else {
00749       /* Setup parameters */
00750       c = o->chan = ast_request(tech, in->nativeformats, stuff, &cause);
00751       if (c) {
00752          if (single)
00753             ast_channel_make_compatible(o->chan, in);
00754          ast_channel_inherit_variables(in, o->chan);
00755          ast_channel_datastore_inherit(in, o->chan);
00756       } else
00757          ast_log(LOG_NOTICE,
00758             "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
00759             tech, stuff, cause);
00760    }
00761    if (!c) {
00762       ast_clear_flag64(o, DIAL_STILLGOING);
00763       handle_cause(cause, num);
00764       ast_hangup(original);
00765    } else {
00766       char *new_cid_num, *new_cid_name;
00767       struct ast_channel *src;
00768 
00769       if (CAN_EARLY_BRIDGE(peerflags, c, in)) {
00770          ast_rtp_make_compatible(c, in, single);
00771       }
00772       if (ast_test_flag64(o, OPT_FORCECLID)) {
00773          new_cid_num = ast_strdup(S_OR(in->macroexten, in->exten));
00774          new_cid_name = NULL; /* XXX no name ? */
00775          src = c; /* XXX possible bug in previous code, which used 'winner' ? it may have changed */
00776       } else {
00777          new_cid_num = ast_strdup(in->cid.cid_num);
00778          new_cid_name = ast_strdup(in->cid.cid_name);
00779          src = in;
00780       }
00781       ast_string_field_set(c, accountcode, src->accountcode);
00782       c->cdrflags = src->cdrflags;
00783       S_REPLACE(c->cid.cid_num, new_cid_num);
00784       S_REPLACE(c->cid.cid_name, new_cid_name);
00785 
00786       if (in->cid.cid_ani) { /* XXX or maybe unconditional ? */
00787          S_REPLACE(c->cid.cid_ani, ast_strdup(in->cid.cid_ani));
00788       }
00789       S_REPLACE(c->cid.cid_rdnis, ast_strdup(S_OR(in->macroexten, in->exten)));
00790       if (ast_call(c, stuff, 0)) {
00791          ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
00792             tech, stuff);
00793          ast_clear_flag64(o, DIAL_STILLGOING);
00794          ast_hangup(original);
00795          ast_hangup(c);
00796          c = o->chan = NULL;
00797          num->nochan++;
00798       } else {
00799          senddialevent(in, c, stuff);
00800          /* After calling, set callerid to extension */
00801          if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
00802             char cidname[AST_MAX_EXTENSION] = "";
00803             ast_set_callerid(c, S_OR(in->macroexten, in->exten), get_cid_name(cidname, sizeof(cidname), in), NULL);
00804          }
00805          /* Hangup the original channel now, in case we needed it */
00806          ast_hangup(original);
00807       }
00808       if (single) {
00809          ast_indicate(in, -1);
00810       }
00811    }
00812 }
00813 
00814 /* argument used for some functions. */
00815 struct privacy_args {
00816    int sentringing;
00817    int privdb_val;
00818    char privcid[256];
00819    char privintro[1024];
00820    char status[256];
00821 };
00822 
00823 static struct ast_channel *wait_for_answer(struct ast_channel *in,
00824    struct chanlist *outgoing, int *to, struct ast_flags64 *peerflags,
00825    struct privacy_args *pa,
00826    const struct cause_args *num_in, int *result)
00827 {
00828    struct cause_args num = *num_in;
00829    int prestart = num.busy + num.congestion + num.nochan;
00830    int orig = *to;
00831    struct ast_channel *peer = NULL;
00832    /* single is set if only one destination is enabled */
00833    int single = outgoing && !outgoing->next && !ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
00834 #ifdef HAVE_EPOLL
00835    struct chanlist *epollo;
00836 #endif
00837 
00838    if (single) {
00839       /* Turn off hold music, etc */
00840       ast_deactivate_generator(in);
00841       /* If we are calling a single channel, make them compatible for in-band tone purpose */
00842       if (ast_channel_make_compatible(outgoing->chan, in) < 0) {
00843          /* If these channels can not be made compatible, 
00844           * there is no point in continuing.  The bridge
00845           * will just fail if it gets that far.
00846           */
00847          *to = -1;
00848          strcpy(pa->status, "CONGESTION");
00849          ast_cdr_failed(in->cdr);
00850          return NULL;
00851       }
00852    }
00853 
00854 #ifdef HAVE_EPOLL
00855    for (epollo = outgoing; epollo; epollo = epollo->next)
00856       ast_poll_channel_add(in, epollo->chan);
00857 #endif
00858 
00859    while (*to && !peer) {
00860       struct chanlist *o;
00861       int pos = 0; /* how many channels do we handle */
00862       int numlines = prestart;
00863       struct ast_channel *winner;
00864       struct ast_channel *watchers[AST_MAX_WATCHERS];
00865 
00866       watchers[pos++] = in;
00867       for (o = outgoing; o; o = o->next) {
00868          /* Keep track of important channels */
00869          if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
00870             watchers[pos++] = o->chan;
00871          numlines++;
00872       }
00873       if (pos == 1) { /* only the input channel is available */
00874          if (numlines == (num.busy + num.congestion + num.nochan)) {
00875             ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
00876             if (num.busy)
00877                strcpy(pa->status, "BUSY");
00878             else if (num.congestion)
00879                strcpy(pa->status, "CONGESTION");
00880             else if (num.nochan)
00881                strcpy(pa->status, "CHANUNAVAIL");
00882          } else {
00883             ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
00884          }
00885          *to = 0;
00886          return NULL;
00887       }
00888       winner = ast_waitfor_n(watchers, pos, to);
00889       for (o = outgoing; o; o = o->next) {
00890          struct ast_frame *f;
00891          struct ast_channel *c = o->chan;
00892 
00893          if (c == NULL)
00894             continue;
00895          if (ast_test_flag64(o, DIAL_STILLGOING) && c->_state == AST_STATE_UP) {
00896             if (!peer) {
00897                ast_verb(3, "%s answered %s\n", c->name, in->name);
00898                peer = c;
00899                ast_copy_flags64(peerflags, o,
00900                   OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
00901                   OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
00902                   OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
00903                   OPT_CALLEE_PARK | OPT_CALLER_PARK |
00904                   OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
00905                   DIAL_NOFORWARDHTML);
00906                ast_string_field_set(c, dialcontext, "");
00907                ast_copy_string(c->exten, "", sizeof(c->exten));
00908             }
00909             continue;
00910          }
00911          if (c != winner)
00912             continue;
00913          /* here, o->chan == c == winner */
00914          if (!ast_strlen_zero(c->call_forward)) {
00915             do_forward(o, &num, peerflags, single);
00916             continue;
00917          }
00918          f = ast_read(winner);
00919          if (!f) {
00920             in->hangupcause = c->hangupcause;
00921 #ifdef HAVE_EPOLL
00922             ast_poll_channel_del(in, c);
00923 #endif
00924             ast_hangup(c);
00925             c = o->chan = NULL;
00926             ast_clear_flag64(o, DIAL_STILLGOING);
00927             handle_cause(in->hangupcause, &num);
00928             continue;
00929          }
00930          if (f->frametype == AST_FRAME_CONTROL) {
00931             switch(f->subclass) {
00932             case AST_CONTROL_ANSWER:
00933                /* This is our guy if someone answered. */
00934                if (!peer) {
00935                   ast_verb(3, "%s answered %s\n", c->name, in->name);
00936                   peer = c;
00937                   if (peer->cdr) {
00938                      peer->cdr->answer = ast_tvnow();
00939                      peer->cdr->disposition = AST_CDR_ANSWERED;
00940                   }
00941                   ast_copy_flags64(peerflags, o,
00942                      OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
00943                      OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
00944                      OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
00945                      OPT_CALLEE_PARK | OPT_CALLER_PARK |
00946                      OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
00947                      DIAL_NOFORWARDHTML);
00948                   ast_string_field_set(c, dialcontext, "");
00949                   ast_copy_string(c->exten, "", sizeof(c->exten));
00950                   if (CAN_EARLY_BRIDGE(peerflags, in, peer))
00951                      /* Setup early bridge if appropriate */
00952                      ast_channel_early_bridge(in, peer);
00953                }
00954                /* If call has been answered, then the eventual hangup is likely to be normal hangup */
00955                in->hangupcause = AST_CAUSE_NORMAL_CLEARING;
00956                c->hangupcause = AST_CAUSE_NORMAL_CLEARING;
00957                break;
00958             case AST_CONTROL_BUSY:
00959                ast_verb(3, "%s is busy\n", c->name);
00960                in->hangupcause = c->hangupcause;
00961                ast_hangup(c);
00962                c = o->chan = NULL;
00963                ast_clear_flag64(o, DIAL_STILLGOING);
00964                handle_cause(AST_CAUSE_BUSY, &num);
00965                break;
00966             case AST_CONTROL_CONGESTION:
00967                ast_verb(3, "%s is circuit-busy\n", c->name);
00968                in->hangupcause = c->hangupcause;
00969                ast_hangup(c);
00970                c = o->chan = NULL;
00971                ast_clear_flag64(o, DIAL_STILLGOING);
00972                handle_cause(AST_CAUSE_CONGESTION, &num);
00973                break;
00974             case AST_CONTROL_RINGING:
00975                ast_verb(3, "%s is ringing\n", c->name);
00976                /* Setup early media if appropriate */
00977                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
00978                   ast_channel_early_bridge(in, c);
00979                if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK)) {
00980                   ast_indicate(in, AST_CONTROL_RINGING);
00981                   pa->sentringing++;
00982                }
00983                break;
00984             case AST_CONTROL_PROGRESS:
00985                ast_verb(3, "%s is making progress passing it to %s\n", c->name, in->name);
00986                /* Setup early media if appropriate */
00987                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
00988                   ast_channel_early_bridge(in, c);
00989                if (!ast_test_flag64(outgoing, OPT_RINGBACK))
00990                   if (single || (!single && !pa->sentringing)) {
00991                      ast_indicate(in, AST_CONTROL_PROGRESS);
00992                   }
00993                break;
00994             case AST_CONTROL_VIDUPDATE:
00995                ast_verb(3, "%s requested a video update, passing it to %s\n", c->name, in->name);
00996                ast_indicate(in, AST_CONTROL_VIDUPDATE);
00997                break;
00998             case AST_CONTROL_SRCUPDATE:
00999                ast_verb(3, "%s requested a source update, passing it to %s\n", c->name, in->name);
01000                ast_indicate(in, AST_CONTROL_SRCUPDATE);
01001                break;
01002             case AST_CONTROL_PROCEEDING:
01003                ast_verb(3, "%s is proceeding passing it to %s\n", c->name, in->name);
01004                if (single && CAN_EARLY_BRIDGE(peerflags, in, c))
01005                   ast_channel_early_bridge(in, c);
01006                if (!ast_test_flag64(outgoing, OPT_RINGBACK))
01007                   ast_indicate(in, AST_CONTROL_PROCEEDING);
01008                break;
01009             case AST_CONTROL_HOLD:
01010                ast_verb(3, "Call on %s placed on hold\n", c->name);
01011                ast_indicate(in, AST_CONTROL_HOLD);
01012                break;
01013             case AST_CONTROL_UNHOLD:
01014                ast_verb(3, "Call on %s left from hold\n", c->name);
01015                ast_indicate(in, AST_CONTROL_UNHOLD);
01016                break;
01017             case AST_CONTROL_OFFHOOK:
01018             case AST_CONTROL_FLASH:
01019                /* Ignore going off hook and flash */
01020                break;
01021             case -1:
01022                if (!ast_test_flag64(outgoing, OPT_RINGBACK | OPT_MUSICBACK)) {
01023                   ast_verb(3, "%s stopped sounds\n", c->name);
01024                   ast_indicate(in, -1);
01025                   pa->sentringing = 0;
01026                }
01027                break;
01028             default:
01029                ast_debug(1, "Dunno what to do with control type %d\n", f->subclass);
01030             }
01031          } else if (single) {
01032             switch (f->frametype) {
01033                case AST_FRAME_VOICE:
01034                case AST_FRAME_IMAGE:
01035                case AST_FRAME_TEXT:
01036                   if (ast_write(in, f)) {
01037                      ast_log(LOG_WARNING, "Unable to write frame\n");
01038                   }
01039                   break;
01040                case AST_FRAME_HTML:
01041                   if (!ast_test_flag64(outgoing, DIAL_NOFORWARDHTML) && ast_channel_sendhtml(in, f->subclass, f->data.ptr, f->datalen) == -1) {
01042                      ast_log(LOG_WARNING, "Unable to send URL\n");
01043                   }
01044                   break;
01045                default:
01046                   break;
01047             }
01048          }
01049          ast_frfree(f);
01050       } /* end for */
01051       if (winner == in) {
01052          struct ast_frame *f = ast_read(in);
01053 #if 0
01054          if (f && (f->frametype != AST_FRAME_VOICE))
01055             printf("Frame type: %d, %d\n", f->frametype, f->subclass);
01056          else if (!f || (f->frametype != AST_FRAME_VOICE))
01057             printf("Hangup received on %s\n", in->name);
01058 #endif
01059          if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) {
01060             /* Got hung up */
01061             *to = -1;
01062             strcpy(pa->status, "CANCEL");
01063             ast_cdr_noanswer(in->cdr);
01064             if (f) {
01065                if (f->data.uint32) {
01066                   in->hangupcause = f->data.uint32;
01067                }
01068                ast_frfree(f);
01069             }
01070             return NULL;
01071          }
01072 
01073          /* now f is guaranteed non-NULL */
01074          if (f->frametype == AST_FRAME_DTMF) {
01075             if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
01076                const char *context;
01077                ast_channel_lock(in);
01078                context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
01079                if (onedigit_goto(in, context, (char) f->subclass, 1)) {
01080                   ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
01081                   *to = 0;
01082                   ast_cdr_noanswer(in->cdr);
01083                   *result = f->subclass;
01084                   strcpy(pa->status, "CANCEL");
01085                   ast_frfree(f);
01086                   ast_channel_unlock(in);
01087                   return NULL;
01088                }
01089                ast_channel_unlock(in);
01090             }
01091 
01092             if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
01093                   (f->subclass == '*')) { /* hmm it it not guaranteed to be '*' anymore. */
01094                ast_verb(3, "User hit %c to disconnect call.\n", f->subclass);
01095                *to = 0;
01096                strcpy(pa->status, "CANCEL");
01097                ast_cdr_noanswer(in->cdr);
01098                ast_frfree(f);
01099                return NULL;
01100             }
01101          }
01102 
01103          /* Forward HTML stuff */
01104          if (single && (f->frametype == AST_FRAME_HTML) && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML))
01105             if (ast_channel_sendhtml(outgoing->chan, f->subclass, f->data.ptr, f->datalen) == -1)
01106                ast_log(LOG_WARNING, "Unable to send URL\n");
01107 
01108          if (single && ((f->frametype == AST_FRAME_VOICE) || (f->frametype == AST_FRAME_DTMF_BEGIN) || (f->frametype == AST_FRAME_DTMF_END)))  {
01109             if (ast_write(outgoing->chan, f))
01110                ast_log(LOG_WARNING, "Unable to forward voice or dtmf\n");
01111          }
01112          if (single && (f->frametype == AST_FRAME_CONTROL) &&
01113             ((f->subclass == AST_CONTROL_HOLD) ||
01114             (f->subclass == AST_CONTROL_UNHOLD) ||
01115             (f->subclass == AST_CONTROL_VIDUPDATE) ||
01116              (f->subclass == AST_CONTROL_SRCUPDATE))) {
01117             ast_verb(3, "%s requested special control %d, passing it to %s\n", in->name, f->subclass, outgoing->chan->name);
01118             ast_indicate_data(outgoing->chan, f->subclass, f->data.ptr, f->datalen);
01119          }
01120          ast_frfree(f);
01121       }
01122       if (!*to)
01123          ast_verb(3, "Nobody picked up in %d ms\n", orig);
01124       if (!*to || ast_check_hangup(in))
01125          ast_cdr_noanswer(in->cdr);
01126    }
01127 
01128 #ifdef HAVE_EPOLL
01129    for (epollo = outgoing; epollo; epollo = epollo->next) {
01130       if (epollo->chan)
01131          ast_poll_channel_del(in, epollo->chan);
01132    }
01133 #endif
01134 
01135    return peer;
01136 }
01137 
01138 static void replace_macro_delimiter(char *s)
01139 {
01140    for (; *s; s++)
01141       if (*s == '^')
01142          *s = ',';
01143 }
01144 
01145 /* returns true if there is a valid privacy reply */
01146 static int valid_priv_reply(struct ast_flags64 *opts, int res)
01147 {
01148    if (res < '1')
01149       return 0;
01150    if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
01151       return 1;
01152    if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
01153       return 1;
01154    return 0;
01155 }
01156 
01157 static int do_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
01158    char *parse, struct timeval *calldurationlimit)
01159 {
01160    char *stringp = ast_strdupa(parse);
01161    char *limit_str, *warning_str, *warnfreq_str;
01162    const char *var;
01163    int play_to_caller = 0, play_to_callee = 0;
01164    int delta;
01165 
01166    limit_str = strsep(&stringp, ":");
01167    warning_str = strsep(&stringp, ":");
01168    warnfreq_str = strsep(&stringp, ":");
01169 
01170    config->timelimit = atol(limit_str);
01171    if (warning_str)
01172       config->play_warning = atol(warning_str);
01173    if (warnfreq_str)
01174       config->warning_freq = atol(warnfreq_str);
01175 
01176    if (!config->timelimit) {
01177       ast_log(LOG_WARNING, "Dial does not accept L(%s), hanging up.\n", limit_str);
01178       config->timelimit = config->play_warning = config->warning_freq = 0;
01179       config->warning_sound = NULL;
01180       return -1; /* error */
01181    } else if ( (delta = config->play_warning - config->timelimit) > 0) {
01182       int w = config->warning_freq;
01183 
01184       /* If the first warning is requested _after_ the entire call would end,
01185          and no warning frequency is requested, then turn off the warning. If
01186          a warning frequency is requested, reduce the 'first warning' time by
01187          that frequency until it falls within the call's total time limit.
01188          Graphically:
01189               timelim->|    delta        |<-playwarning
01190          0__________________|_________________|
01191                 | w  |    |    |    |
01192 
01193          so the number of intervals to cut is 1+(delta-1)/w
01194       */
01195 
01196       if (w == 0) {
01197          config->play_warning = 0;
01198       } else {
01199          config->play_warning -= w * ( 1 + (delta-1)/w );
01200          if (config->play_warning < 1)
01201             config->play_warning = config->warning_freq = 0;
01202       }
01203    }
01204    
01205    ast_channel_lock(chan);
01206 
01207    var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
01208 
01209    play_to_caller = var ? ast_true(var) : 1;
01210 
01211    var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
01212    play_to_callee = var ? ast_true(var) : 0;
01213 
01214    if (!play_to_caller && !play_to_callee)
01215       play_to_caller = 1;
01216 
01217    var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
01218    config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
01219 
01220    /* The code looking at config wants a NULL, not just "", to decide
01221     * that the message should not be played, so we replace "" with NULL.
01222     * Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
01223     * not found.
01224     */
01225 
01226    var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
01227    config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
01228 
01229    var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
01230    config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
01231 
01232    ast_channel_unlock(chan);
01233 
01234    /* undo effect of S(x) in case they are both used */
01235    calldurationlimit->tv_sec = 0;
01236    calldurationlimit->tv_usec = 0;
01237 
01238    /* more efficient to do it like S(x) does since no advanced opts */
01239    if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
01240       calldurationlimit->tv_sec = config->timelimit / 1000;
01241       calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
01242       ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
01243          calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
01244       config->timelimit = play_to_caller = play_to_callee =
01245       config->play_warning = config->warning_freq = 0;
01246    } else {
01247       ast_verb(3, "Limit Data for this call:\n");
01248       ast_verb(4, "timelimit      = %ld\n", config->timelimit);
01249       ast_verb(4, "play_warning   = %ld\n", config->play_warning);
01250       ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
01251       ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
01252       ast_verb(4, "warning_freq   = %ld\n", config->warning_freq);
01253       ast_verb(4, "start_sound    = %s\n", S_OR(config->start_sound, ""));
01254       ast_verb(4, "warning_sound  = %s\n", config->warning_sound);
01255       ast_verb(4, "end_sound      = %s\n", S_OR(config->end_sound, ""));
01256    }
01257    if (play_to_caller)
01258       ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
01259    if (play_to_callee)
01260       ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
01261    return 0;
01262 }
01263 
01264 static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
01265    struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
01266 {
01267 
01268    int res2;
01269    int loopcount = 0;
01270 
01271    /* Get the user's intro, store it in priv-callerintros/$CID,
01272       unless it is already there-- this should be done before the
01273       call is actually dialed  */
01274 
01275    /* all ring indications and moh for the caller has been halted as soon as the
01276       target extension was picked up. We are going to have to kill some
01277       time and make the caller believe the peer hasn't picked up yet */
01278 
01279    if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
01280       char *original_moh = ast_strdupa(chan->musicclass);
01281       ast_indicate(chan, -1);
01282       ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
01283       ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
01284       ast_string_field_set(chan, musicclass, original_moh);
01285    } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
01286       ast_indicate(chan, AST_CONTROL_RINGING);
01287       pa->sentringing++;
01288    }
01289 
01290    /* Start autoservice on the other chan ?? */
01291    res2 = ast_autoservice_start(chan);
01292    /* Now Stream the File */
01293    for (loopcount = 0; loopcount < 3; loopcount++) {
01294       if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
01295          break;
01296       if (!res2) /* on timeout, play the message again */
01297          res2 = ast_play_and_wait(peer, "priv-callpending");
01298       if (!valid_priv_reply(opts, res2))
01299          res2 = 0;
01300       /* priv-callpending script:
01301          "I have a caller waiting, who introduces themselves as:"
01302       */
01303       if (!res2)
01304          res2 = ast_play_and_wait(peer, pa->privintro);
01305       if (!valid_priv_reply(opts, res2))
01306          res2 = 0;
01307       /* now get input from the called party, as to their choice */
01308       if (!res2) {
01309          /* XXX can we have both, or they are mutually exclusive ? */
01310          if (ast_test_flag64(opts, OPT_PRIVACY))
01311             res2 = ast_play_and_wait(peer, "priv-callee-options");
01312          if (ast_test_flag64(opts, OPT_SCREENING))
01313             res2 = ast_play_and_wait(peer, "screen-callee-options");
01314       }
01315       /*! \page DialPrivacy Dial Privacy scripts
01316       \par priv-callee-options script:
01317          "Dial 1 if you wish this caller to reach you directly in the future,
01318             and immediately connect to their incoming call
01319           Dial 2 if you wish to send this caller to voicemail now and
01320             forevermore.
01321           Dial 3 to send this caller to the torture menus, now and forevermore.
01322           Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
01323           Dial 5 to allow this caller to come straight thru to you in the future,
01324             but right now, just this once, send them to voicemail."
01325       \par screen-callee-options script:
01326          "Dial 1 if you wish to immediately connect to the incoming call
01327           Dial 2 if you wish to send this caller to voicemail.
01328           Dial 3 to send this caller to the torture menus.
01329           Dial 4 to send this caller to a simple "go away" menu.
01330       */
01331       if (valid_priv_reply(opts, res2))
01332          break;
01333       /* invalid option */
01334       res2 = ast_play_and_wait(peer, "vm-sorry");
01335    }
01336 
01337    if (ast_test_flag64(opts, OPT_MUSICBACK)) {
01338       ast_moh_stop(chan);
01339    } else if (ast_test_flag64(opts, OPT_RINGBACK)) {
01340       ast_indicate(chan, -1);
01341       pa->sentringing = 0;
01342    }
01343    ast_autoservice_stop(chan);
01344    if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
01345       /* map keypresses to various things, the index is res2 - '1' */
01346       static const char *_val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
01347       static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
01348       int i = res2 - '1';
01349       ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
01350          opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
01351       ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
01352    }
01353    switch (res2) {
01354    case '1':
01355       break;
01356    case '2':
01357       ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
01358       break;
01359    case '3':
01360       ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
01361       break;
01362    case '4':
01363       ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
01364       break;
01365    case '5':
01366       /* XXX should we set status to DENY ? */
01367       if (ast_test_flag64(opts, OPT_PRIVACY))
01368          break;
01369       /* if not privacy, then 5 is the same as "default" case */
01370    default: /* bad input or -1 if failure to start autoservice */
01371       /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do?  */
01372       /* well, there seems basically two choices. Just patch the caller thru immediately,
01373            or,... put 'em thru to voicemail. */
01374       /* since the callee may have hung up, let's do the voicemail thing, no database decision */
01375       ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
01376       /* XXX should we set status to DENY ? */
01377       /* XXX what about the privacy flags ? */
01378       break;
01379    }
01380 
01381    if (res2 == '1') { /* the only case where we actually connect */
01382       /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
01383          just clog things up, and it's not useful information, not being tied to a CID */
01384       if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
01385          ast_filedelete(pa->privintro, NULL);
01386          if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
01387             ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
01388          else
01389             ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
01390       }
01391       return 0; /* the good exit path */
01392    } else {
01393       ast_hangup(peer); /* hang up on the callee -- he didn't want to talk anyway! */
01394       return -1;
01395    }
01396 }
01397 
01398 /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
01399 static int setup_privacy_args(struct privacy_args *pa,
01400    struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
01401 {
01402    char callerid[60];
01403    int res;
01404    char *l;
01405    int silencethreshold;
01406 
01407    if (!ast_strlen_zero(chan->cid.cid_num)) {
01408       l = ast_strdupa(chan->cid.cid_num);
01409       ast_shrink_phone_number(l);
01410       if (ast_test_flag64(opts, OPT_PRIVACY) ) {
01411          ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
01412          pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
01413       } else {
01414          ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
01415          pa->privdb_val = AST_PRIVACY_UNKNOWN;
01416       }
01417    } else {
01418       char *tnam, *tn2;
01419 
01420       tnam = ast_strdupa(chan->name);
01421       /* clean the channel name so slashes don't try to end up in disk file name */
01422       for (tn2 = tnam; *tn2; tn2++) {
01423          if (*tn2 == '/')  /* any other chars to be afraid of? */
01424             *tn2 = '=';
01425       }
01426       ast_verb(3, "Privacy-- callerid is empty\n");
01427 
01428       snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", chan->exten, tnam);
01429       l = callerid;
01430       pa->privdb_val = AST_PRIVACY_UNKNOWN;
01431    }
01432 
01433    ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
01434 
01435    if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCLID)) {
01436       /* if callerid is set and OPT_SCREEN_NOCLID is set also */
01437       ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
01438       pa->privdb_val = AST_PRIVACY_ALLOW;
01439    } else if (ast_test_flag64(opts, OPT_SCREEN_NOCLID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
01440       ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
01441    }
01442    
01443    if (pa->privdb_val == AST_PRIVACY_DENY) {
01444       ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
01445       ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
01446       return 0;
01447    } else if (pa->privdb_val == AST_PRIVACY_KILL) {
01448       ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
01449       return 0; /* Is this right? */
01450    } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
01451       ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
01452       return 0; /* is this right??? */
01453    } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
01454       /* Get the user's intro, store it in priv-callerintros/$CID,
01455          unless it is already there-- this should be done before the
01456          call is actually dialed  */
01457 
01458       /* make sure the priv-callerintros dir actually exists */
01459       snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
01460       if ((res = ast_mkdir(pa->privintro, 0755))) {
01461          ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
01462          return -1;
01463       }
01464 
01465       snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
01466       if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
01467          /* the DELUX version of this code would allow this caller the
01468             option to hear and retape their previously recorded intro.
01469          */
01470       } else {
01471          int duration; /* for feedback from play_and_wait */
01472          /* the file doesn't exist yet. Let the caller submit his
01473             vocal intro for posterity */
01474          /* priv-recordintro script:
01475 
01476             "At the tone, please say your name:"
01477 
01478          */
01479          silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
01480          ast_answer(chan);
01481          res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, silencethreshold, 2000, 0);  /* NOTE: I've reduced the total time to 4 sec */
01482                            /* don't think we'll need a lock removed, we took care of
01483                               conflicts by naming the pa.privintro file */
01484          if (res == -1) {
01485             /* Delete the file regardless since they hung up during recording */
01486             ast_filedelete(pa->privintro, NULL);
01487             if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
01488                ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
01489             else
01490                ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
01491             return -1;
01492          }
01493          if (!ast_streamfile(chan, "vm-dialout", chan->language) )
01494             ast_waitstream(chan, "");
01495       }
01496    }
01497    return 1; /* success */
01498 }
01499 
01500 static void end_bridge_callback(void *data)
01501 {
01502    char buf[80];
01503    time_t end;
01504    struct ast_channel *chan = data;
01505 
01506    if (!chan->cdr) {
01507       return;
01508    }
01509 
01510    time(&end);
01511 
01512    ast_channel_lock(chan);
01513    if (chan->cdr->answer.tv_sec) {
01514       snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->answer.tv_sec);
01515       pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
01516    }
01517 
01518    if (chan->cdr->start.tv_sec) {
01519       snprintf(buf, sizeof(buf), "%ld", (long) end - chan->cdr->start.tv_sec);
01520       pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
01521    }
01522    ast_channel_unlock(chan);
01523 }
01524 
01525 static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
01526    bconfig->end_bridge_callback_data = originator;
01527 }
01528 
01529 static int dial_exec_full(struct ast_channel *chan, void *data, struct ast_flags64 *peerflags, int *continue_exec)
01530 {
01531    int res = -1; /* default: error */
01532    char *rest, *cur; /* scan the list of destinations */
01533    struct chanlist *outgoing = NULL; /* list of destinations */
01534    struct ast_channel *peer;
01535    int to; /* timeout */
01536    struct cause_args num = { chan, 0, 0, 0 };
01537    int cause;
01538    char numsubst[256];
01539    char cidname[AST_MAX_EXTENSION] = "";
01540 
01541    struct ast_bridge_config config = { { 0, } };
01542    struct timeval calldurationlimit = { 0, };
01543    char *dtmfcalled = NULL, *dtmfcalling = NULL;
01544    struct privacy_args pa = {
01545       .sentringing = 0,
01546       .privdb_val = 0,
01547       .status = "INVALIDARGS",
01548    };
01549    int sentringing = 0, moh = 0;
01550    const char *outbound_group = NULL;
01551    int result = 0;
01552    char *parse;
01553    int opermode = 0;
01554    int delprivintro = 0;
01555    AST_DECLARE_APP_ARGS(args,
01556       AST_APP_ARG(peers);
01557       AST_APP_ARG(timeout);
01558       AST_APP_ARG(options);
01559       AST_APP_ARG(url);
01560    );
01561    struct ast_flags64 opts = { 0, };
01562    char *opt_args[OPT_ARG_ARRAY_SIZE];
01563    struct ast_datastore *datastore = NULL;
01564    int fulldial = 0, num_dialed = 0;
01565 
01566    /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
01567    pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
01568    pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
01569    pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
01570    pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
01571    pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
01572 
01573    if (ast_strlen_zero(data)) {
01574       ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
01575       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01576       return -1;
01577    }
01578 
01579    parse = ast_strdupa(data);
01580 
01581    AST_STANDARD_APP_ARGS(args, parse);
01582 
01583    if (!ast_strlen_zero(args.options) &&
01584       ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
01585       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01586       goto done;
01587    }
01588 
01589    if (ast_strlen_zero(args.peers)) {
01590       ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
01591       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01592       goto done;
01593    }
01594 
01595 
01596    if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
01597       delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
01598 
01599       if (delprivintro < 0 || delprivintro > 1) {
01600          ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
01601          delprivintro = 0;
01602       }
01603    }
01604 
01605    if (ast_test_flag64(&opts, OPT_OPERMODE)) {
01606       opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
01607       ast_verb(3, "Setting operator services mode to %d.\n", opermode);
01608    }
01609    
01610    if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
01611       calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
01612       if (!calldurationlimit.tv_sec) {
01613          ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
01614          pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01615          goto done;
01616       }
01617       ast_verb(3, "Setting call duration limit to %.3lf milliseconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
01618    }
01619 
01620    if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
01621       dtmfcalling = opt_args[OPT_ARG_SENDDTMF];
01622       dtmfcalled = strsep(&dtmfcalling, ":");
01623    }
01624 
01625    if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
01626       if (do_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
01627          goto done;
01628    }
01629 
01630    if (ast_test_flag64(&opts, OPT_RESETCDR) && chan->cdr)
01631       ast_cdr_reset(chan->cdr, NULL);
01632    if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
01633       opt_args[OPT_ARG_PRIVACY] = ast_strdupa(chan->exten);
01634 
01635    if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
01636       res = setup_privacy_args(&pa, &opts, opt_args, chan);
01637       if (res <= 0)
01638          goto out;
01639       res = -1; /* reset default */
01640    }
01641 
01642    if (ast_test_flag64(&opts, OPT_DTMF_EXIT)) {
01643       __ast_answer(chan, 0, 0);
01644    }
01645 
01646    if (continue_exec)
01647       *continue_exec = 0;
01648 
01649    /* If a channel group has been specified, get it for use when we create peer channels */
01650 
01651    ast_channel_lock(chan);
01652    if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
01653       outbound_group = ast_strdupa(outbound_group);   
01654       pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
01655    } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
01656       outbound_group = ast_strdupa(outbound_group);
01657    }
01658    ast_channel_unlock(chan);  
01659    ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB);
01660 
01661    /* loop through the list of dial destinations */
01662    rest = args.peers;
01663    while ((cur = strsep(&rest, "&")) ) {
01664       struct chanlist *tmp;
01665       struct ast_channel *tc; /* channel for this destination */
01666       /* Get a technology/[device:]number pair */
01667       char *number = cur;
01668       char *interface = ast_strdupa(number);
01669       char *tech = strsep(&number, "/");
01670       /* find if we already dialed this interface */
01671       struct ast_dialed_interface *di;
01672       AST_LIST_HEAD(, ast_dialed_interface) *dialed_interfaces;
01673       num_dialed++;
01674       if (ast_strlen_zero(number)) {
01675          ast_log(LOG_WARNING, "Dial argument takes format (technology/[device:]number1)\n");
01676          goto out;
01677       }
01678       if (!(tmp = ast_calloc(1, sizeof(*tmp))))
01679          goto out;
01680       if (opts.flags) {
01681          ast_copy_flags64(tmp, &opts,
01682             OPT_CANCEL_ELSEWHERE |
01683             OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
01684             OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
01685             OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
01686             OPT_CALLEE_PARK | OPT_CALLER_PARK |
01687             OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
01688             OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID);
01689          ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
01690       }
01691       ast_copy_string(numsubst, number, sizeof(numsubst));
01692       /* Request the peer */
01693 
01694       ast_channel_lock(chan);
01695       datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
01696       ast_channel_unlock(chan);
01697 
01698       if (datastore)
01699          dialed_interfaces = datastore->data;
01700       else {
01701          if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
01702             ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
01703             ast_free(tmp);
01704             goto out;
01705          }
01706 
01707          datastore->inheritance = DATASTORE_INHERIT_FOREVER;
01708 
01709          if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
01710             ast_datastore_free(datastore);
01711             ast_free(tmp);
01712             goto out;
01713          }
01714 
01715          datastore->data = dialed_interfaces;
01716          AST_LIST_HEAD_INIT(dialed_interfaces);
01717 
01718          ast_channel_lock(chan);
01719          ast_channel_datastore_add(chan, datastore);
01720          ast_channel_unlock(chan);
01721       }
01722 
01723       AST_LIST_LOCK(dialed_interfaces);
01724       AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
01725          if (!strcasecmp(di->interface, interface)) {
01726             ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
01727                di->interface);
01728             break;
01729          }
01730       }
01731       AST_LIST_UNLOCK(dialed_interfaces);
01732 
01733       if (di) {
01734          fulldial++;
01735          ast_free(tmp);
01736          continue;
01737       }
01738 
01739       /* It is always ok to dial a Local interface.  We only keep track of
01740        * which "real" interfaces have been dialed.  The Local channel will
01741        * inherit this list so that if it ends up dialing a real interface,
01742        * it won't call one that has already been called. */
01743       if (strcasecmp(tech, "Local")) {
01744          if (!(di = ast_calloc(1, sizeof(*di) + strlen(interface)))) {
01745             AST_LIST_UNLOCK(dialed_interfaces);
01746             ast_free(tmp);
01747             goto out;
01748          }
01749          strcpy(di->interface, interface);
01750 
01751          AST_LIST_LOCK(dialed_interfaces);
01752          AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
01753          AST_LIST_UNLOCK(dialed_interfaces);
01754       }
01755 
01756       tc = ast_request(tech, chan->nativeformats, numsubst, &cause);
01757       if (!tc) {
01758          /* If we can't, just go on to the next call */
01759          ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
01760             tech, cause, ast_cause2str(cause));
01761          handle_cause(cause, &num);
01762          if (!rest) /* we are on the last destination */
01763             chan->hangupcause = cause;
01764          ast_free(tmp);
01765          continue;
01766       }
01767       pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", numsubst);
01768 
01769       /* Setup outgoing SDP to match incoming one */
01770       if (CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
01771          ast_rtp_make_compatible(tc, chan, !outgoing && !rest);
01772       }
01773       
01774       /* Inherit specially named variables from parent channel */
01775       ast_channel_inherit_variables(chan, tc);
01776       ast_channel_datastore_inherit(chan, tc);
01777 
01778       tc->appl = "AppDial";
01779       tc->data = "(Outgoing Line)";
01780       memset(&tc->whentohangup, 0, sizeof(tc->whentohangup));
01781 
01782       S_REPLACE(tc->cid.cid_num, ast_strdup(chan->cid.cid_num));
01783       S_REPLACE(tc->cid.cid_name, ast_strdup(chan->cid.cid_name));
01784       S_REPLACE(tc->cid.cid_ani, ast_strdup(chan->cid.cid_ani));
01785       S_REPLACE(tc->cid.cid_rdnis, ast_strdup(chan->cid.cid_rdnis));
01786       
01787       ast_string_field_set(tc, accountcode, chan->accountcode);
01788       tc->cdrflags = chan->cdrflags;
01789       if (ast_strlen_zero(tc->musicclass))
01790          ast_string_field_set(tc, musicclass, chan->musicclass);
01791       /* Pass callingpres, type of number, tns, ADSI CPE, transfer capability */
01792       tc->cid.cid_pres = chan->cid.cid_pres;
01793       tc->cid.cid_ton = chan->cid.cid_ton;
01794       tc->cid.cid_tns = chan->cid.cid_tns;
01795       tc->cid.cid_ani2 = chan->cid.cid_ani2;
01796       tc->adsicpe = chan->adsicpe;
01797       tc->transfercapability = chan->transfercapability;
01798 
01799       /* If we have an outbound group, set this peer channel to it */
01800       if (outbound_group)
01801          ast_app_group_set_channel(tc, outbound_group);
01802       /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
01803       if (ast_test_flag(chan, AST_FLAG_ANSWERED_ELSEWHERE))
01804          ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
01805 
01806       /* Check if we're forced by configuration */
01807       if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
01808           ast_set_flag(tc, AST_FLAG_ANSWERED_ELSEWHERE);
01809 
01810 
01811       /* Inherit context and extension */
01812       ast_string_field_set(tc, dialcontext, ast_strlen_zero(chan->macrocontext) ? chan->context : chan->macrocontext);
01813       if (!ast_strlen_zero(chan->macroexten))
01814          ast_copy_string(tc->exten, chan->macroexten, sizeof(tc->exten));
01815       else
01816          ast_copy_string(tc->exten, chan->exten, sizeof(tc->exten));
01817 
01818       res = ast_call(tc, numsubst, 0); /* Place the call, but don't wait on the answer */
01819 
01820       /* Save the info in cdr's that we called them */
01821       if (chan->cdr)
01822          ast_cdr_setdestchan(chan->cdr, tc->name);
01823 
01824       /* check the results of ast_call */
01825       if (res) {
01826          /* Again, keep going even if there's an error */
01827          ast_debug(1, "ast call on peer returned %d\n", res);
01828          ast_verb(3, "Couldn't call %s\n", numsubst);
01829          if (tc->hangupcause) {
01830             chan->hangupcause = tc->hangupcause;
01831          }
01832          ast_hangup(tc);
01833          tc = NULL;
01834          ast_free(tmp);
01835          continue;
01836       } else {
01837          senddialevent(chan, tc, numsubst);
01838          ast_verb(3, "Called %s\n", numsubst);
01839          if (!ast_test_flag64(peerflags, OPT_ORIGINAL_CLID))
01840             ast_set_callerid(tc, S_OR(chan->macroexten, chan->exten), get_cid_name(cidname, sizeof(cidname), chan), NULL);
01841       }
01842       /* Put them in the list of outgoing thingies...  We're ready now.
01843          XXX If we're forcibly removed, these outgoing calls won't get
01844          hung up XXX */
01845       ast_set_flag64(tmp, DIAL_STILLGOING);
01846       tmp->chan = tc;
01847       tmp->next = outgoing;
01848       outgoing = tmp;
01849       /* If this line is up, don't try anybody else */
01850       if (outgoing->chan->_state == AST_STATE_UP)
01851          break;
01852    }
01853    
01854    if (ast_strlen_zero(args.timeout)) {
01855       to = -1;
01856    } else {
01857       to = atoi(args.timeout);
01858       if (to > 0)
01859          to *= 1000;
01860       else {
01861          ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
01862          to = -1;
01863       }
01864    }
01865 
01866    if (!outgoing) {
01867       strcpy(pa.status, "CHANUNAVAIL");
01868       if (fulldial == num_dialed) {
01869          res = -1;
01870          goto out;
01871       }
01872    } else {
01873       /* Our status will at least be NOANSWER */
01874       strcpy(pa.status, "NOANSWER");
01875       if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
01876          moh = 1;
01877          if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
01878             char *original_moh = ast_strdupa(chan->musicclass);
01879             ast_string_field_set(chan, musicclass, opt_args[OPT_ARG_MUSICBACK]);
01880             ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
01881             ast_string_field_set(chan, musicclass, original_moh);
01882          } else {
01883             ast_moh_start(chan, NULL, NULL);
01884          }
01885          ast_indicate(chan, AST_CONTROL_PROGRESS);
01886       } else if (ast_test_flag64(outgoing, OPT_RINGBACK)) {
01887          ast_indicate(chan, AST_CONTROL_RINGING);
01888          sentringing++;
01889       }
01890    }
01891 
01892    peer = wait_for_answer(chan, outgoing, &to, peerflags, &pa, &num, &result);
01893 
01894    /* The ast_channel_datastore_remove() function could fail here if the
01895     * datastore was moved to another channel during a masquerade. If this is
01896     * the case, don't free the datastore here because later, when the channel
01897     * to which the datastore was moved hangs up, it will attempt to free this
01898     * datastore again, causing a crash
01899     */
01900    if (!ast_channel_datastore_remove(chan, datastore))
01901       ast_datastore_free(datastore);
01902    if (!peer) {
01903       if (result) {
01904          res = result;
01905       } else if (to) { /* Musta gotten hung up */
01906          res = -1;
01907       } else { /* Nobody answered, next please? */
01908          res = 0;
01909       }
01910 
01911       /* SIP, in particular, sends back this error code to indicate an
01912        * overlap dialled number needs more digits. */
01913       if (chan->hangupcause == AST_CAUSE_INVALID_NUMBER_FORMAT) {
01914          res = AST_PBX_INCOMPLETE;
01915       }
01916 
01917       /* almost done, although the 'else' block is 400 lines */
01918    } else {
01919       const char *number;
01920 
01921       strcpy(pa.status, "ANSWER");
01922       pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
01923       /* Ah ha!  Someone answered within the desired timeframe.  Of course after this
01924          we will always return with -1 so that it is hung up properly after the
01925          conversation.  */
01926       hanguptree(outgoing, peer, 1);
01927       outgoing = NULL;
01928       /* If appropriate, log that we have a destination channel */
01929       if (chan->cdr)
01930          ast_cdr_setdestchan(chan->cdr, peer->name);
01931       if (peer->name)
01932          pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", peer->name);
01933       
01934       ast_channel_lock(peer);
01935       number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER"); 
01936       if (!number)
01937          number = numsubst;
01938       pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
01939       ast_channel_unlock(peer);
01940 
01941       if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
01942          ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
01943          ast_channel_sendurl( peer, args.url );
01944       }
01945       if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
01946          if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
01947             res = 0;
01948             goto out;
01949          }
01950       }
01951       if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
01952          res = 0;
01953       } else {
01954          int digit = 0;
01955          struct ast_channel *chans[2];
01956          struct ast_channel *active_chan;
01957 
01958          chans[0] = chan;
01959          chans[1] = peer;
01960 
01961          /* we need to stream the announcment while monitoring the caller for a hangup */
01962 
01963          /* stream the file */
01964          res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], peer->language);
01965          if (res) {
01966             res = 0;
01967             ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
01968          }
01969 
01970          ast_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
01971          while (peer->stream) {
01972             int ms;
01973 
01974             ms = ast_sched_wait(peer->sched);
01975 
01976             if (ms < 0 && !peer->timingfunc) {
01977                ast_stopstream(peer);
01978                break;
01979             }
01980             if (ms < 0)
01981                ms = 1000;
01982 
01983             active_chan = ast_waitfor_n(chans, 2, &ms);
01984             if (active_chan) {
01985                struct ast_frame *fr = ast_read(active_chan);
01986                if (!fr) {
01987                   ast_hangup(peer);
01988                   res = -1;
01989                   goto done;
01990                }
01991                switch(fr->frametype) {
01992                   case AST_FRAME_DTMF_END:
01993                      digit = fr->subclass;
01994                      if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
01995                         ast_stopstream(peer);
01996                         res = ast_senddigit(chan, digit, 0);
01997                      }
01998                      break;
01999                   case AST_FRAME_CONTROL:
02000                      switch (fr->subclass) {
02001                         case AST_CONTROL_HANGUP:
02002                            ast_frfree(fr);
02003                            ast_hangup(peer);
02004                            res = -1;
02005                            goto done;
02006                         default:
02007                            break;
02008                      }
02009                      break;
02010                   default:
02011                      /* Ignore all others */
02012                      break;
02013                }
02014                ast_frfree(fr);
02015             }
02016             ast_sched_runq(peer->sched);
02017          }
02018          ast_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
02019       }
02020 
02021       if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
02022          /* chan and peer are going into the PBX, they both
02023           * should probably get CDR records. */
02024          ast_clear_flag(chan->cdr, AST_CDR_FLAG_DIALED);
02025          ast_clear_flag(peer->cdr, AST_CDR_FLAG_DIALED);
02026 
02027          replace_macro_delimiter(opt_args[OPT_ARG_GOTO]);
02028          ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
02029          /* peer goes to the same context and extension as chan, so just copy info from chan*/
02030          ast_copy_string(peer->context, chan->context, sizeof(peer->context));
02031          ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
02032          peer->priority = chan->priority + 2;
02033          ast_pbx_start(peer);
02034          hanguptree(outgoing, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
02035          if (continue_exec)
02036             *continue_exec = 1;
02037          res = 0;
02038          goto done;
02039       }
02040 
02041       if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
02042          struct ast_app *theapp;
02043          const char *macro_result;
02044 
02045          res = ast_autoservice_start(chan);
02046          if (res) {
02047             ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
02048             res = -1;
02049          }
02050 
02051          theapp = pbx_findapp("Macro");
02052 
02053          if (theapp && !res) { /* XXX why check res here ? */
02054             /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
02055             ast_copy_string(peer->context, chan->context, sizeof(peer->context));
02056             ast_copy_string(peer->exten, chan->exten, sizeof(peer->exten));
02057 
02058             replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
02059             res = pbx_exec(peer, theapp, opt_args[OPT_ARG_CALLEE_MACRO]);
02060             ast_debug(1, "Macro exited with status %d\n", res);
02061             res = 0;
02062          } else {
02063             ast_log(LOG_ERROR, "Could not find application Macro\n");
02064             res = -1;
02065          }
02066 
02067          if (ast_autoservice_stop(chan) < 0) {
02068             res = -1;
02069          }
02070 
02071          ast_channel_lock(peer);
02072 
02073          if (!res && (macro_result = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
02074             char *macro_transfer_dest;
02075 
02076             if (!strcasecmp(macro_result, "BUSY")) {
02077                ast_copy_string(pa.status, macro_result, sizeof(pa.status));
02078                ast_set_flag64(peerflags, OPT_GO_ON);
02079                res = -1;
02080             } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
02081                ast_copy_string(pa.status, macro_result, sizeof(pa.status));
02082                ast_set_flag64(peerflags, OPT_GO_ON);
02083                res = -1;
02084             } else if (!strcasecmp(macro_result, "CONTINUE")) {
02085                /* hangup peer and keep chan alive assuming the macro has changed
02086                   the context / exten / priority or perhaps
02087                   the next priority in the current exten is desired.
02088                */
02089                ast_set_flag64(peerflags, OPT_GO_ON);
02090                res = -1;
02091             } else if (!strcasecmp(macro_result, "ABORT")) {
02092                /* Hangup both ends unless the caller has the g flag */
02093                res = -1;
02094             } else if (!strncasecmp(macro_result, "GOTO:", 5) && (macro_transfer_dest = ast_strdupa(macro_result + 5))) {
02095                res = -1;
02096                /* perform a transfer to a new extension */
02097                if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
02098                   replace_macro_delimiter(macro_transfer_dest);
02099                   if (!ast_parseable_goto(chan, macro_transfer_dest))
02100                      ast_set_flag64(peerflags, OPT_GO_ON);
02101                }
02102             }
02103          }
02104 
02105          ast_channel_unlock(peer);
02106       }
02107 
02108       if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
02109          struct ast_app *theapp;
02110          const char *gosub_result;
02111          char *gosub_args, *gosub_argstart;
02112          int res9 = -1;
02113 
02114          res9 = ast_autoservice_start(chan);
02115          if (res9) {
02116             ast_log(LOG_ERROR, "Unable to start autoservice on calling channel\n");
02117             res9 = -1;
02118          }
02119 
02120          theapp = pbx_findapp("Gosub");
02121 
02122          if (theapp && !res9) {
02123             replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
02124 
02125             /* Set where we came from */
02126             ast_copy_string(peer->context, "app_dial_gosub_virtual_context", sizeof(peer->context));
02127             ast_copy_string(peer->exten, "s", sizeof(peer->exten));
02128             peer->priority = 0;
02129 
02130             gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
02131             if (gosub_argstart) {
02132                *gosub_argstart = 0;
02133                if (asprintf(&gosub_args, "%s,s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], gosub_argstart + 1) < 0) {
02134                   ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
02135                   gosub_args = NULL;
02136                }
02137                *gosub_argstart = ',';
02138             } else {
02139                if (asprintf(&gosub_args, "%s,s,1", opt_args[OPT_ARG_CALLEE_GOSUB]) < 0) {
02140                   ast_log(LOG_WARNING, "asprintf() failed: %s\n", strerror(errno));
02141                   gosub_args = NULL;
02142                }
02143             }
02144 
02145             if (gosub_args) {
02146                res9 = pbx_exec(peer, theapp, gosub_args);
02147                if (!res9) {
02148                   struct ast_pbx_args args;
02149                   /* A struct initializer fails to compile for this case ... */
02150                   memset(&args, 0, sizeof(args));
02151                   args.no_hangup_chan = 1;
02152                   ast_pbx_run_args(peer, &args);
02153                }
02154                ast_free(gosub_args);
02155                ast_debug(1, "Gosub exited with status %d\n", res9);
02156             } else {
02157                ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
02158             }
02159 
02160          } else if (!res9) {
02161             ast_log(LOG_ERROR, "Could not find application Gosub\n");
02162             res9 = -1;
02163          }
02164 
02165          if (ast_autoservice_stop(chan) < 0) {
02166             ast_log(LOG_ERROR, "Could not stop autoservice on calling channel\n");
02167             res9 = -1;
02168          }
02169          
02170          ast_channel_lock(peer);
02171 
02172          if (!res9 && (gosub_result = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
02173             char *gosub_transfer_dest;
02174 
02175             if (!strcasecmp(gosub_result, "BUSY")) {
02176                ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
02177                ast_set_flag64(peerflags, OPT_GO_ON);
02178                res = -1;
02179             } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
02180                ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
02181                ast_set_flag64(peerflags, OPT_GO_ON);
02182                res = -1;
02183             } else if (!strcasecmp(gosub_result, "CONTINUE")) {
02184                /* hangup peer and keep chan alive assuming the macro has changed
02185                   the context / exten / priority or perhaps
02186                   the next priority in the current exten is desired.
02187                */
02188                ast_set_flag64(peerflags, OPT_GO_ON);
02189                res = -1;
02190             } else if (!strcasecmp(gosub_result, "ABORT")) {
02191                /* Hangup both ends unless the caller has the g flag */
02192                res = -1;
02193             } else if (!strncasecmp(gosub_result, "GOTO:", 5) && (gosub_transfer_dest = ast_strdupa(gosub_result + 5))) {
02194                res = -1;
02195                /* perform a transfer to a new extension */
02196                if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
02197                   replace_macro_delimiter(gosub_transfer_dest);
02198                   if (!ast_parseable_goto(chan, gosub_transfer_dest))
02199                      ast_set_flag64(peerflags, OPT_GO_ON);
02200                }
02201             }
02202          }
02203 
02204          ast_channel_unlock(peer);  
02205       }
02206 
02207       if (!res) {
02208          if (!ast_tvzero(calldurationlimit)) {
02209             struct timeval whentohangup = calldurationlimit;
02210             peer->whentohangup = ast_tvadd(ast_tvnow(), whentohangup);
02211          }
02212          if (!ast_strlen_zero(dtmfcalled)) {
02213             ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
02214             res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
02215          }
02216          if (!ast_strlen_zero(dtmfcalling)) {
02217             ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
02218             res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
02219          }
02220       }
02221 
02222       if (res) { /* some error */
02223          res = -1;
02224       } else {
02225          if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
02226             ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
02227          if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
02228             ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
02229          if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
02230             ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
02231          if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
02232             ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
02233          if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
02234             ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
02235          if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
02236             ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
02237          if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
02238             ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
02239          if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
02240             ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
02241          if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
02242             ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
02243          if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
02244             ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
02245          if (ast_test_flag64(peerflags, OPT_GO_ON))
02246             ast_set_flag(&(config.features_caller), AST_FEATURE_NO_H_EXTEN);
02247 
02248          config.end_bridge_callback = end_bridge_callback;
02249          config.end_bridge_callback_data = chan;
02250          config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
02251          
02252          if (moh) {
02253             moh = 0;
02254             ast_moh_stop(chan);
02255          } else if (sentringing) {
02256             sentringing = 0;
02257             ast_indicate(chan, -1);
02258          }
02259          /* Be sure no generators are left on it and reset the visible indication */
02260          ast_deactivate_generator(chan);
02261          chan->visible_indication = 0;
02262          /* Make sure channels are compatible */
02263          res = ast_channel_make_compatible(chan, peer);
02264          if (res < 0) {
02265             ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", chan->name, peer->name);
02266             ast_hangup(peer);
02267             res = -1;
02268             goto done;
02269          }
02270          if (opermode) {
02271             struct oprmode oprmode;
02272 
02273             oprmode.peer = peer;
02274             oprmode.mode = opermode;
02275 
02276             ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
02277          }
02278          res = ast_bridge_call(chan, peer, &config);
02279       }
02280 
02281       strcpy(peer->context, chan->context);
02282 
02283       if (ast_test_flag64(&opts, OPT_PEER_H) && ast_exists_extension(peer, peer->context, "h", 1, peer->cid.cid_num)) {
02284          int autoloopflag;
02285          int found;
02286          int res9;
02287          
02288          strcpy(peer->exten, "h");
02289          peer->priority = 1;
02290          autoloopflag = ast_test_flag(peer, AST_FLAG_IN_AUTOLOOP); /* save value to restore at the end */
02291          ast_set_flag(peer, AST_FLAG_IN_AUTOLOOP);
02292 
02293          while ((res9 = ast_spawn_extension(peer, peer->context, peer->exten, peer->priority, peer->cid.cid_num, &found, 1)) == 0)
02294             peer->priority++;
02295 
02296          if (found && res9) {
02297             /* Something bad happened, or a hangup has been requested. */
02298             ast_debug(1, "Spawn extension (%s,%s,%d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
02299             ast_verb(2, "Spawn extension (%s, %s, %d) exited non-zero on '%s'\n", peer->context, peer->exten, peer->priority, peer->name);
02300          }
02301          ast_set2_flag(peer, autoloopflag, AST_FLAG_IN_AUTOLOOP);  /* set it back the way it was */
02302       }
02303       if (!ast_check_hangup(peer) && ast_test_flag64(&opts, OPT_CALLEE_GO_ON) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GO_ON])) {      
02304          replace_macro_delimiter(opt_args[OPT_ARG_CALLEE_GO_ON]);
02305          ast_parseable_goto(peer, opt_args[OPT_ARG_CALLEE_GO_ON]);
02306          ast_pbx_start(peer);
02307       } else {
02308          if (!ast_check_hangup(chan))
02309             chan->hangupcause = peer->hangupcause;
02310          ast_hangup(peer);
02311       }
02312    }
02313 out:
02314    if (moh) {
02315       moh = 0;
02316       ast_moh_stop(chan);
02317    } else if (sentringing) {
02318       sentringing = 0;
02319       ast_indicate(chan, -1);
02320    }
02321 
02322    if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
02323       ast_filedelete(pa.privintro, NULL);
02324       if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
02325          ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
02326       } else {
02327          ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
02328       }
02329    }
02330 
02331    ast_channel_early_bridge(chan, NULL);
02332    hanguptree(outgoing, NULL, 0); /* In this case, there's no answer anywhere */
02333    pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
02334    senddialendevent(chan, pa.status);
02335    ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
02336    
02337    if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
02338       if (!ast_tvzero(calldurationlimit))
02339          memset(&chan->whentohangup, 0, sizeof(chan->whentohangup));
02340       res = 0;
02341    }
02342 
02343 done:
02344    if (config.warning_sound) {
02345       ast_free((char *)config.warning_sound);
02346    }
02347    if (config.end_sound) {
02348       ast_free((char *)config.end_sound);
02349    }
02350    if (config.start_sound) {
02351       ast_free((char *)config.start_sound);
02352    }
02353    return res;
02354 }
02355 
02356 static int dial_exec(struct ast_channel *chan, void *data)
02357 {
02358    struct ast_flags64 peerflags;
02359 
02360    memset(&peerflags, 0, sizeof(peerflags));
02361 
02362    return dial_exec_full(chan, data, &peerflags, NULL);
02363 }
02364 
02365 static int retrydial_exec(struct ast_channel *chan, void *data)
02366 {
02367    char *parse;
02368    const char *context = NULL;
02369    int sleepms = 0, loops = 0, res = -1;
02370    struct ast_flags64 peerflags = { 0, };
02371    AST_DECLARE_APP_ARGS(args,
02372       AST_APP_ARG(announce);
02373       AST_APP_ARG(sleep);
02374       AST_APP_ARG(retries);
02375       AST_APP_ARG(dialdata);
02376    );
02377 
02378    if (ast_strlen_zero(data)) {
02379       ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
02380       return -1;
02381    }
02382 
02383    parse = ast_strdupa(data);
02384    AST_STANDARD_APP_ARGS(args, parse);
02385 
02386    if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
02387       sleepms *= 1000;
02388 
02389    if (!ast_strlen_zero(args.retries)) {
02390       loops = atoi(args.retries);
02391    }
02392 
02393    if (!args.dialdata) {
02394       ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
02395       goto done;
02396    }
02397 
02398    if (sleepms < 1000)
02399       sleepms = 10000;
02400 
02401    if (!loops)
02402       loops = -1; /* run forever */
02403 
02404    ast_channel_lock(chan);
02405    context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
02406    context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
02407    ast_channel_unlock(chan);
02408 
02409    res = 0;
02410    while (loops) {
02411       int continue_exec;
02412 
02413       chan->data = "Retrying";
02414       if (ast_test_flag(chan, AST_FLAG_MOH))
02415          ast_moh_stop(chan);
02416 
02417       res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
02418       if (continue_exec)
02419          break;
02420 
02421       if (res == 0) {
02422          if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
02423             if (!ast_strlen_zero(args.announce)) {
02424                if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
02425                   if (!(res = ast_streamfile(chan, args.announce, chan->language)))
02426                      ast_waitstream(chan, AST_DIGIT_ANY);
02427                } else
02428                   ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
02429             }
02430             if (!res && sleepms) {
02431                if (!ast_test_flag(chan, AST_FLAG_MOH))
02432                   ast_moh_start(chan, NULL, NULL);
02433                res = ast_waitfordigit(chan, sleepms);
02434             }
02435          } else {
02436             if (!ast_strlen_zero(args.announce)) {
02437                if (ast_fileexists(args.announce, NULL, chan->language) > 0) {
02438                   if (!(res = ast_streamfile(chan, args.announce, chan->language)))
02439                      res = ast_waitstream(chan, "");
02440                } else
02441                   ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
02442             }
02443             if (sleepms) {
02444                if (!ast_test_flag(chan, AST_FLAG_MOH))
02445                   ast_moh_start(chan, NULL, NULL);
02446                if (!res)
02447                   res = ast_waitfordigit(chan, sleepms);
02448             }
02449          }
02450       }
02451 
02452       if (res < 0 || res == AST_PBX_INCOMPLETE) {
02453          break;
02454       } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
02455          if (onedigit_goto(chan, context, (char) res, 1)) {
02456             res = 0;
02457             break;
02458          }
02459       }
02460       loops--;
02461    }
02462    if (loops == 0)
02463       res = 0;
02464    else if (res == 1)
02465       res = 0;
02466 
02467    if (ast_test_flag(chan, AST_FLAG_MOH))
02468       ast_moh_stop(chan);
02469  done:
02470    return res;
02471 }
02472 
02473 static int unload_module(void)
02474 {
02475    int res;
02476    struct ast_context *con;
02477 
02478    res = ast_unregister_application(app);
02479    res |= ast_unregister_application(rapp);
02480 
02481    if ((con = ast_context_find("app_dial_gosub_virtual_context"))) {
02482       ast_context_remove_extension2(con, "s", 1, NULL, 0);
02483       ast_context_destroy(con, "app_dial"); /* leave nothing behind */
02484    }
02485 
02486    return res;
02487 }
02488 
02489 static int load_module(void)
02490 {
02491    int res;
02492    struct ast_context *con;
02493 
02494    con = ast_context_find_or_create(NULL, NULL, "app_dial_gosub_virtual_context", "app_dial");
02495    if (!con)
02496       ast_log(LOG_ERROR, "Dial virtual context 'app_dial_gosub_virtual_context' does not exist and unable to create\n");
02497    else
02498       ast_add_extension2(con, 1, "s", 1, NULL, NULL, "NoOp", ast_strdup(""), ast_free_ptr, "app_dial");
02499 
02500    res = ast_register_application_xml(app, dial_exec);
02501    res |= ast_register_application_xml(rapp, retrydial_exec);
02502 
02503    return res;
02504 }
02505 
02506 AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");