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- /*
- * Asterisk -- An open source telephony toolkit.
- *
- * Copyright (C) 1999 - 2012, Digium, Inc.
- *
- * Mark Spencer <markster@digium.com>
- *
- * See http://www.asterisk.org for more information about
- * the Asterisk project. Please do not directly contact
- * any of the maintainers of this project for assistance;
- * the project provides a web site, mailing lists and IRC
- * channels for your use.
- *
- * This program is free software, distributed under the terms of
- * the GNU General Public License Version 2. See the LICENSE file
- * at the top of the source tree.
- */
- /*! \file
- *
- * \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
- *
- * \author Mark Spencer <markster@digium.com>
- *
- * \ingroup applications
- */
- /*** MODULEINFO
- <support_level>core</support_level>
- ***/
- #include "asterisk.h"
- ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
- #include <sys/time.h>
- #include <sys/signal.h>
- #include <sys/stat.h>
- #include <netinet/in.h>
- #include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
- #include "asterisk/lock.h"
- #include "asterisk/file.h"
- #include "asterisk/channel.h"
- #include "asterisk/pbx.h"
- #include "asterisk/module.h"
- #include "asterisk/translate.h"
- #include "asterisk/say.h"
- #include "asterisk/config.h"
- #include "asterisk/features.h"
- #include "asterisk/musiconhold.h"
- #include "asterisk/callerid.h"
- #include "asterisk/utils.h"
- #include "asterisk/app.h"
- #include "asterisk/causes.h"
- #include "asterisk/rtp_engine.h"
- #include "asterisk/manager.h"
- #include "asterisk/privacy.h"
- #include "asterisk/stringfields.h"
- #include "asterisk/dsp.h"
- #include "asterisk/aoc.h"
- #include "asterisk/ccss.h"
- #include "asterisk/indications.h"
- #include "asterisk/framehook.h"
- #include "asterisk/dial.h"
- #include "asterisk/stasis_channels.h"
- #include "asterisk/bridge_after.h"
- #include "asterisk/features_config.h"
- #include "asterisk/max_forwards.h"
- /*** DOCUMENTATION
- <application name="Dial" language="en_US">
- <synopsis>
- Attempt to connect to another device or endpoint and bridge the call.
- </synopsis>
- <syntax>
- <parameter name="Technology/Resource" required="true" argsep="&">
- <argument name="Technology/Resource" required="true">
- <para>Specification of the device(s) to dial. These must be in the format of
- <literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
- represents a particular channel driver, and <replaceable>Resource</replaceable>
- represents a resource available to that particular channel driver.</para>
- </argument>
- <argument name="Technology2/Resource2" required="false" multiple="true">
- <para>Optional extra devices to dial in parallel</para>
- <para>If you need more than one enter them as
- Technology2/Resource2&Technology3/Resource3&.....</para>
- </argument>
- </parameter>
- <parameter name="timeout" required="false">
- <para>Specifies the number of seconds we attempt to dial the specified devices.</para>
- <para>If not specified, this defaults to 136 years.</para>
- </parameter>
- <parameter name="options" required="false">
- <optionlist>
- <option name="A">
- <argument name="x" required="true">
- <para>The file to play to the called party</para>
- </argument>
- <para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
- </option>
- <option name="a">
- <para>Immediately answer the calling channel when the called channel answers in
- all cases. Normally, the calling channel is answered when the called channel
- answers, but when options such as <literal>A()</literal> and
- <literal>M()</literal> are used, the calling channel is
- not answered until all actions on the called channel (such as playing an
- announcement) are completed. This option can be used to answer the calling
- channel before doing anything on the called channel. You will rarely need to use
- this option, the default behavior is adequate in most cases.</para>
- </option>
- <option name="b" argsep="^">
- <para>Before initiating an outgoing call, <literal>Gosub</literal> to the specified
- location using the newly created channel. The <literal>Gosub</literal> will be
- executed for each destination channel.</para>
- <argument name="context" required="false" />
- <argument name="exten" required="false" />
- <argument name="priority" required="true" hasparams="optional" argsep="^">
- <argument name="arg1" multiple="true" required="true" />
- <argument name="argN" />
- </argument>
- </option>
- <option name="B" argsep="^">
- <para>Before initiating the outgoing call(s), <literal>Gosub</literal> to the
- specified location using the current channel.</para>
- <argument name="context" required="false" />
- <argument name="exten" required="false" />
- <argument name="priority" required="true" hasparams="optional" argsep="^">
- <argument name="arg1" multiple="true" required="true" />
- <argument name="argN" />
- </argument>
- </option>
- <option name="C">
- <para>Reset the call detail record (CDR) for this call.</para>
- </option>
- <option name="c">
- <para>If the Dial() application cancels this call, always set
- <variable>HANGUPCAUSE</variable> to 'answered elsewhere'</para>
- </option>
- <option name="d">
- <para>Allow the calling user to dial a 1 digit extension while waiting for
- a call to be answered. Exit to that extension if it exists in the
- current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
- if it exists.</para>
- <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
- connected. If you wish to use this option with these phones, you
- can use the <literal>Answer</literal> application before dialing.</para>
- </option>
- <option name="D" argsep=":">
- <argument name="called" />
- <argument name="calling" />
- <argument name="progress" />
- <para>Send the specified DTMF strings <emphasis>after</emphasis> the called
- party has answered, but before the call gets bridged. The
- <replaceable>called</replaceable> DTMF string is sent to the called party, and the
- <replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
- can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
- to the called party immediately after receiving a <literal>PROGRESS</literal> message.</para>
- <para>See <literal>SendDTMF</literal> for valid digits.</para>
- </option>
- <option name="e">
- <para>Execute the <literal>h</literal> extension for peer after the call ends</para>
- </option>
- <option name="f">
- <argument name="x" required="false" />
- <para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
- deflection to the dialplan extension of this <literal>Dial()</literal> using a dialplan <literal>hint</literal>.
- For example, some PSTNs do not allow CallerID to be set to anything
- other than the numbers assigned to you.
- If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
- </option>
- <option name="F" argsep="^">
- <argument name="context" required="false" />
- <argument name="exten" required="false" />
- <argument name="priority" required="true" />
- <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
- to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
- <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').</para>
- </option>
- <option name="F">
- <para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
- and <emphasis>start</emphasis> execution at that location.</para>
- <para>NOTE: Any channel variables you want the called channel to inherit from the caller channel must be
- prefixed with one or two underbars ('_').</para>
- <para>NOTE: Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
- </option>
- <option name="g">
- <para>Proceed with dialplan execution at the next priority in the current extension if the
- destination channel hangs up.</para>
- </option>
- <option name="G" argsep="^">
- <argument name="context" required="false" />
- <argument name="exten" required="false" />
- <argument name="priority" required="true" />
- <para>If the call is answered, transfer the calling party to
- the specified <replaceable>priority</replaceable> and the called party to the specified
- <replaceable>priority</replaceable> plus one.</para>
- <para>NOTE: You cannot use any additional action post answer options in conjunction with this option.</para>
- </option>
- <option name="h">
- <para>Allow the called party to hang up by sending the DTMF sequence
- defined for disconnect in <filename>features.conf</filename>.</para>
- </option>
- <option name="H">
- <para>Allow the calling party to hang up by sending the DTMF sequence
- defined for disconnect in <filename>features.conf</filename>.</para>
- <para>NOTE: Many SIP and ISDN phones cannot send DTMF digits until the call is
- connected. If you wish to allow DTMF disconnect before the dialed
- party answers with these phones, you can use the <literal>Answer</literal>
- application before dialing.</para>
- </option>
- <option name="i">
- <para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
- </option>
- <option name="I">
- <para>Asterisk will ignore any connected line update requests or any redirecting party
- update requests it may receive on this dial attempt.</para>
- </option>
- <option name="k">
- <para>Allow the called party to enable parking of the call by sending
- the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
- </option>
- <option name="K">
- <para>Allow the calling party to enable parking of the call by sending
- the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
- </option>
- <option name="L" argsep=":">
- <argument name="x" required="true">
- <para>Maximum call time, in milliseconds</para>
- </argument>
- <argument name="y">
- <para>Warning time, in milliseconds</para>
- </argument>
- <argument name="z">
- <para>Repeat time, in milliseconds</para>
- </argument>
- <para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
- left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
- <para>This option is affected by the following variables:</para>
- <variablelist>
- <variable name="LIMIT_PLAYAUDIO_CALLER">
- <value name="yes" default="true" />
- <value name="no" />
- <para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
- </variable>
- <variable name="LIMIT_PLAYAUDIO_CALLEE">
- <value name="yes" />
- <value name="no" default="true"/>
- <para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
- </variable>
- <variable name="LIMIT_TIMEOUT_FILE">
- <value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
- If not set, the time remaining will be announced.</para>
- </variable>
- <variable name="LIMIT_CONNECT_FILE">
- <value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
- If not set, the time remaining will be announced.</para>
- </variable>
- <variable name="LIMIT_WARNING_FILE">
- <value name="filename"/>
- <para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
- a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
- </variable>
- </variablelist>
- </option>
- <option name="m">
- <argument name="class" required="false"/>
- <para>Provide hold music to the calling party until a requested
- channel answers. A specific music on hold <replaceable>class</replaceable>
- (as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
- </option>
- <option name="M" argsep="^">
- <argument name="macro" required="true">
- <para>Name of the macro that should be executed.</para>
- </argument>
- <argument name="arg" multiple="true">
- <para>Macro arguments</para>
- </argument>
- <para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
- before connecting to the calling channel. Arguments can be specified to the Macro
- using <literal>^</literal> as a delimiter. The macro can set the variable
- <variable>MACRO_RESULT</variable> to specify the following actions after the macro is
- finished executing:</para>
- <variablelist>
- <variable name="MACRO_RESULT">
- <para>If set, this action will be taken after the macro finished executing.</para>
- <value name="ABORT">
- Hangup both legs of the call
- </value>
- <value name="CONGESTION">
- Behave as if line congestion was encountered
- </value>
- <value name="BUSY">
- Behave as if a busy signal was encountered
- </value>
- <value name="CONTINUE">
- Hangup the called party and allow the calling party to continue dialplan execution at the next priority
- </value>
- <value name="GOTO:[[<context>^]<exten>^]<priority>">
- Transfer the call to the specified destination.
- </value>
- </variable>
- </variablelist>
- <para>NOTE: You cannot use any additional action post answer options in conjunction
- with this option. Also, pbx services are run on the peer (called) channel,
- so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this macro.</para>
- <para>WARNING: Be aware of the limitations that macros have, specifically with regards to use of
- the <literal>WaitExten</literal> application. For more information, see the documentation for
- <literal>Macro()</literal>.</para>
- </option>
- <option name="n">
- <argument name="delete">
- <para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
- the recorded introduction will not be deleted if the caller hangs up while the remote party has not
- yet answered.</para>
- <para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
- always be deleted.</para>
- </argument>
- <para>This option is a modifier for the call screening/privacy mode. (See the
- <literal>p</literal> and <literal>P</literal> options.) It specifies
- that no introductions are to be saved in the <directory>priv-callerintros</directory>
- directory.</para>
- </option>
- <option name="N">
- <para>This option is a modifier for the call screening/privacy mode. It specifies
- that if CallerID is present, do not screen the call.</para>
- </option>
- <option name="o">
- <argument name="x" required="false" />
- <para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
- <emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
- This was the behavior of Asterisk 1.0 and earlier.
- If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
- Note that <literal>o(${CALLERID(all)})</literal> is similar to option <literal>o</literal> without the parameter.</para>
- </option>
- <option name="O">
- <argument name="mode">
- <para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
- the originator hanging up will cause the phone to ring back immediately.</para>
- <para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
- flashes the trunk, it will ring their phone back.</para>
- </argument>
- <para>Enables <emphasis>operator services</emphasis> mode. This option only
- works when bridging a DAHDI channel to another DAHDI channel
- only. if specified on non-DAHDI interfaces, it will be ignored.
- When the destination answers (presumably an operator services
- station), the originator no longer has control of their line.
- They may hang up, but the switch will not release their line
- until the destination party (the operator) hangs up.</para>
- </option>
- <option name="p">
- <para>This option enables screening mode. This is basically Privacy mode
- without memory.</para>
- </option>
- <option name="P">
- <argument name="x" />
- <para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
- it is provided. The current extension is used if a database family/key is not specified.</para>
- </option>
- <option name="Q">
- <argument name="cause" required="true"/>
- <para>Specify the Q.850/Q.931 <replaceable>cause</replaceable> to send on
- unanswered channels when another channel answers the call.
- As with <literal>Hangup()</literal>, <replaceable>cause</replaceable>
- can be a numeric cause code or a name such as
- <literal>NO_ANSWER</literal>,
- <literal>USER_BUSY</literal>,
- <literal>CALL_REJECTED</literal> or
- <literal>ANSWERED_ELSEWHERE</literal> (the default if Q isn't specified).
- You can also specify <literal>0</literal> or <literal>NONE</literal>
- to send no cause. See the <filename>causes.h</filename> file for the
- full list of valid causes and names.
- </para>
- <para>NOTE: chan_sip does not support setting the cause on a CANCEL to anything
- other than ANSWERED_ELSEWHERE.</para>
- </option>
- <option name="r">
- <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
- party until the called channel has answered.</para>
- <argument name="tone" required="false">
- <para>Indicate progress to calling party. Send audio 'tone' from the <filename>indications.conf</filename> tonezone currently in use.</para>
- </argument>
- </option>
- <option name="R">
- <para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing.
- Allow interruption of the ringback if early media is received on the channel.</para>
- </option>
- <option name="S">
- <argument name="x" required="true" />
- <para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
- answered the call.</para>
- </option>
- <option name="s">
- <argument name="x" required="true" />
- <para>Force the outgoing CallerID tag parameter to be set to the string <replaceable>x</replaceable>.</para>
- <para>Works with the <literal>f</literal> option.</para>
- </option>
- <option name="t">
- <para>Allow the called party to transfer the calling party by sending the
- DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
- transfers initiated by other methods.</para>
- </option>
- <option name="T">
- <para>Allow the calling party to transfer the called party by sending the
- DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
- transfers initiated by other methods.</para>
- </option>
- <option name="U" argsep="^">
- <argument name="x" required="true">
- <para>Name of the subroutine context to execute via <literal>Gosub</literal>.
- The subroutine execution starts in the named context at the s exten and priority 1.</para>
- </argument>
- <argument name="arg" multiple="true" required="false">
- <para>Arguments for the <literal>Gosub</literal> routine</para>
- </argument>
- <para>Execute via <literal>Gosub</literal> the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
- to the calling channel. Arguments can be specified to the <literal>Gosub</literal>
- using <literal>^</literal> as a delimiter. The <literal>Gosub</literal> routine can set the variable
- <variable>GOSUB_RESULT</variable> to specify the following actions after the <literal>Gosub</literal> returns.</para>
- <variablelist>
- <variable name="GOSUB_RESULT">
- <value name="ABORT">
- Hangup both legs of the call.
- </value>
- <value name="CONGESTION">
- Behave as if line congestion was encountered.
- </value>
- <value name="BUSY">
- Behave as if a busy signal was encountered.
- </value>
- <value name="CONTINUE">
- Hangup the called party and allow the calling party
- to continue dialplan execution at the next priority.
- </value>
- <value name="GOTO:[[<context>^]<exten>^]<priority>">
- Transfer the call to the specified destination.
- </value>
- </variable>
- </variablelist>
- <para>NOTE: You cannot use any additional action post answer options in conjunction
- with this option. Also, pbx services are run on the <emphasis>called</emphasis> channel,
- so you will not be able to set timeouts via the <literal>TIMEOUT()</literal> function in this routine.</para>
- </option>
- <option name="u">
- <argument name = "x" required="true">
- <para>Force the outgoing callerid presentation indicator parameter to be set
- to one of the values passed in <replaceable>x</replaceable>:
- <literal>allowed_not_screened</literal>
- <literal>allowed_passed_screen</literal>
- <literal>allowed_failed_screen</literal>
- <literal>allowed</literal>
- <literal>prohib_not_screened</literal>
- <literal>prohib_passed_screen</literal>
- <literal>prohib_failed_screen</literal>
- <literal>prohib</literal>
- <literal>unavailable</literal></para>
- </argument>
- <para>Works with the <literal>f</literal> option.</para>
- </option>
- <option name="w">
- <para>Allow the called party to enable recording of the call by sending
- the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
- </option>
- <option name="W">
- <para>Allow the calling party to enable recording of the call by sending
- the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
- </option>
- <option name="x">
- <para>Allow the called party to enable recording of the call by sending
- the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
- </option>
- <option name="X">
- <para>Allow the calling party to enable recording of the call by sending
- the DTMF sequence defined for one-touch automixmonitor in <filename>features.conf</filename>.</para>
- </option>
- <option name="z">
- <para>On a call forward, cancel any dial timeout which has been set for this call.</para>
- </option>
- </optionlist>
- </parameter>
- <parameter name="URL">
- <para>The optional URL will be sent to the called party if the channel driver supports it.</para>
- </parameter>
- </syntax>
- <description>
- <para>This application will place calls to one or more specified channels. As soon
- as one of the requested channels answers, the originating channel will be
- answered, if it has not already been answered. These two channels will then
- be active in a bridged call. All other channels that were requested will then
- be hung up.</para>
- <para>Unless there is a timeout specified, the Dial application will wait
- indefinitely until one of the called channels answers, the user hangs up, or
- if all of the called channels are busy or unavailable. Dialplan execution will
- continue if no requested channels can be called, or if the timeout expires.
- This application will report normal termination if the originating channel
- hangs up, or if the call is bridged and either of the parties in the bridge
- ends the call.</para>
- <para>If the <variable>OUTBOUND_GROUP</variable> variable is set, all peer channels created by this
- application will be put into that group (as in <literal>Set(GROUP()=...</literal>).
- If the <variable>OUTBOUND_GROUP_ONCE</variable> variable is set, all peer channels created by this
- application will be put into that group (as in <literal>Set(GROUP()=...</literal>). Unlike <variable>OUTBOUND_GROUP</variable>,
- however, the variable will be unset after use.</para>
- <example title="Dial with 30 second timeout">
- same => n,Dial(PJSIP/alice,30)
- </example>
- <example title="Parallel dial with 45 second timeout">
- same => n,Dial(PJSIP/alice&PJIP/bob,45)
- </example>
- <example title="Dial with 'g' continuation option">
- same => n,Dial(PJSIP/alice,,g)
- same => n,Log(NOTICE, Alice call result: ${DIALSTATUS})
- </example>
- <example title="Dial with transfer/recording features for calling party">
- same => n,Dial(PJSIP/alice,,TX)
- </example>
- <example title="Dial with call length limit">
- same => n,Dial(PJSIP/alice,,L(60000:30000:10000))
- </example>
- <example title="Dial alice and bob and send NO_ANSWER to bob instead of ANSWERED_ELSEWHERE when alice answers">
- same => n,Dial(PJSIP/alice&PJSIP/bob,,Q(NO_ANSWER))
- </example>
- <example title="Dial with pre-dial subroutines">
- [default]
- exten => callee_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
- same => n,Log(NOTICE, I'm called on channel ${CHANNEL} prior to it starting the dial attempt)
- same => n,Return()
- exten => called_channel,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
- same => n,Log(NOTICE, I'm called on outbound channel ${CHANNEL} prior to it being used to dial someone)
- same => n,Return()
- exten => _X.,1,NoOp()
- same => n,Dial(PJSIP/alice,,b(default^called_channel^1(my_gosub_arg1^my_gosub_arg2))B(default^callee_channel^1(my_gosub_arg1^my_gosub_arg2)))
- same => n,Hangup()
- </example>
- <example title="Dial with post-answer subroutine executed on outbound channel">
- [my_gosub_routine]
- exten => s,1,NoOp(ARG1=${ARG1} ARG2=${ARG2})
- same => n,Playback(hello)
- same => n,Return()
- [default]
- exten => _X.,1,NoOp()
- same => n,Dial(PJSIP/alice,,U(my_gosub_routine^my_gosub_arg1^my_gosub_arg2))
- same => n,Hangup()
- </example>
- <example title="Dial into ConfBridge using 'G' option">
- same => n,Dial(PJSIP/alice,,G(jump_to_here))
- same => n(jump_to_here),Goto(confbridge)
- same => n,Goto(confbridge)
- same => n(confbridge),ConfBridge(${EXTEN})
- </example>
- <para>This application sets the following channel variables:</para>
- <variablelist>
- <variable name="DIALEDTIME">
- <para>This is the time from dialing a channel until when it is disconnected.</para>
- </variable>
- <variable name="ANSWEREDTIME">
- <para>This is the amount of time for actual call.</para>
- </variable>
- <variable name="DIALEDPEERNAME">
- <para>The name of the outbound channel that answered the call.</para>
- </variable>
- <variable name="DIALEDPEERNUMBER">
- <para>The number that was dialed for the answered outbound channel.</para>
- </variable>
- <variable name="FORWARDERNAME">
- <para>If a call forward occurred, the name of the forwarded channel.</para>
- </variable>
- <variable name="DIALSTATUS">
- <para>This is the status of the call</para>
- <value name="CHANUNAVAIL" />
- <value name="CONGESTION" />
- <value name="NOANSWER" />
- <value name="BUSY" />
- <value name="ANSWER" />
- <value name="CANCEL" />
- <value name="DONTCALL">
- For the Privacy and Screening Modes.
- Will be set if the called party chooses to send the calling party to the 'Go Away' script.
- </value>
- <value name="TORTURE">
- For the Privacy and Screening Modes.
- Will be set if the called party chooses to send the calling party to the 'torture' script.
- </value>
- <value name="INVALIDARGS" />
- </variable>
- </variablelist>
- </description>
- <see-also>
- <ref type="application">RetryDial</ref>
- <ref type="application">SendDTMF</ref>
- <ref type="application">Gosub</ref>
- <ref type="application">Macro</ref>
- </see-also>
- </application>
- <application name="RetryDial" language="en_US">
- <synopsis>
- Place a call, retrying on failure allowing an optional exit extension.
- </synopsis>
- <syntax>
- <parameter name="announce" required="true">
- <para>Filename of sound that will be played when no channel can be reached</para>
- </parameter>
- <parameter name="sleep" required="true">
- <para>Number of seconds to wait after a dial attempt failed before a new attempt is made</para>
- </parameter>
- <parameter name="retries" required="true">
- <para>Number of retries</para>
- <para>When this is reached flow will continue at the next priority in the dialplan</para>
- </parameter>
- <parameter name="dialargs" required="true">
- <para>Same format as arguments provided to the Dial application</para>
- </parameter>
- </syntax>
- <description>
- <para>This application will attempt to place a call using the normal Dial application.
- If no channel can be reached, the <replaceable>announce</replaceable> file will be played.
- Then, it will wait <replaceable>sleep</replaceable> number of seconds before retrying the call.
- After <replaceable>retries</replaceable> number of attempts, the calling channel will continue at the next priority in the dialplan.
- If the <replaceable>retries</replaceable> setting is set to 0, this application will retry endlessly.
- While waiting to retry a call, a 1 digit extension may be dialed. If that
- extension exists in either the context defined in <variable>EXITCONTEXT</variable> or the current
- one, The call will jump to that extension immediately.
- The <replaceable>dialargs</replaceable> are specified in the same format that arguments are provided
- to the Dial application.</para>
- </description>
- <see-also>
- <ref type="application">Dial</ref>
- </see-also>
- </application>
- ***/
- static const char app[] = "Dial";
- static const char rapp[] = "RetryDial";
- enum {
- OPT_ANNOUNCE = (1 << 0),
- OPT_RESETCDR = (1 << 1),
- OPT_DTMF_EXIT = (1 << 2),
- OPT_SENDDTMF = (1 << 3),
- OPT_FORCECLID = (1 << 4),
- OPT_GO_ON = (1 << 5),
- OPT_CALLEE_HANGUP = (1 << 6),
- OPT_CALLER_HANGUP = (1 << 7),
- OPT_ORIGINAL_CLID = (1 << 8),
- OPT_DURATION_LIMIT = (1 << 9),
- OPT_MUSICBACK = (1 << 10),
- OPT_CALLEE_MACRO = (1 << 11),
- OPT_SCREEN_NOINTRO = (1 << 12),
- OPT_SCREEN_NOCALLERID = (1 << 13),
- OPT_IGNORE_CONNECTEDLINE = (1 << 14),
- OPT_SCREENING = (1 << 15),
- OPT_PRIVACY = (1 << 16),
- OPT_RINGBACK = (1 << 17),
- OPT_DURATION_STOP = (1 << 18),
- OPT_CALLEE_TRANSFER = (1 << 19),
- OPT_CALLER_TRANSFER = (1 << 20),
- OPT_CALLEE_MONITOR = (1 << 21),
- OPT_CALLER_MONITOR = (1 << 22),
- OPT_GOTO = (1 << 23),
- OPT_OPERMODE = (1 << 24),
- OPT_CALLEE_PARK = (1 << 25),
- OPT_CALLER_PARK = (1 << 26),
- OPT_IGNORE_FORWARDING = (1 << 27),
- OPT_CALLEE_GOSUB = (1 << 28),
- OPT_CALLEE_MIXMONITOR = (1 << 29),
- OPT_CALLER_MIXMONITOR = (1 << 30),
- };
- /* flags are now 64 bits, so keep it up! */
- #define DIAL_STILLGOING (1LLU << 31)
- #define DIAL_NOFORWARDHTML (1LLU << 32)
- #define DIAL_CALLERID_ABSENT (1LLU << 33) /* TRUE if caller id is not available for connected line. */
- #define OPT_CANCEL_ELSEWHERE (1LLU << 34)
- #define OPT_PEER_H (1LLU << 35)
- #define OPT_CALLEE_GO_ON (1LLU << 36)
- #define OPT_CANCEL_TIMEOUT (1LLU << 37)
- #define OPT_FORCE_CID_TAG (1LLU << 38)
- #define OPT_FORCE_CID_PRES (1LLU << 39)
- #define OPT_CALLER_ANSWER (1LLU << 40)
- #define OPT_PREDIAL_CALLEE (1LLU << 41)
- #define OPT_PREDIAL_CALLER (1LLU << 42)
- #define OPT_RING_WITH_EARLY_MEDIA (1LLU << 43)
- #define OPT_HANGUPCAUSE (1LLU << 44)
- enum {
- OPT_ARG_ANNOUNCE = 0,
- OPT_ARG_SENDDTMF,
- OPT_ARG_GOTO,
- OPT_ARG_DURATION_LIMIT,
- OPT_ARG_MUSICBACK,
- OPT_ARG_CALLEE_MACRO,
- OPT_ARG_RINGBACK,
- OPT_ARG_CALLEE_GOSUB,
- OPT_ARG_CALLEE_GO_ON,
- OPT_ARG_PRIVACY,
- OPT_ARG_DURATION_STOP,
- OPT_ARG_OPERMODE,
- OPT_ARG_SCREEN_NOINTRO,
- OPT_ARG_ORIGINAL_CLID,
- OPT_ARG_FORCECLID,
- OPT_ARG_FORCE_CID_TAG,
- OPT_ARG_FORCE_CID_PRES,
- OPT_ARG_PREDIAL_CALLEE,
- OPT_ARG_PREDIAL_CALLER,
- OPT_ARG_HANGUPCAUSE,
- /* note: this entry _MUST_ be the last one in the enum */
- OPT_ARG_ARRAY_SIZE
- };
- AST_APP_OPTIONS(dial_exec_options, BEGIN_OPTIONS
- AST_APP_OPTION_ARG('A', OPT_ANNOUNCE, OPT_ARG_ANNOUNCE),
- AST_APP_OPTION('a', OPT_CALLER_ANSWER),
- AST_APP_OPTION_ARG('b', OPT_PREDIAL_CALLEE, OPT_ARG_PREDIAL_CALLEE),
- AST_APP_OPTION_ARG('B', OPT_PREDIAL_CALLER, OPT_ARG_PREDIAL_CALLER),
- AST_APP_OPTION('C', OPT_RESETCDR),
- AST_APP_OPTION('c', OPT_CANCEL_ELSEWHERE),
- AST_APP_OPTION('d', OPT_DTMF_EXIT),
- AST_APP_OPTION_ARG('D', OPT_SENDDTMF, OPT_ARG_SENDDTMF),
- AST_APP_OPTION('e', OPT_PEER_H),
- AST_APP_OPTION_ARG('f', OPT_FORCECLID, OPT_ARG_FORCECLID),
- AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
- AST_APP_OPTION('g', OPT_GO_ON),
- AST_APP_OPTION_ARG('G', OPT_GOTO, OPT_ARG_GOTO),
- AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
- AST_APP_OPTION('H', OPT_CALLER_HANGUP),
- AST_APP_OPTION('i', OPT_IGNORE_FORWARDING),
- AST_APP_OPTION('I', OPT_IGNORE_CONNECTEDLINE),
- AST_APP_OPTION('k', OPT_CALLEE_PARK),
- AST_APP_OPTION('K', OPT_CALLER_PARK),
- AST_APP_OPTION_ARG('L', OPT_DURATION_LIMIT, OPT_ARG_DURATION_LIMIT),
- AST_APP_OPTION_ARG('m', OPT_MUSICBACK, OPT_ARG_MUSICBACK),
- AST_APP_OPTION_ARG('M', OPT_CALLEE_MACRO, OPT_ARG_CALLEE_MACRO),
- AST_APP_OPTION_ARG('n', OPT_SCREEN_NOINTRO, OPT_ARG_SCREEN_NOINTRO),
- AST_APP_OPTION('N', OPT_SCREEN_NOCALLERID),
- AST_APP_OPTION_ARG('o', OPT_ORIGINAL_CLID, OPT_ARG_ORIGINAL_CLID),
- AST_APP_OPTION_ARG('O', OPT_OPERMODE, OPT_ARG_OPERMODE),
- AST_APP_OPTION('p', OPT_SCREENING),
- AST_APP_OPTION_ARG('P', OPT_PRIVACY, OPT_ARG_PRIVACY),
- AST_APP_OPTION_ARG('Q', OPT_HANGUPCAUSE, OPT_ARG_HANGUPCAUSE),
- AST_APP_OPTION_ARG('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
- AST_APP_OPTION('R', OPT_RING_WITH_EARLY_MEDIA),
- AST_APP_OPTION_ARG('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
- AST_APP_OPTION_ARG('s', OPT_FORCE_CID_TAG, OPT_ARG_FORCE_CID_TAG),
- AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
- AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
- AST_APP_OPTION_ARG('u', OPT_FORCE_CID_PRES, OPT_ARG_FORCE_CID_PRES),
- AST_APP_OPTION_ARG('U', OPT_CALLEE_GOSUB, OPT_ARG_CALLEE_GOSUB),
- AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
- AST_APP_OPTION('W', OPT_CALLER_MONITOR),
- AST_APP_OPTION('x', OPT_CALLEE_MIXMONITOR),
- AST_APP_OPTION('X', OPT_CALLER_MIXMONITOR),
- AST_APP_OPTION('z', OPT_CANCEL_TIMEOUT),
- END_OPTIONS );
- #define CAN_EARLY_BRIDGE(flags,chan,peer) (!ast_test_flag64(flags, OPT_CALLEE_HANGUP | \
- OPT_CALLER_HANGUP | OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER | \
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR | OPT_CALLEE_PARK | \
- OPT_CALLER_PARK | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB) && \
- !ast_channel_audiohooks(chan) && !ast_channel_audiohooks(peer) && \
- ast_framehook_list_is_empty(ast_channel_framehooks(chan)) && ast_framehook_list_is_empty(ast_channel_framehooks(peer)))
- /*
- * The list of active channels
- */
- struct chanlist {
- AST_LIST_ENTRY(chanlist) node;
- struct ast_channel *chan;
- /*! Channel interface dialing string (is tech/number). (Stored in stuff[]) */
- const char *interface;
- /*! Channel technology name. (Stored in stuff[]) */
- const char *tech;
- /*! Channel device addressing. (Stored in stuff[]) */
- const char *number;
- /*! Original channel name. Must be freed. Could be NULL if allocation failed. */
- char *orig_chan_name;
- uint64_t flags;
- /*! Saved connected party info from an AST_CONTROL_CONNECTED_LINE. */
- struct ast_party_connected_line connected;
- /*! TRUE if an AST_CONTROL_CONNECTED_LINE update was saved to the connected element. */
- unsigned int pending_connected_update:1;
- struct ast_aoc_decoded *aoc_s_rate_list;
- /*! The interface, tech, and number strings are stuffed here. */
- char stuff[0];
- };
- AST_LIST_HEAD_NOLOCK(dial_head, chanlist);
- static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode);
- static void chanlist_free(struct chanlist *outgoing)
- {
- ast_party_connected_line_free(&outgoing->connected);
- ast_aoc_destroy_decoded(outgoing->aoc_s_rate_list);
- ast_free(outgoing->orig_chan_name);
- ast_free(outgoing);
- }
- static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int hangupcause)
- {
- /* Hang up a tree of stuff */
- struct chanlist *outgoing;
- while ((outgoing = AST_LIST_REMOVE_HEAD(out_chans, node))) {
- /* Hangup any existing lines we have open */
- if (outgoing->chan && (outgoing->chan != exception)) {
- if (hangupcause >= 0) {
- /* This is for the channel drivers */
- ast_channel_hangupcause_set(outgoing->chan, hangupcause);
- }
- ast_hangup(outgoing->chan);
- }
- chanlist_free(outgoing);
- }
- }
- #define AST_MAX_WATCHERS 256
- /*
- * argument to handle_cause() and other functions.
- */
- struct cause_args {
- struct ast_channel *chan;
- int busy;
- int congestion;
- int nochan;
- };
- static void handle_cause(int cause, struct cause_args *num)
- {
- switch(cause) {
- case AST_CAUSE_BUSY:
- num->busy++;
- break;
- case AST_CAUSE_CONGESTION:
- num->congestion++;
- break;
- case AST_CAUSE_NO_ROUTE_DESTINATION:
- case AST_CAUSE_UNREGISTERED:
- num->nochan++;
- break;
- case AST_CAUSE_NO_ANSWER:
- case AST_CAUSE_NORMAL_CLEARING:
- break;
- default:
- num->nochan++;
- break;
- }
- }
- static int onedigit_goto(struct ast_channel *chan, const char *context, char exten, int pri)
- {
- char rexten[2] = { exten, '\0' };
- if (context) {
- if (!ast_goto_if_exists(chan, context, rexten, pri))
- return 1;
- } else {
- if (!ast_goto_if_exists(chan, ast_channel_context(chan), rexten, pri))
- return 1;
- else if (!ast_strlen_zero(ast_channel_macrocontext(chan))) {
- if (!ast_goto_if_exists(chan, ast_channel_macrocontext(chan), rexten, pri))
- return 1;
- }
- }
- return 0;
- }
- /* do not call with chan lock held */
- static const char *get_cid_name(char *name, int namelen, struct ast_channel *chan)
- {
- const char *context;
- const char *exten;
- ast_channel_lock(chan);
- context = ast_strdupa(S_OR(ast_channel_macrocontext(chan), ast_channel_context(chan)));
- exten = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
- ast_channel_unlock(chan);
- return ast_get_hint(NULL, 0, name, namelen, chan, context, exten) ? name : "";
- }
- /*!
- * helper function for wait_for_answer()
- *
- * \param o Outgoing call channel list.
- * \param num Incoming call channel cause accumulation
- * \param peerflags Dial option flags
- * \param single TRUE if there is only one outgoing call.
- * \param caller_entertained TRUE if the caller is being entertained by MOH or ringback.
- * \param to Remaining call timeout time.
- * \param forced_clid OPT_FORCECLID caller id to send
- * \param stored_clid Caller id representing the called party if needed
- *
- * XXX this code is highly suspicious, as it essentially overwrites
- * the outgoing channel without properly deleting it.
- *
- * \todo eventually this function should be intergrated into and replaced by ast_call_forward()
- */
- static void do_forward(struct chanlist *o, struct cause_args *num,
- struct ast_flags64 *peerflags, int single, int caller_entertained, int *to,
- struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
- {
- char tmpchan[256];
- char forwarder[AST_CHANNEL_NAME];
- struct ast_channel *original = o->chan;
- struct ast_channel *c = o->chan; /* the winner */
- struct ast_channel *in = num->chan; /* the input channel */
- char *stuff;
- char *tech;
- int cause;
- struct ast_party_caller caller;
- ast_copy_string(forwarder, ast_channel_name(c), sizeof(forwarder));
- ast_copy_string(tmpchan, ast_channel_call_forward(c), sizeof(tmpchan));
- if ((stuff = strchr(tmpchan, '/'))) {
- *stuff++ = '\0';
- tech = tmpchan;
- } else {
- const char *forward_context;
- ast_channel_lock(c);
- forward_context = pbx_builtin_getvar_helper(c, "FORWARD_CONTEXT");
- if (ast_strlen_zero(forward_context)) {
- forward_context = NULL;
- }
- snprintf(tmpchan, sizeof(tmpchan), "%s@%s", ast_channel_call_forward(c), forward_context ? forward_context : ast_channel_context(c));
- ast_channel_unlock(c);
- stuff = tmpchan;
- tech = "Local";
- }
- if (!strcasecmp(tech, "Local")) {
- /*
- * Drop the connected line update block for local channels since
- * this is going to run dialplan and the user can change his
- * mind about what connected line information he wants to send.
- */
- ast_clear_flag64(o, OPT_IGNORE_CONNECTEDLINE);
- }
- /* Before processing channel, go ahead and check for forwarding */
- ast_verb(3, "Now forwarding %s to '%s/%s' (thanks to %s)\n", ast_channel_name(in), tech, stuff, ast_channel_name(c));
- /* If we have been told to ignore forwards, just set this channel to null and continue processing extensions normally */
- if (ast_test_flag64(peerflags, OPT_IGNORE_FORWARDING)) {
- ast_verb(3, "Forwarding %s to '%s/%s' prevented.\n", ast_channel_name(in), tech, stuff);
- c = o->chan = NULL;
- cause = AST_CAUSE_BUSY;
- } else {
- struct ast_format_cap *nativeformats;
- ast_channel_lock(in);
- nativeformats = ao2_bump(ast_channel_nativeformats(in));
- ast_channel_unlock(in);
- /* Setup parameters */
- c = o->chan = ast_request(tech, nativeformats, NULL, in, stuff, &cause);
- ao2_cleanup(nativeformats);
- if (c) {
- if (single && !caller_entertained) {
- ast_channel_make_compatible(in, o->chan);
- }
- ast_channel_lock_both(in, o->chan);
- ast_channel_inherit_variables(in, o->chan);
- ast_channel_datastore_inherit(in, o->chan);
- pbx_builtin_setvar_helper(o->chan, "FORWARDERNAME", forwarder);
- ast_max_forwards_decrement(o->chan);
- ast_channel_unlock(in);
- ast_channel_unlock(o->chan);
- /* When a call is forwarded, we don't want to track new interfaces
- * dialed for CC purposes. Setting the done flag will ensure that
- * any Dial operations that happen later won't record CC interfaces.
- */
- ast_ignore_cc(o->chan);
- ast_log(LOG_NOTICE, "Not accepting call completion offers from call-forward recipient %s\n", ast_channel_name(o->chan));
- } else
- ast_log(LOG_NOTICE,
- "Forwarding failed to create channel to dial '%s/%s' (cause = %d)\n",
- tech, stuff, cause);
- }
- if (!c) {
- ast_channel_publish_dial(in, original, stuff, "BUSY");
- ast_clear_flag64(o, DIAL_STILLGOING);
- handle_cause(cause, num);
- ast_hangup(original);
- } else {
- ast_channel_lock_both(c, original);
- ast_party_redirecting_copy(ast_channel_redirecting(c),
- ast_channel_redirecting(original));
- ast_channel_unlock(c);
- ast_channel_unlock(original);
- ast_channel_lock_both(c, in);
- if (single && !caller_entertained && CAN_EARLY_BRIDGE(peerflags, c, in)) {
- ast_rtp_instance_early_bridge_make_compatible(c, in);
- }
- if (!ast_channel_redirecting(c)->from.number.valid
- || ast_strlen_zero(ast_channel_redirecting(c)->from.number.str)) {
- /*
- * The call was not previously redirected so it is
- * now redirected from this number.
- */
- ast_party_number_free(&ast_channel_redirecting(c)->from.number);
- ast_party_number_init(&ast_channel_redirecting(c)->from.number);
- ast_channel_redirecting(c)->from.number.valid = 1;
- ast_channel_redirecting(c)->from.number.str =
- ast_strdup(S_OR(ast_channel_macroexten(in), ast_channel_exten(in)));
- }
- ast_channel_dialed(c)->transit_network_select = ast_channel_dialed(in)->transit_network_select;
- /* Determine CallerID to store in outgoing channel. */
- ast_party_caller_set_init(&caller, ast_channel_caller(c));
- if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
- caller.id = *stored_clid;
- ast_channel_set_caller_event(c, &caller, NULL);
- ast_set_flag64(o, DIAL_CALLERID_ABSENT);
- } else if (ast_strlen_zero(S_COR(ast_channel_caller(c)->id.number.valid,
- ast_channel_caller(c)->id.number.str, NULL))) {
- /*
- * The new channel has no preset CallerID number by the channel
- * driver. Use the dialplan extension and hint name.
- */
- caller.id = *stored_clid;
- ast_channel_set_caller_event(c, &caller, NULL);
- ast_set_flag64(o, DIAL_CALLERID_ABSENT);
- } else {
- ast_clear_flag64(o, DIAL_CALLERID_ABSENT);
- }
- /* Determine CallerID for outgoing channel to send. */
- if (ast_test_flag64(o, OPT_FORCECLID)) {
- struct ast_party_connected_line connected;
- ast_party_connected_line_init(&connected);
- connected.id = *forced_clid;
- ast_party_connected_line_copy(ast_channel_connected(c), &connected);
- } else {
- ast_connected_line_copy_from_caller(ast_channel_connected(c), ast_channel_caller(in));
- }
- ast_channel_req_accountcodes(c, in, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
- ast_channel_appl_set(c, "AppDial");
- ast_channel_data_set(c, "(Outgoing Line)");
- ast_channel_publish_snapshot(c);
- ast_channel_unlock(in);
- if (single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- struct ast_party_redirecting redirecting;
- /*
- * Redirecting updates to the caller make sense only on single
- * calls.
- *
- * We must unlock c before calling
- * ast_channel_redirecting_macro, because we put c into
- * autoservice there. That is pretty much a guaranteed
- * deadlock. This is why the handling of c's lock may seem a
- * bit unusual here.
- */
- ast_party_redirecting_init(&redirecting);
- ast_party_redirecting_copy(&redirecting, ast_channel_redirecting(c));
- ast_channel_unlock(c);
- if (ast_channel_redirecting_sub(c, in, &redirecting, 0) &&
- ast_channel_redirecting_macro(c, in, &redirecting, 1, 0)) {
- ast_channel_update_redirecting(in, &redirecting, NULL);
- }
- ast_party_redirecting_free(&redirecting);
- } else {
- ast_channel_unlock(c);
- }
- if (ast_test_flag64(peerflags, OPT_CANCEL_TIMEOUT)) {
- *to = -1;
- }
- if (ast_call(c, stuff, 0)) {
- ast_log(LOG_NOTICE, "Forwarding failed to dial '%s/%s'\n",
- tech, stuff);
- ast_channel_publish_dial(in, original, stuff, "CONGESTION");
- ast_clear_flag64(o, DIAL_STILLGOING);
- ast_hangup(original);
- ast_hangup(c);
- c = o->chan = NULL;
- num->nochan++;
- } else {
- ast_channel_publish_dial_forward(in, original, c, NULL, "CANCEL",
- ast_channel_call_forward(original));
- ast_channel_publish_dial(in, c, stuff, NULL);
- /* Hangup the original channel now, in case we needed it */
- ast_hangup(original);
- }
- if (single && !caller_entertained) {
- ast_indicate(in, -1);
- }
- }
- }
- /* argument used for some functions. */
- struct privacy_args {
- int sentringing;
- int privdb_val;
- char privcid[256];
- char privintro[1024];
- char status[256];
- };
- static void publish_dial_end_event(struct ast_channel *in, struct dial_head *out_chans, struct ast_channel *exception, const char *status)
- {
- struct chanlist *outgoing;
- AST_LIST_TRAVERSE(out_chans, outgoing, node) {
- if (!outgoing->chan || outgoing->chan == exception) {
- continue;
- }
- ast_channel_publish_dial(in, outgoing->chan, NULL, status);
- }
- }
- /*!
- * \internal
- * \brief Update connected line on chan from peer.
- * \since 13.6.0
- *
- * \param chan Channel to get connected line updated.
- * \param peer Channel providing connected line information.
- * \param is_caller Non-zero if chan is the calling channel.
- *
- * \return Nothing
- */
- static void update_connected_line_from_peer(struct ast_channel *chan, struct ast_channel *peer, int is_caller)
- {
- struct ast_party_connected_line connected_caller;
- ast_party_connected_line_init(&connected_caller);
- ast_channel_lock(peer);
- ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(peer));
- ast_channel_unlock(peer);
- connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
- if (ast_channel_connected_line_sub(peer, chan, &connected_caller, 0)
- && ast_channel_connected_line_macro(peer, chan, &connected_caller, is_caller, 0)) {
- ast_channel_update_connected_line(chan, &connected_caller, NULL);
- }
- ast_party_connected_line_free(&connected_caller);
- }
- static struct ast_channel *wait_for_answer(struct ast_channel *in,
- struct dial_head *out_chans, int *to, struct ast_flags64 *peerflags,
- char *opt_args[],
- struct privacy_args *pa,
- const struct cause_args *num_in, int *result, char *dtmf_progress,
- const int ignore_cc,
- struct ast_party_id *forced_clid, struct ast_party_id *stored_clid)
- {
- struct cause_args num = *num_in;
- int prestart = num.busy + num.congestion + num.nochan;
- int orig = *to;
- struct ast_channel *peer = NULL;
- #ifdef HAVE_EPOLL
- struct chanlist *epollo;
- #endif
- struct chanlist *outgoing = AST_LIST_FIRST(out_chans);
- /* single is set if only one destination is enabled */
- int single = outgoing && !AST_LIST_NEXT(outgoing, node);
- int caller_entertained = outgoing
- && ast_test_flag64(outgoing, OPT_MUSICBACK | OPT_RINGBACK);
- struct ast_str *featurecode = ast_str_alloca(AST_FEATURE_MAX_LEN + 1);
- int cc_recall_core_id;
- int is_cc_recall;
- int cc_frame_received = 0;
- int num_ringing = 0;
- struct timeval start = ast_tvnow();
- if (single) {
- /* Turn off hold music, etc */
- if (!caller_entertained) {
- ast_deactivate_generator(in);
- /* If we are calling a single channel, and not providing ringback or music, */
- /* then, make them compatible for in-band tone purpose */
- if (ast_channel_make_compatible(in, outgoing->chan) < 0) {
- /* If these channels can not be made compatible,
- * there is no point in continuing. The bridge
- * will just fail if it gets that far.
- */
- *to = -1;
- strcpy(pa->status, "CONGESTION");
- ast_channel_publish_dial(in, outgoing->chan, NULL, pa->status);
- return NULL;
- }
- }
- if (!ast_test_flag64(outgoing, OPT_IGNORE_CONNECTEDLINE)
- && !ast_test_flag64(outgoing, DIAL_CALLERID_ABSENT)) {
- update_connected_line_from_peer(in, outgoing->chan, 1);
- }
- }
- is_cc_recall = ast_cc_is_recall(in, &cc_recall_core_id, NULL);
- #ifdef HAVE_EPOLL
- AST_LIST_TRAVERSE(out_chans, epollo, node) {
- ast_poll_channel_add(in, epollo->chan);
- }
- #endif
- while ((*to = ast_remaining_ms(start, orig)) && !peer) {
- struct chanlist *o;
- int pos = 0; /* how many channels do we handle */
- int numlines = prestart;
- struct ast_channel *winner;
- struct ast_channel *watchers[AST_MAX_WATCHERS];
- watchers[pos++] = in;
- AST_LIST_TRAVERSE(out_chans, o, node) {
- /* Keep track of important channels */
- if (ast_test_flag64(o, DIAL_STILLGOING) && o->chan)
- watchers[pos++] = o->chan;
- numlines++;
- }
- if (pos == 1) { /* only the input channel is available */
- if (numlines == (num.busy + num.congestion + num.nochan)) {
- ast_verb(2, "Everyone is busy/congested at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
- if (num.busy)
- strcpy(pa->status, "BUSY");
- else if (num.congestion)
- strcpy(pa->status, "CONGESTION");
- else if (num.nochan)
- strcpy(pa->status, "CHANUNAVAIL");
- } else {
- ast_verb(3, "No one is available to answer at this time (%d:%d/%d/%d)\n", numlines, num.busy, num.congestion, num.nochan);
- }
- *to = 0;
- if (is_cc_recall) {
- ast_cc_failed(cc_recall_core_id, "Everyone is busy/congested for the recall. How sad");
- }
- return NULL;
- }
- winner = ast_waitfor_n(watchers, pos, to);
- AST_LIST_TRAVERSE(out_chans, o, node) {
- struct ast_frame *f;
- struct ast_channel *c = o->chan;
- if (c == NULL)
- continue;
- if (ast_test_flag64(o, DIAL_STILLGOING) && ast_channel_state(c) == AST_STATE_UP) {
- if (!peer) {
- ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
- if (o->orig_chan_name
- && strcmp(o->orig_chan_name, ast_channel_name(c))) {
- /*
- * The channel name changed so we must generate COLP update.
- * Likely because a call pickup channel masqueraded in.
- */
- update_connected_line_from_peer(in, c, 1);
- } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- if (o->pending_connected_update) {
- if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
- ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
- ast_channel_update_connected_line(in, &o->connected, NULL);
- }
- } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
- update_connected_line_from_peer(in, c, 1);
- }
- }
- if (o->aoc_s_rate_list) {
- size_t encoded_size;
- struct ast_aoc_encoded *encoded;
- if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
- ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
- ast_aoc_destroy_encoded(encoded);
- }
- }
- peer = c;
- publish_dial_end_event(in, out_chans, peer, "CANCEL");
- ast_copy_flags64(peerflags, o,
- OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
- OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
- OPT_CALLEE_PARK | OPT_CALLER_PARK |
- OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
- DIAL_NOFORWARDHTML);
- ast_channel_dialcontext_set(c, "");
- ast_channel_exten_set(c, "");
- }
- continue;
- }
- if (c != winner)
- continue;
- /* here, o->chan == c == winner */
- if (!ast_strlen_zero(ast_channel_call_forward(c))) {
- pa->sentringing = 0;
- if (!ignore_cc && (f = ast_read(c))) {
- if (f->frametype == AST_FRAME_CONTROL && f->subclass.integer == AST_CONTROL_CC) {
- /* This channel is forwarding the call, and is capable of CC, so
- * be sure to add the new device interface to the list
- */
- ast_handle_cc_control_frame(in, c, f->data.ptr);
- }
- ast_frfree(f);
- }
- if (o->pending_connected_update) {
- /*
- * Re-seed the chanlist's connected line information with
- * previously acquired connected line info from the incoming
- * channel. The previously acquired connected line info could
- * have been set through the CONNECTED_LINE dialplan function.
- */
- o->pending_connected_update = 0;
- ast_channel_lock(in);
- ast_party_connected_line_copy(&o->connected, ast_channel_connected(in));
- ast_channel_unlock(in);
- }
- do_forward(o, &num, peerflags, single, caller_entertained, &orig,
- forced_clid, stored_clid);
- if (o->chan) {
- ast_free(o->orig_chan_name);
- o->orig_chan_name = ast_strdup(ast_channel_name(o->chan));
- if (single
- && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
- && !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
- update_connected_line_from_peer(in, o->chan, 1);
- }
- }
- continue;
- }
- f = ast_read(winner);
- if (!f) {
- ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
- #ifdef HAVE_EPOLL
- ast_poll_channel_del(in, c);
- #endif
- ast_channel_publish_dial(in, c, NULL, ast_hangup_cause_to_dial_status(ast_channel_hangupcause(c)));
- ast_hangup(c);
- c = o->chan = NULL;
- ast_clear_flag64(o, DIAL_STILLGOING);
- handle_cause(ast_channel_hangupcause(in), &num);
- continue;
- }
- switch (f->frametype) {
- case AST_FRAME_CONTROL:
- switch (f->subclass.integer) {
- case AST_CONTROL_ANSWER:
- /* This is our guy if someone answered. */
- if (!peer) {
- ast_verb(3, "%s answered %s\n", ast_channel_name(c), ast_channel_name(in));
- if (o->orig_chan_name
- && strcmp(o->orig_chan_name, ast_channel_name(c))) {
- /*
- * The channel name changed so we must generate COLP update.
- * Likely because a call pickup channel masqueraded in.
- */
- update_connected_line_from_peer(in, c, 1);
- } else if (!single && !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- if (o->pending_connected_update) {
- if (ast_channel_connected_line_sub(c, in, &o->connected, 0) &&
- ast_channel_connected_line_macro(c, in, &o->connected, 1, 0)) {
- ast_channel_update_connected_line(in, &o->connected, NULL);
- }
- } else if (!ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
- update_connected_line_from_peer(in, c, 1);
- }
- }
- if (o->aoc_s_rate_list) {
- size_t encoded_size;
- struct ast_aoc_encoded *encoded;
- if ((encoded = ast_aoc_encode(o->aoc_s_rate_list, &encoded_size, o->chan))) {
- ast_indicate_data(in, AST_CONTROL_AOC, encoded, encoded_size);
- ast_aoc_destroy_encoded(encoded);
- }
- }
- peer = c;
- /* Inform everyone else that they've been canceled.
- * The dial end event for the peer will be sent out after
- * other Dial options have been handled.
- */
- publish_dial_end_event(in, out_chans, peer, "CANCEL");
- ast_copy_flags64(peerflags, o,
- OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
- OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
- OPT_CALLEE_PARK | OPT_CALLER_PARK |
- OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
- DIAL_NOFORWARDHTML);
- ast_channel_dialcontext_set(c, "");
- ast_channel_exten_set(c, "");
- if (CAN_EARLY_BRIDGE(peerflags, in, peer)) {
- /* Setup early bridge if appropriate */
- ast_channel_early_bridge(in, peer);
- }
- }
- /* If call has been answered, then the eventual hangup is likely to be normal hangup */
- ast_channel_hangupcause_set(in, AST_CAUSE_NORMAL_CLEARING);
- ast_channel_hangupcause_set(c, AST_CAUSE_NORMAL_CLEARING);
- break;
- case AST_CONTROL_BUSY:
- ast_verb(3, "%s is busy\n", ast_channel_name(c));
- ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
- ast_channel_publish_dial(in, c, NULL, "BUSY");
- ast_hangup(c);
- c = o->chan = NULL;
- ast_clear_flag64(o, DIAL_STILLGOING);
- handle_cause(AST_CAUSE_BUSY, &num);
- break;
- case AST_CONTROL_CONGESTION:
- ast_verb(3, "%s is circuit-busy\n", ast_channel_name(c));
- ast_channel_hangupcause_set(in, ast_channel_hangupcause(c));
- ast_channel_publish_dial(in, c, NULL, "CONGESTION");
- ast_hangup(c);
- c = o->chan = NULL;
- ast_clear_flag64(o, DIAL_STILLGOING);
- handle_cause(AST_CAUSE_CONGESTION, &num);
- break;
- case AST_CONTROL_RINGING:
- /* This is a tricky area to get right when using a native
- * CC agent. The reason is that we do the best we can to send only a
- * single ringing notification to the caller.
- *
- * Call completion complicates the logic used here. CCNR is typically
- * offered during a ringing message. Let's say that party A calls
- * parties B, C, and D. B and C do not support CC requests, but D
- * does. If we were to receive a ringing notification from B before
- * the others, then we would end up sending a ringing message to
- * A with no CCNR offer present.
- *
- * The approach that we have taken is that if we receive a ringing
- * response from a party and no CCNR offer is present, we need to
- * wait. Specifically, we need to wait until either a) a called party
- * offers CCNR in its ringing response or b) all called parties have
- * responded in some way to our call and none offers CCNR.
- *
- * The drawback to this is that if one of the parties has a delayed
- * response or, god forbid, one just plain doesn't respond to our
- * outgoing call, then this will result in a significant delay between
- * when the caller places the call and hears ringback.
- *
- * Note also that if CC is disabled for this call, then it is perfectly
- * fine for ringing frames to get sent through.
- */
- ++num_ringing;
- if (ignore_cc || cc_frame_received || num_ringing == numlines) {
- ast_verb(3, "%s is ringing\n", ast_channel_name(c));
- /* Setup early media if appropriate */
- if (single && !caller_entertained
- && CAN_EARLY_BRIDGE(peerflags, in, c)) {
- ast_channel_early_bridge(in, c);
- }
- if (!(pa->sentringing) && !ast_test_flag64(outgoing, OPT_MUSICBACK) && ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
- ast_indicate(in, AST_CONTROL_RINGING);
- pa->sentringing++;
- }
- }
- break;
- case AST_CONTROL_PROGRESS:
- ast_verb(3, "%s is making progress passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
- /* Setup early media if appropriate */
- if (single && !caller_entertained
- && CAN_EARLY_BRIDGE(peerflags, in, c)) {
- ast_channel_early_bridge(in, c);
- }
- if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
- if (single || (!single && !pa->sentringing)) {
- ast_indicate(in, AST_CONTROL_PROGRESS);
- }
- }
- if (!ast_strlen_zero(dtmf_progress)) {
- ast_verb(3,
- "Sending DTMF '%s' to the called party as result of receiving a PROGRESS message.\n",
- dtmf_progress);
- ast_dtmf_stream(c, in, dtmf_progress, 250, 0);
- }
- break;
- case AST_CONTROL_VIDUPDATE:
- case AST_CONTROL_SRCUPDATE:
- case AST_CONTROL_SRCCHANGE:
- if (!single || caller_entertained) {
- break;
- }
- ast_verb(3, "%s requested media update control %d, passing it to %s\n",
- ast_channel_name(c), f->subclass.integer, ast_channel_name(in));
- ast_indicate(in, f->subclass.integer);
- break;
- case AST_CONTROL_CONNECTED_LINE:
- if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(in));
- break;
- }
- if (!single) {
- struct ast_party_connected_line connected;
- ast_verb(3, "%s connected line has changed. Saving it until answer for %s\n",
- ast_channel_name(c), ast_channel_name(in));
- ast_party_connected_line_set_init(&connected, &o->connected);
- ast_connected_line_parse_data(f->data.ptr, f->datalen, &connected);
- ast_party_connected_line_set(&o->connected, &connected, NULL);
- ast_party_connected_line_free(&connected);
- o->pending_connected_update = 1;
- break;
- }
- if (ast_channel_connected_line_sub(c, in, f, 1) &&
- ast_channel_connected_line_macro(c, in, f, 1, 1)) {
- ast_indicate_data(in, AST_CONTROL_CONNECTED_LINE, f->data.ptr, f->datalen);
- }
- break;
- case AST_CONTROL_AOC:
- {
- struct ast_aoc_decoded *decoded = ast_aoc_decode(f->data.ptr, f->datalen, o->chan);
- if (decoded && (ast_aoc_get_msg_type(decoded) == AST_AOC_S)) {
- ast_aoc_destroy_decoded(o->aoc_s_rate_list);
- o->aoc_s_rate_list = decoded;
- } else {
- ast_aoc_destroy_decoded(decoded);
- }
- }
- break;
- case AST_CONTROL_REDIRECTING:
- if (!single) {
- /*
- * Redirecting updates to the caller make sense only on single
- * calls.
- */
- break;
- }
- if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(in));
- break;
- }
- ast_verb(3, "%s redirecting info has changed, passing it to %s\n",
- ast_channel_name(c), ast_channel_name(in));
- if (ast_channel_redirecting_sub(c, in, f, 1) &&
- ast_channel_redirecting_macro(c, in, f, 1, 1)) {
- ast_indicate_data(in, AST_CONTROL_REDIRECTING, f->data.ptr, f->datalen);
- }
- pa->sentringing = 0;
- break;
- case AST_CONTROL_PROCEEDING:
- ast_verb(3, "%s is proceeding passing it to %s\n", ast_channel_name(c), ast_channel_name(in));
- if (single && !caller_entertained
- && CAN_EARLY_BRIDGE(peerflags, in, c)) {
- ast_channel_early_bridge(in, c);
- }
- if (!ast_test_flag64(outgoing, OPT_RINGBACK))
- ast_indicate(in, AST_CONTROL_PROCEEDING);
- break;
- case AST_CONTROL_HOLD:
- /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
- ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(c));
- ast_indicate_data(in, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
- break;
- case AST_CONTROL_UNHOLD:
- /* XXX this should be saved like AST_CONTROL_CONNECTED_LINE for !single || caller_entertained */
- ast_verb(3, "Call on %s left from hold\n", ast_channel_name(c));
- ast_indicate(in, AST_CONTROL_UNHOLD);
- break;
- case AST_CONTROL_OFFHOOK:
- case AST_CONTROL_FLASH:
- /* Ignore going off hook and flash */
- break;
- case AST_CONTROL_CC:
- if (!ignore_cc) {
- ast_handle_cc_control_frame(in, c, f->data.ptr);
- cc_frame_received = 1;
- }
- break;
- case AST_CONTROL_PVT_CAUSE_CODE:
- ast_indicate_data(in, AST_CONTROL_PVT_CAUSE_CODE, f->data.ptr, f->datalen);
- break;
- case -1:
- if (single && !caller_entertained) {
- ast_verb(3, "%s stopped sounds\n", ast_channel_name(c));
- ast_indicate(in, -1);
- pa->sentringing = 0;
- }
- break;
- default:
- ast_debug(1, "Dunno what to do with control type %d\n", f->subclass.integer);
- break;
- }
- break;
- case AST_FRAME_VIDEO:
- case AST_FRAME_VOICE:
- case AST_FRAME_IMAGE:
- if (caller_entertained) {
- break;
- }
- /* Fall through */
- case AST_FRAME_TEXT:
- if (single && ast_write(in, f)) {
- ast_log(LOG_WARNING, "Unable to write frametype: %u\n",
- f->frametype);
- }
- break;
- case AST_FRAME_HTML:
- if (single && !ast_test_flag64(outgoing, DIAL_NOFORWARDHTML)
- && ast_channel_sendhtml(in, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
- ast_log(LOG_WARNING, "Unable to send URL\n");
- }
- break;
- default:
- break;
- }
- ast_frfree(f);
- } /* end for */
- if (winner == in) {
- struct ast_frame *f = ast_read(in);
- #if 0
- if (f && (f->frametype != AST_FRAME_VOICE))
- printf("Frame type: %d, %d\n", f->frametype, f->subclass);
- else if (!f || (f->frametype != AST_FRAME_VOICE))
- printf("Hangup received on %s\n", in->name);
- #endif
- if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass.integer == AST_CONTROL_HANGUP))) {
- /* Got hung up */
- *to = -1;
- strcpy(pa->status, "CANCEL");
- publish_dial_end_event(in, out_chans, NULL, pa->status);
- if (f) {
- if (f->data.uint32) {
- ast_channel_hangupcause_set(in, f->data.uint32);
- }
- ast_frfree(f);
- }
- if (is_cc_recall) {
- ast_cc_completed(in, "CC completed, although the caller hung up (cancelled)");
- }
- return NULL;
- }
- /* now f is guaranteed non-NULL */
- if (f->frametype == AST_FRAME_DTMF) {
- if (ast_test_flag64(peerflags, OPT_DTMF_EXIT)) {
- const char *context;
- ast_channel_lock(in);
- context = pbx_builtin_getvar_helper(in, "EXITCONTEXT");
- if (onedigit_goto(in, context, (char) f->subclass.integer, 1)) {
- ast_verb(3, "User hit %c to disconnect call.\n", f->subclass.integer);
- *to = 0;
- *result = f->subclass.integer;
- strcpy(pa->status, "CANCEL");
- publish_dial_end_event(in, out_chans, NULL, pa->status);
- ast_frfree(f);
- ast_channel_unlock(in);
- if (is_cc_recall) {
- ast_cc_completed(in, "CC completed, but the caller used DTMF to exit");
- }
- return NULL;
- }
- ast_channel_unlock(in);
- }
- if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP) &&
- detect_disconnect(in, f->subclass.integer, &featurecode)) {
- ast_verb(3, "User requested call disconnect.\n");
- *to = 0;
- strcpy(pa->status, "CANCEL");
- publish_dial_end_event(in, out_chans, NULL, pa->status);
- ast_frfree(f);
- if (is_cc_recall) {
- ast_cc_completed(in, "CC completed, but the caller hung up with DTMF");
- }
- return NULL;
- }
- }
- /* Send the frame from the in channel to all outgoing channels. */
- AST_LIST_TRAVERSE(out_chans, o, node) {
- if (!o->chan || !ast_test_flag64(o, DIAL_STILLGOING)) {
- /* This outgoing channel has died so don't send the frame to it. */
- continue;
- }
- switch (f->frametype) {
- case AST_FRAME_HTML:
- /* Forward HTML stuff */
- if (!ast_test_flag64(o, DIAL_NOFORWARDHTML)
- && ast_channel_sendhtml(o->chan, f->subclass.integer, f->data.ptr, f->datalen) == -1) {
- ast_log(LOG_WARNING, "Unable to send URL\n");
- }
- break;
- case AST_FRAME_VIDEO:
- case AST_FRAME_VOICE:
- case AST_FRAME_IMAGE:
- if (!single || caller_entertained) {
- /*
- * We are calling multiple parties or caller is being
- * entertained and has thus not been made compatible.
- * No need to check any other called parties.
- */
- goto skip_frame;
- }
- /* Fall through */
- case AST_FRAME_TEXT:
- case AST_FRAME_DTMF_BEGIN:
- case AST_FRAME_DTMF_END:
- if (ast_write(o->chan, f)) {
- ast_log(LOG_WARNING, "Unable to forward frametype: %u\n",
- f->frametype);
- }
- break;
- case AST_FRAME_CONTROL:
- switch (f->subclass.integer) {
- case AST_CONTROL_HOLD:
- ast_verb(3, "Call on %s placed on hold\n", ast_channel_name(o->chan));
- ast_indicate_data(o->chan, AST_CONTROL_HOLD, f->data.ptr, f->datalen);
- break;
- case AST_CONTROL_UNHOLD:
- ast_verb(3, "Call on %s left from hold\n", ast_channel_name(o->chan));
- ast_indicate(o->chan, AST_CONTROL_UNHOLD);
- break;
- case AST_CONTROL_VIDUPDATE:
- case AST_CONTROL_SRCUPDATE:
- case AST_CONTROL_SRCCHANGE:
- if (!single || caller_entertained) {
- /*
- * We are calling multiple parties or caller is being
- * entertained and has thus not been made compatible.
- * No need to check any other called parties.
- */
- goto skip_frame;
- }
- ast_verb(3, "%s requested media update control %d, passing it to %s\n",
- ast_channel_name(in), f->subclass.integer, ast_channel_name(o->chan));
- ast_indicate(o->chan, f->subclass.integer);
- break;
- case AST_CONTROL_CONNECTED_LINE:
- if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- ast_verb(3, "Connected line update to %s prevented.\n", ast_channel_name(o->chan));
- break;
- }
- if (ast_channel_connected_line_sub(in, o->chan, f, 1) &&
- ast_channel_connected_line_macro(in, o->chan, f, 0, 1)) {
- ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
- }
- break;
- case AST_CONTROL_REDIRECTING:
- if (ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)) {
- ast_verb(3, "Redirecting update to %s prevented.\n", ast_channel_name(o->chan));
- break;
- }
- if (ast_channel_redirecting_sub(in, o->chan, f, 1) &&
- ast_channel_redirecting_macro(in, o->chan, f, 0, 1)) {
- ast_indicate_data(o->chan, f->subclass.integer, f->data.ptr, f->datalen);
- }
- break;
- default:
- /* We are not going to do anything with this frame. */
- goto skip_frame;
- }
- break;
- default:
- /* We are not going to do anything with this frame. */
- goto skip_frame;
- }
- }
- skip_frame:;
- ast_frfree(f);
- }
- }
- if (!*to || ast_check_hangup(in)) {
- ast_verb(3, "Nobody picked up in %d ms\n", orig);
- publish_dial_end_event(in, out_chans, NULL, "NOANSWER");
- }
- #ifdef HAVE_EPOLL
- AST_LIST_TRAVERSE(out_chans, epollo, node) {
- if (epollo->chan)
- ast_poll_channel_del(in, epollo->chan);
- }
- #endif
- if (is_cc_recall) {
- ast_cc_completed(in, "Recall completed!");
- }
- return peer;
- }
- static int detect_disconnect(struct ast_channel *chan, char code, struct ast_str **featurecode)
- {
- char disconnect_code[AST_FEATURE_MAX_LEN];
- int res;
- ast_str_append(featurecode, 1, "%c", code);
- res = ast_get_builtin_feature(chan, "disconnect", disconnect_code, sizeof(disconnect_code));
- if (res) {
- ast_str_reset(*featurecode);
- return 0;
- }
- if (strlen(disconnect_code) > ast_str_strlen(*featurecode)) {
- /* Could be a partial match, anyway */
- if (strncmp(disconnect_code, ast_str_buffer(*featurecode), ast_str_strlen(*featurecode))) {
- ast_str_reset(*featurecode);
- }
- return 0;
- }
- if (strcmp(disconnect_code, ast_str_buffer(*featurecode))) {
- ast_str_reset(*featurecode);
- return 0;
- }
- return 1;
- }
- /* returns true if there is a valid privacy reply */
- static int valid_priv_reply(struct ast_flags64 *opts, int res)
- {
- if (res < '1')
- return 0;
- if (ast_test_flag64(opts, OPT_PRIVACY) && res <= '5')
- return 1;
- if (ast_test_flag64(opts, OPT_SCREENING) && res <= '4')
- return 1;
- return 0;
- }
- static int do_privacy(struct ast_channel *chan, struct ast_channel *peer,
- struct ast_flags64 *opts, char **opt_args, struct privacy_args *pa)
- {
- int res2;
- int loopcount = 0;
- /* Get the user's intro, store it in priv-callerintros/$CID,
- unless it is already there-- this should be done before the
- call is actually dialed */
- /* all ring indications and moh for the caller has been halted as soon as the
- target extension was picked up. We are going to have to kill some
- time and make the caller believe the peer hasn't picked up yet */
- if (ast_test_flag64(opts, OPT_MUSICBACK) && !ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
- char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
- ast_indicate(chan, -1);
- ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
- ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
- ast_channel_musicclass_set(chan, original_moh);
- } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
- ast_indicate(chan, AST_CONTROL_RINGING);
- pa->sentringing++;
- }
- /* Start autoservice on the other chan ?? */
- res2 = ast_autoservice_start(chan);
- /* Now Stream the File */
- for (loopcount = 0; loopcount < 3; loopcount++) {
- if (res2 && loopcount == 0) /* error in ast_autoservice_start() */
- break;
- if (!res2) /* on timeout, play the message again */
- res2 = ast_play_and_wait(peer, "priv-callpending");
- if (!valid_priv_reply(opts, res2))
- res2 = 0;
- /* priv-callpending script:
- "I have a caller waiting, who introduces themselves as:"
- */
- if (!res2)
- res2 = ast_play_and_wait(peer, pa->privintro);
- if (!valid_priv_reply(opts, res2))
- res2 = 0;
- /* now get input from the called party, as to their choice */
- if (!res2) {
- /* XXX can we have both, or they are mutually exclusive ? */
- if (ast_test_flag64(opts, OPT_PRIVACY))
- res2 = ast_play_and_wait(peer, "priv-callee-options");
- if (ast_test_flag64(opts, OPT_SCREENING))
- res2 = ast_play_and_wait(peer, "screen-callee-options");
- }
- /*! \page DialPrivacy Dial Privacy scripts
- * \par priv-callee-options script:
- * \li Dial 1 if you wish this caller to reach you directly in the future,
- * and immediately connect to their incoming call.
- * \li Dial 2 if you wish to send this caller to voicemail now and forevermore.
- * \li Dial 3 to send this caller to the torture menus, now and forevermore.
- * \li Dial 4 to send this caller to a simple "go away" menu, now and forevermore.
- * \li Dial 5 to allow this caller to come straight thru to you in the future,
- * but right now, just this once, send them to voicemail.
- *
- * \par screen-callee-options script:
- * \li Dial 1 if you wish to immediately connect to the incoming call
- * \li Dial 2 if you wish to send this caller to voicemail.
- * \li Dial 3 to send this caller to the torture menus.
- * \li Dial 4 to send this caller to a simple "go away" menu.
- */
- if (valid_priv_reply(opts, res2))
- break;
- /* invalid option */
- res2 = ast_play_and_wait(peer, "vm-sorry");
- }
- if (ast_test_flag64(opts, OPT_MUSICBACK)) {
- ast_moh_stop(chan);
- } else if (ast_test_flag64(opts, OPT_RINGBACK) || ast_test_flag64(opts, OPT_RING_WITH_EARLY_MEDIA)) {
- ast_indicate(chan, -1);
- pa->sentringing = 0;
- }
- ast_autoservice_stop(chan);
- if (ast_test_flag64(opts, OPT_PRIVACY) && (res2 >= '1' && res2 <= '5')) {
- /* map keypresses to various things, the index is res2 - '1' */
- static const char * const _val[] = { "ALLOW", "DENY", "TORTURE", "KILL", "ALLOW" };
- static const int _flag[] = { AST_PRIVACY_ALLOW, AST_PRIVACY_DENY, AST_PRIVACY_TORTURE, AST_PRIVACY_KILL, AST_PRIVACY_ALLOW};
- int i = res2 - '1';
- ast_verb(3, "--Set privacy database entry %s/%s to %s\n",
- opt_args[OPT_ARG_PRIVACY], pa->privcid, _val[i]);
- ast_privacy_set(opt_args[OPT_ARG_PRIVACY], pa->privcid, _flag[i]);
- }
- switch (res2) {
- case '1':
- break;
- case '2':
- ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
- break;
- case '3':
- ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
- break;
- case '4':
- ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
- break;
- case '5':
- if (ast_test_flag64(opts, OPT_PRIVACY)) {
- ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
- break;
- }
- /* if not privacy, then 5 is the same as "default" case */
- default: /* bad input or -1 if failure to start autoservice */
- /* well, if the user messes up, ... he had his chance... What Is The Best Thing To Do? */
- /* well, there seems basically two choices. Just patch the caller thru immediately,
- or,... put 'em thru to voicemail. */
- /* since the callee may have hung up, let's do the voicemail thing, no database decision */
- ast_log(LOG_NOTICE, "privacy: no valid response from the callee. Sending the caller to voicemail, the callee isn't responding\n");
- /* XXX should we set status to DENY ? */
- /* XXX what about the privacy flags ? */
- break;
- }
- if (res2 == '1') { /* the only case where we actually connect */
- /* if the intro is NOCALLERID, then there's no reason to leave it on disk, it'll
- just clog things up, and it's not useful information, not being tied to a CID */
- if (strncmp(pa->privcid, "NOCALLERID", 10) == 0 || ast_test_flag64(opts, OPT_SCREEN_NOINTRO)) {
- ast_filedelete(pa->privintro, NULL);
- if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
- ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
- else
- ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
- }
- return 0; /* the good exit path */
- } else {
- return -1;
- }
- }
- /*! \brief returns 1 if successful, 0 or <0 if the caller should 'goto out' */
- static int setup_privacy_args(struct privacy_args *pa,
- struct ast_flags64 *opts, char *opt_args[], struct ast_channel *chan)
- {
- char callerid[60];
- int res;
- char *l;
- if (ast_channel_caller(chan)->id.number.valid
- && !ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
- l = ast_strdupa(ast_channel_caller(chan)->id.number.str);
- ast_shrink_phone_number(l);
- if (ast_test_flag64(opts, OPT_PRIVACY) ) {
- ast_verb(3, "Privacy DB is '%s', clid is '%s'\n", opt_args[OPT_ARG_PRIVACY], l);
- pa->privdb_val = ast_privacy_check(opt_args[OPT_ARG_PRIVACY], l);
- } else {
- ast_verb(3, "Privacy Screening, clid is '%s'\n", l);
- pa->privdb_val = AST_PRIVACY_UNKNOWN;
- }
- } else {
- char *tnam, *tn2;
- tnam = ast_strdupa(ast_channel_name(chan));
- /* clean the channel name so slashes don't try to end up in disk file name */
- for (tn2 = tnam; *tn2; tn2++) {
- if (*tn2 == '/') /* any other chars to be afraid of? */
- *tn2 = '=';
- }
- ast_verb(3, "Privacy-- callerid is empty\n");
- snprintf(callerid, sizeof(callerid), "NOCALLERID_%s%s", ast_channel_exten(chan), tnam);
- l = callerid;
- pa->privdb_val = AST_PRIVACY_UNKNOWN;
- }
- ast_copy_string(pa->privcid, l, sizeof(pa->privcid));
- if (strncmp(pa->privcid, "NOCALLERID", 10) != 0 && ast_test_flag64(opts, OPT_SCREEN_NOCALLERID)) {
- /* if callerid is set and OPT_SCREEN_NOCALLERID is set also */
- ast_verb(3, "CallerID set (%s); N option set; Screening should be off\n", pa->privcid);
- pa->privdb_val = AST_PRIVACY_ALLOW;
- } else if (ast_test_flag64(opts, OPT_SCREEN_NOCALLERID) && strncmp(pa->privcid, "NOCALLERID", 10) == 0) {
- ast_verb(3, "CallerID blank; N option set; Screening should happen; dbval is %d\n", pa->privdb_val);
- }
- if (pa->privdb_val == AST_PRIVACY_DENY) {
- ast_verb(3, "Privacy DB reports PRIVACY_DENY for this callerid. Dial reports unavailable\n");
- ast_copy_string(pa->status, "NOANSWER", sizeof(pa->status));
- return 0;
- } else if (pa->privdb_val == AST_PRIVACY_KILL) {
- ast_copy_string(pa->status, "DONTCALL", sizeof(pa->status));
- return 0; /* Is this right? */
- } else if (pa->privdb_val == AST_PRIVACY_TORTURE) {
- ast_copy_string(pa->status, "TORTURE", sizeof(pa->status));
- return 0; /* is this right??? */
- } else if (pa->privdb_val == AST_PRIVACY_UNKNOWN) {
- /* Get the user's intro, store it in priv-callerintros/$CID,
- unless it is already there-- this should be done before the
- call is actually dialed */
- /* make sure the priv-callerintros dir actually exists */
- snprintf(pa->privintro, sizeof(pa->privintro), "%s/sounds/priv-callerintros", ast_config_AST_DATA_DIR);
- if ((res = ast_mkdir(pa->privintro, 0755))) {
- ast_log(LOG_WARNING, "privacy: can't create directory priv-callerintros: %s\n", strerror(res));
- return -1;
- }
- snprintf(pa->privintro, sizeof(pa->privintro), "priv-callerintros/%s", pa->privcid);
- if (ast_fileexists(pa->privintro, NULL, NULL ) > 0 && strncmp(pa->privcid, "NOCALLERID", 10) != 0) {
- /* the DELUX version of this code would allow this caller the
- option to hear and retape their previously recorded intro.
- */
- } else {
- int duration; /* for feedback from play_and_wait */
- /* the file doesn't exist yet. Let the caller submit his
- vocal intro for posterity */
- /* priv-recordintro script:
- "At the tone, please say your name:"
- */
- int silencethreshold = ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE);
- ast_answer(chan);
- res = ast_play_and_record(chan, "priv-recordintro", pa->privintro, 4, "sln", &duration, NULL, silencethreshold, 2000, 0); /* NOTE: I've reduced the total time to 4 sec */
- /* don't think we'll need a lock removed, we took care of
- conflicts by naming the pa.privintro file */
- if (res == -1) {
- /* Delete the file regardless since they hung up during recording */
- ast_filedelete(pa->privintro, NULL);
- if (ast_fileexists(pa->privintro, NULL, NULL) > 0)
- ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa->privintro);
- else
- ast_verb(3, "Successfully deleted %s intro file\n", pa->privintro);
- return -1;
- }
- if (!ast_streamfile(chan, "vm-dialout", ast_channel_language(chan)) )
- ast_waitstream(chan, "");
- }
- }
- return 1; /* success */
- }
- static void end_bridge_callback(void *data)
- {
- char buf[80];
- time_t end;
- struct ast_channel *chan = data;
- time(&end);
- ast_channel_lock(chan);
- ast_channel_stage_snapshot(chan);
- snprintf(buf, sizeof(buf), "%d", ast_channel_get_up_time(chan));
- pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", buf);
- snprintf(buf, sizeof(buf), "%d", ast_channel_get_duration(chan));
- pbx_builtin_setvar_helper(chan, "DIALEDTIME", buf);
- ast_channel_stage_snapshot_done(chan);
- ast_channel_unlock(chan);
- }
- static void end_bridge_callback_data_fixup(struct ast_bridge_config *bconfig, struct ast_channel *originator, struct ast_channel *terminator) {
- bconfig->end_bridge_callback_data = originator;
- }
- static int dial_handle_playtones(struct ast_channel *chan, const char *data)
- {
- struct ast_tone_zone_sound *ts = NULL;
- int res;
- const char *str = data;
- if (ast_strlen_zero(str)) {
- ast_debug(1,"Nothing to play\n");
- return -1;
- }
- ts = ast_get_indication_tone(ast_channel_zone(chan), str);
- if (ts && ts->data[0]) {
- res = ast_playtones_start(chan, 0, ts->data, 0);
- } else {
- res = -1;
- }
- if (ts) {
- ts = ast_tone_zone_sound_unref(ts);
- }
- if (res) {
- ast_log(LOG_WARNING, "Unable to start playtone \'%s\'\n", str);
- }
- return res;
- }
- /*!
- * \internal
- * \brief Setup the after bridge goto location on the peer.
- * \since 12.0.0
- *
- * \param chan Calling channel for bridge.
- * \param peer Peer channel for bridge.
- * \param opts Dialing option flags.
- * \param opt_args Dialing option argument strings.
- *
- * \return Nothing
- */
- static void setup_peer_after_bridge_goto(struct ast_channel *chan, struct ast_channel *peer, struct ast_flags64 *opts, char *opt_args[])
- {
- const char *context;
- const char *extension;
- int priority;
- if (ast_test_flag64(opts, OPT_PEER_H)) {
- ast_channel_lock(chan);
- context = ast_strdupa(ast_channel_context(chan));
- ast_channel_unlock(chan);
- ast_bridge_set_after_h(peer, context);
- } else if (ast_test_flag64(opts, OPT_CALLEE_GO_ON)) {
- ast_channel_lock(chan);
- context = ast_strdupa(ast_channel_context(chan));
- extension = ast_strdupa(ast_channel_exten(chan));
- priority = ast_channel_priority(chan);
- ast_channel_unlock(chan);
- ast_bridge_set_after_go_on(peer, context, extension, priority,
- opt_args[OPT_ARG_CALLEE_GO_ON]);
- }
- }
- static int dial_exec_full(struct ast_channel *chan, const char *data, struct ast_flags64 *peerflags, int *continue_exec)
- {
- int res = -1; /* default: error */
- char *rest, *cur; /* scan the list of destinations */
- struct dial_head out_chans = AST_LIST_HEAD_NOLOCK_INIT_VALUE; /* list of destinations */
- struct chanlist *outgoing;
- struct chanlist *tmp;
- struct ast_channel *peer;
- int to; /* timeout */
- struct cause_args num = { chan, 0, 0, 0 };
- int cause;
- struct ast_bridge_config config = { { 0, } };
- struct timeval calldurationlimit = { 0, };
- char *dtmfcalled = NULL, *dtmfcalling = NULL, *dtmf_progress=NULL;
- struct privacy_args pa = {
- .sentringing = 0,
- .privdb_val = 0,
- .status = "INVALIDARGS",
- };
- int sentringing = 0, moh = 0;
- const char *outbound_group = NULL;
- int result = 0;
- char *parse;
- int opermode = 0;
- int delprivintro = 0;
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(peers);
- AST_APP_ARG(timeout);
- AST_APP_ARG(options);
- AST_APP_ARG(url);
- );
- struct ast_flags64 opts = { 0, };
- char *opt_args[OPT_ARG_ARRAY_SIZE];
- int fulldial = 0, num_dialed = 0;
- int ignore_cc = 0;
- char device_name[AST_CHANNEL_NAME];
- char forced_clid_name[AST_MAX_EXTENSION];
- char stored_clid_name[AST_MAX_EXTENSION];
- int force_forwards_only; /*!< TRUE if force CallerID on call forward only. Legacy behaviour.*/
- /*!
- * \brief Forced CallerID party information to send.
- * \note This will not have any malloced strings so do not free it.
- */
- struct ast_party_id forced_clid;
- /*!
- * \brief Stored CallerID information if needed.
- *
- * \note If OPT_ORIGINAL_CLID set then this is the o option
- * CallerID. Otherwise it is the dialplan extension and hint
- * name.
- *
- * \note This will not have any malloced strings so do not free it.
- */
- struct ast_party_id stored_clid;
- /*!
- * \brief CallerID party information to store.
- * \note This will not have any malloced strings so do not free it.
- */
- struct ast_party_caller caller;
- int max_forwards;
- /* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
- ast_channel_lock(chan);
- ast_channel_stage_snapshot(chan);
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", "");
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", "");
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", "");
- pbx_builtin_setvar_helper(chan, "ANSWEREDTIME", "");
- pbx_builtin_setvar_helper(chan, "DIALEDTIME", "");
- ast_channel_stage_snapshot_done(chan);
- max_forwards = ast_max_forwards_get(chan);
- ast_channel_unlock(chan);
- if (max_forwards <= 0) {
- ast_log(LOG_WARNING, "Cannot place outbound call from channel '%s'. Max forwards exceeded\n",
- ast_channel_name(chan));
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", "BUSY");
- return -1;
- }
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- return -1;
- }
- if (ast_check_hangup_locked(chan)) {
- /*
- * Caller hung up before we could dial. If dial is executed
- * within an AGI then the AGI has likely eaten all queued
- * frames before executing the dial in DeadAGI mode. With
- * the caller hung up and no pending frames from the caller's
- * read queue, dial would not know that the call has hung up
- * until a called channel answers. It is rather annoying to
- * whoever just answered the non-existent call.
- *
- * Dial should not continue execution in DeadAGI mode, hangup
- * handlers, or the h exten.
- */
- ast_verb(3, "Caller hung up before dial.\n");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", "CANCEL");
- return -1;
- }
- parse = ast_strdupa(data);
- AST_STANDARD_APP_ARGS(args, parse);
- if (!ast_strlen_zero(args.options) &&
- ast_app_parse_options64(dial_exec_options, &opts, opt_args, args.options)) {
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- goto done;
- }
- if (ast_strlen_zero(args.peers)) {
- ast_log(LOG_WARNING, "Dial requires an argument (technology/resource)\n");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- goto done;
- }
- if (ast_cc_call_init(chan, &ignore_cc)) {
- goto done;
- }
- if (ast_test_flag64(&opts, OPT_SCREEN_NOINTRO) && !ast_strlen_zero(opt_args[OPT_ARG_SCREEN_NOINTRO])) {
- delprivintro = atoi(opt_args[OPT_ARG_SCREEN_NOINTRO]);
- if (delprivintro < 0 || delprivintro > 1) {
- ast_log(LOG_WARNING, "Unknown argument %d specified to n option, ignoring\n", delprivintro);
- delprivintro = 0;
- }
- }
- if (!ast_test_flag64(&opts, OPT_RINGBACK)) {
- opt_args[OPT_ARG_RINGBACK] = NULL;
- }
- if (ast_test_flag64(&opts, OPT_OPERMODE)) {
- opermode = ast_strlen_zero(opt_args[OPT_ARG_OPERMODE]) ? 1 : atoi(opt_args[OPT_ARG_OPERMODE]);
- ast_verb(3, "Setting operator services mode to %d.\n", opermode);
- }
- if (ast_test_flag64(&opts, OPT_DURATION_STOP) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_STOP])) {
- calldurationlimit.tv_sec = atoi(opt_args[OPT_ARG_DURATION_STOP]);
- if (!calldurationlimit.tv_sec) {
- ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- goto done;
- }
- ast_verb(3, "Setting call duration limit to %.3lf seconds.\n", calldurationlimit.tv_sec + calldurationlimit.tv_usec / 1000000.0);
- }
- if (ast_test_flag64(&opts, OPT_SENDDTMF) && !ast_strlen_zero(opt_args[OPT_ARG_SENDDTMF])) {
- dtmf_progress = opt_args[OPT_ARG_SENDDTMF];
- dtmfcalled = strsep(&dtmf_progress, ":");
- dtmfcalling = strsep(&dtmf_progress, ":");
- }
- if (ast_test_flag64(&opts, OPT_DURATION_LIMIT) && !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])) {
- if (ast_bridge_timelimit(chan, &config, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit))
- goto done;
- }
- /* Setup the forced CallerID information to send if used. */
- ast_party_id_init(&forced_clid);
- force_forwards_only = 0;
- if (ast_test_flag64(&opts, OPT_FORCECLID)) {
- if (ast_strlen_zero(opt_args[OPT_ARG_FORCECLID])) {
- ast_channel_lock(chan);
- forced_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
- ast_channel_unlock(chan);
- forced_clid_name[0] = '\0';
- forced_clid.name.str = (char *) get_cid_name(forced_clid_name,
- sizeof(forced_clid_name), chan);
- force_forwards_only = 1;
- } else {
- /* Note: The opt_args[OPT_ARG_FORCECLID] string value is altered here. */
- ast_callerid_parse(opt_args[OPT_ARG_FORCECLID], &forced_clid.name.str,
- &forced_clid.number.str);
- }
- if (!ast_strlen_zero(forced_clid.name.str)) {
- forced_clid.name.valid = 1;
- }
- if (!ast_strlen_zero(forced_clid.number.str)) {
- forced_clid.number.valid = 1;
- }
- }
- if (ast_test_flag64(&opts, OPT_FORCE_CID_TAG)
- && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_TAG])) {
- forced_clid.tag = opt_args[OPT_ARG_FORCE_CID_TAG];
- }
- forced_clid.number.presentation = AST_PRES_ALLOWED_USER_NUMBER_PASSED_SCREEN;
- if (ast_test_flag64(&opts, OPT_FORCE_CID_PRES)
- && !ast_strlen_zero(opt_args[OPT_ARG_FORCE_CID_PRES])) {
- int pres;
- pres = ast_parse_caller_presentation(opt_args[OPT_ARG_FORCE_CID_PRES]);
- if (0 <= pres) {
- forced_clid.number.presentation = pres;
- }
- }
- /* Setup the stored CallerID information if needed. */
- ast_party_id_init(&stored_clid);
- if (ast_test_flag64(&opts, OPT_ORIGINAL_CLID)) {
- if (ast_strlen_zero(opt_args[OPT_ARG_ORIGINAL_CLID])) {
- ast_channel_lock(chan);
- ast_party_id_set_init(&stored_clid, &ast_channel_caller(chan)->id);
- if (!ast_strlen_zero(ast_channel_caller(chan)->id.name.str)) {
- stored_clid.name.str = ast_strdupa(ast_channel_caller(chan)->id.name.str);
- }
- if (!ast_strlen_zero(ast_channel_caller(chan)->id.number.str)) {
- stored_clid.number.str = ast_strdupa(ast_channel_caller(chan)->id.number.str);
- }
- if (!ast_strlen_zero(ast_channel_caller(chan)->id.subaddress.str)) {
- stored_clid.subaddress.str = ast_strdupa(ast_channel_caller(chan)->id.subaddress.str);
- }
- if (!ast_strlen_zero(ast_channel_caller(chan)->id.tag)) {
- stored_clid.tag = ast_strdupa(ast_channel_caller(chan)->id.tag);
- }
- ast_channel_unlock(chan);
- } else {
- /* Note: The opt_args[OPT_ARG_ORIGINAL_CLID] string value is altered here. */
- ast_callerid_parse(opt_args[OPT_ARG_ORIGINAL_CLID], &stored_clid.name.str,
- &stored_clid.number.str);
- if (!ast_strlen_zero(stored_clid.name.str)) {
- stored_clid.name.valid = 1;
- }
- if (!ast_strlen_zero(stored_clid.number.str)) {
- stored_clid.number.valid = 1;
- }
- }
- } else {
- /*
- * In case the new channel has no preset CallerID number by the
- * channel driver, setup the dialplan extension and hint name.
- */
- stored_clid_name[0] = '\0';
- stored_clid.name.str = (char *) get_cid_name(stored_clid_name,
- sizeof(stored_clid_name), chan);
- if (ast_strlen_zero(stored_clid.name.str)) {
- stored_clid.name.str = NULL;
- } else {
- stored_clid.name.valid = 1;
- }
- ast_channel_lock(chan);
- stored_clid.number.str = ast_strdupa(S_OR(ast_channel_macroexten(chan), ast_channel_exten(chan)));
- stored_clid.number.valid = 1;
- ast_channel_unlock(chan);
- }
- if (ast_test_flag64(&opts, OPT_RESETCDR)) {
- ast_cdr_reset(ast_channel_name(chan), 0);
- }
- if (ast_test_flag64(&opts, OPT_PRIVACY) && ast_strlen_zero(opt_args[OPT_ARG_PRIVACY]))
- opt_args[OPT_ARG_PRIVACY] = ast_strdupa(ast_channel_exten(chan));
- if (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) {
- res = setup_privacy_args(&pa, &opts, opt_args, chan);
- if (res <= 0)
- goto out;
- res = -1; /* reset default */
- }
- if (continue_exec)
- *continue_exec = 0;
- /* If a channel group has been specified, get it for use when we create peer channels */
- ast_channel_lock(chan);
- if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP_ONCE"))) {
- outbound_group = ast_strdupa(outbound_group);
- pbx_builtin_setvar_helper(chan, "OUTBOUND_GROUP_ONCE", NULL);
- } else if ((outbound_group = pbx_builtin_getvar_helper(chan, "OUTBOUND_GROUP"))) {
- outbound_group = ast_strdupa(outbound_group);
- }
- ast_channel_unlock(chan);
- /* Set per dial instance flags. These flags are also passed back to RetryDial. */
- ast_copy_flags64(peerflags, &opts, OPT_DTMF_EXIT | OPT_GO_ON | OPT_ORIGINAL_CLID
- | OPT_CALLER_HANGUP | OPT_IGNORE_FORWARDING | OPT_CANCEL_TIMEOUT
- | OPT_ANNOUNCE | OPT_CALLEE_MACRO | OPT_CALLEE_GOSUB | OPT_FORCECLID);
- /* PREDIAL: Run gosub on the caller's channel */
- if (ast_test_flag64(&opts, OPT_PREDIAL_CALLER)
- && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLER])) {
- ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLER]);
- ast_app_exec_sub(NULL, chan, opt_args[OPT_ARG_PREDIAL_CALLER], 0);
- }
- /* loop through the list of dial destinations */
- rest = args.peers;
- while ((cur = strsep(&rest, "&")) ) {
- struct ast_channel *tc; /* channel for this destination */
- /* Get a technology/resource pair */
- char *number = cur;
- char *tech = strsep(&number, "/");
- size_t tech_len;
- size_t number_len;
- struct ast_format_cap *nativeformats;
- num_dialed++;
- if (ast_strlen_zero(number)) {
- ast_log(LOG_WARNING, "Dial argument takes format (technology/resource)\n");
- goto out;
- }
- tech_len = strlen(tech) + 1;
- number_len = strlen(number) + 1;
- tmp = ast_calloc(1, sizeof(*tmp) + (2 * tech_len) + number_len);
- if (!tmp) {
- goto out;
- }
- /* Save tech, number, and interface. */
- cur = tmp->stuff;
- strcpy(cur, tech);
- tmp->tech = cur;
- cur += tech_len;
- strcpy(cur, tech);
- cur[tech_len - 1] = '/';
- tmp->interface = cur;
- cur += tech_len;
- strcpy(cur, number);
- tmp->number = cur;
- if (opts.flags) {
- /* Set per outgoing call leg options. */
- ast_copy_flags64(tmp, &opts,
- OPT_CANCEL_ELSEWHERE |
- OPT_CALLEE_TRANSFER | OPT_CALLER_TRANSFER |
- OPT_CALLEE_HANGUP | OPT_CALLER_HANGUP |
- OPT_CALLEE_MONITOR | OPT_CALLER_MONITOR |
- OPT_CALLEE_PARK | OPT_CALLER_PARK |
- OPT_CALLEE_MIXMONITOR | OPT_CALLER_MIXMONITOR |
- OPT_RINGBACK | OPT_MUSICBACK | OPT_FORCECLID | OPT_IGNORE_CONNECTEDLINE |
- OPT_RING_WITH_EARLY_MEDIA);
- ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
- }
- /* Request the peer */
- ast_channel_lock(chan);
- /*
- * Seed the chanlist's connected line information with previously
- * acquired connected line info from the incoming channel. The
- * previously acquired connected line info could have been set
- * through the CONNECTED_LINE dialplan function.
- */
- ast_party_connected_line_copy(&tmp->connected, ast_channel_connected(chan));
- nativeformats = ao2_bump(ast_channel_nativeformats(chan));
- ast_channel_unlock(chan);
- tc = ast_request(tmp->tech, nativeformats, NULL, chan, tmp->number, &cause);
- ao2_cleanup(nativeformats);
- if (!tc) {
- /* If we can't, just go on to the next call */
- ast_log(LOG_WARNING, "Unable to create channel of type '%s' (cause %d - %s)\n",
- tmp->tech, cause, ast_cause2str(cause));
- handle_cause(cause, &num);
- if (!rest) {
- /* we are on the last destination */
- ast_channel_hangupcause_set(chan, cause);
- }
- if (!ignore_cc && (cause == AST_CAUSE_BUSY || cause == AST_CAUSE_CONGESTION)) {
- if (!ast_cc_callback(chan, tmp->tech, tmp->number, ast_cc_busy_interface)) {
- ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, "");
- }
- }
- chanlist_free(tmp);
- continue;
- }
- ast_channel_get_device_name(tc, device_name, sizeof(device_name));
- if (!ignore_cc) {
- ast_cc_extension_monitor_add_dialstring(chan, tmp->interface, device_name);
- }
- ast_channel_lock_both(tc, chan);
- ast_channel_stage_snapshot(tc);
- pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
- /* Setup outgoing SDP to match incoming one */
- if (!AST_LIST_FIRST(&out_chans) && !rest && CAN_EARLY_BRIDGE(peerflags, chan, tc)) {
- /* We are on the only destination. */
- ast_rtp_instance_early_bridge_make_compatible(tc, chan);
- }
- /* Inherit specially named variables from parent channel */
- ast_channel_inherit_variables(chan, tc);
- ast_channel_datastore_inherit(chan, tc);
- ast_max_forwards_decrement(tc);
- ast_channel_appl_set(tc, "AppDial");
- ast_channel_data_set(tc, "(Outgoing Line)");
- memset(ast_channel_whentohangup(tc), 0, sizeof(*ast_channel_whentohangup(tc)));
- /* Determine CallerID to store in outgoing channel. */
- ast_party_caller_set_init(&caller, ast_channel_caller(tc));
- if (ast_test_flag64(peerflags, OPT_ORIGINAL_CLID)) {
- caller.id = stored_clid;
- ast_channel_set_caller_event(tc, &caller, NULL);
- ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
- } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.number.valid,
- ast_channel_caller(tc)->id.number.str, NULL))) {
- /*
- * The new channel has no preset CallerID number by the channel
- * driver. Use the dialplan extension and hint name.
- */
- caller.id = stored_clid;
- if (!caller.id.name.valid
- && !ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
- ast_channel_connected(chan)->id.name.str, NULL))) {
- /*
- * No hint name available. We have a connected name supplied by
- * the dialplan we can use instead.
- */
- caller.id.name.valid = 1;
- caller.id.name = ast_channel_connected(chan)->id.name;
- }
- ast_channel_set_caller_event(tc, &caller, NULL);
- ast_set_flag64(tmp, DIAL_CALLERID_ABSENT);
- } else if (ast_strlen_zero(S_COR(ast_channel_caller(tc)->id.name.valid, ast_channel_caller(tc)->id.name.str,
- NULL))) {
- /* The new channel has no preset CallerID name by the channel driver. */
- if (!ast_strlen_zero(S_COR(ast_channel_connected(chan)->id.name.valid,
- ast_channel_connected(chan)->id.name.str, NULL))) {
- /*
- * We have a connected name supplied by the dialplan we can
- * use instead.
- */
- caller.id.name.valid = 1;
- caller.id.name = ast_channel_connected(chan)->id.name;
- ast_channel_set_caller_event(tc, &caller, NULL);
- }
- }
- /* Determine CallerID for outgoing channel to send. */
- if (ast_test_flag64(peerflags, OPT_FORCECLID) && !force_forwards_only) {
- struct ast_party_connected_line connected;
- ast_party_connected_line_set_init(&connected, ast_channel_connected(tc));
- connected.id = forced_clid;
- ast_channel_set_connected_line(tc, &connected, NULL);
- } else {
- ast_connected_line_copy_from_caller(ast_channel_connected(tc), ast_channel_caller(chan));
- }
- ast_party_redirecting_copy(ast_channel_redirecting(tc), ast_channel_redirecting(chan));
- ast_channel_dialed(tc)->transit_network_select = ast_channel_dialed(chan)->transit_network_select;
- ast_channel_req_accountcodes(tc, chan, AST_CHANNEL_REQUESTOR_BRIDGE_PEER);
- if (ast_strlen_zero(ast_channel_musicclass(tc))) {
- ast_channel_musicclass_set(tc, ast_channel_musicclass(chan));
- }
- /* Pass ADSI CPE and transfer capability */
- ast_channel_adsicpe_set(tc, ast_channel_adsicpe(chan));
- ast_channel_transfercapability_set(tc, ast_channel_transfercapability(chan));
- /* If we have an outbound group, set this peer channel to it */
- if (outbound_group)
- ast_app_group_set_channel(tc, outbound_group);
- /* If the calling channel has the ANSWERED_ELSEWHERE flag set, inherit it. This is to support local channels */
- if (ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE)
- ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
- /* Check if we're forced by configuration */
- if (ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE))
- ast_channel_hangupcause_set(tc, AST_CAUSE_ANSWERED_ELSEWHERE);
- /* Inherit context and extension */
- ast_channel_dialcontext_set(tc, ast_strlen_zero(ast_channel_macrocontext(chan)) ? ast_channel_context(chan) : ast_channel_macrocontext(chan));
- if (!ast_strlen_zero(ast_channel_macroexten(chan)))
- ast_channel_exten_set(tc, ast_channel_macroexten(chan));
- else
- ast_channel_exten_set(tc, ast_channel_exten(chan));
- ast_channel_stage_snapshot_done(tc);
- /* Save the original channel name to detect call pickup masquerading in. */
- tmp->orig_chan_name = ast_strdup(ast_channel_name(tc));
- ast_channel_unlock(tc);
- ast_channel_unlock(chan);
- /* Put channel in the list of outgoing thingies. */
- tmp->chan = tc;
- AST_LIST_INSERT_TAIL(&out_chans, tmp, node);
- }
- /*
- * PREDIAL: Run gosub on all of the callee channels
- *
- * We run the callee predial before ast_call() in case the user
- * wishes to do something on the newly created channels before
- * the channel does anything important.
- *
- * Inside the target gosub we will be able to do something with
- * the newly created channel name ie: now the calling channel
- * can know what channel will be used to call the destination
- * ex: now we will know that SIP/abc-123 is calling SIP/def-124
- */
- if (ast_test_flag64(&opts, OPT_PREDIAL_CALLEE)
- && !ast_strlen_zero(opt_args[OPT_ARG_PREDIAL_CALLEE])
- && !AST_LIST_EMPTY(&out_chans)) {
- const char *predial_callee;
- ast_replace_subargument_delimiter(opt_args[OPT_ARG_PREDIAL_CALLEE]);
- predial_callee = ast_app_expand_sub_args(chan, opt_args[OPT_ARG_PREDIAL_CALLEE]);
- if (predial_callee) {
- ast_autoservice_start(chan);
- AST_LIST_TRAVERSE(&out_chans, tmp, node) {
- ast_pre_call(tmp->chan, predial_callee);
- }
- ast_autoservice_stop(chan);
- ast_free((char *) predial_callee);
- }
- }
- /* Start all outgoing calls */
- AST_LIST_TRAVERSE_SAFE_BEGIN(&out_chans, tmp, node) {
- res = ast_call(tmp->chan, tmp->number, 0); /* Place the call, but don't wait on the answer */
- ast_channel_lock(chan);
- /* check the results of ast_call */
- if (res) {
- /* Again, keep going even if there's an error */
- ast_debug(1, "ast call on peer returned %d\n", res);
- ast_verb(3, "Couldn't call %s\n", tmp->interface);
- if (ast_channel_hangupcause(tmp->chan)) {
- ast_channel_hangupcause_set(chan, ast_channel_hangupcause(tmp->chan));
- }
- ast_channel_unlock(chan);
- ast_cc_call_failed(chan, tmp->chan, tmp->interface);
- ast_hangup(tmp->chan);
- tmp->chan = NULL;
- AST_LIST_REMOVE_CURRENT(node);
- chanlist_free(tmp);
- continue;
- }
- ast_channel_publish_dial(chan, tmp->chan, tmp->number, NULL);
- ast_channel_unlock(chan);
- ast_verb(3, "Called %s\n", tmp->interface);
- ast_set_flag64(tmp, DIAL_STILLGOING);
- /* If this line is up, don't try anybody else */
- if (ast_channel_state(tmp->chan) == AST_STATE_UP) {
- break;
- }
- }
- AST_LIST_TRAVERSE_SAFE_END;
- if (ast_strlen_zero(args.timeout)) {
- to = -1;
- } else {
- to = atoi(args.timeout);
- if (to > 0)
- to *= 1000;
- else {
- ast_log(LOG_WARNING, "Invalid timeout specified: '%s'. Setting timeout to infinite\n", args.timeout);
- to = -1;
- }
- }
- outgoing = AST_LIST_FIRST(&out_chans);
- if (!outgoing) {
- strcpy(pa.status, "CHANUNAVAIL");
- if (fulldial == num_dialed) {
- res = -1;
- goto out;
- }
- } else {
- /* Our status will at least be NOANSWER */
- strcpy(pa.status, "NOANSWER");
- if (ast_test_flag64(outgoing, OPT_MUSICBACK)) {
- moh = 1;
- if (!ast_strlen_zero(opt_args[OPT_ARG_MUSICBACK])) {
- char *original_moh = ast_strdupa(ast_channel_musicclass(chan));
- ast_channel_musicclass_set(chan, opt_args[OPT_ARG_MUSICBACK]);
- ast_moh_start(chan, opt_args[OPT_ARG_MUSICBACK], NULL);
- ast_channel_musicclass_set(chan, original_moh);
- } else {
- ast_moh_start(chan, NULL, NULL);
- }
- ast_indicate(chan, AST_CONTROL_PROGRESS);
- } else if (ast_test_flag64(outgoing, OPT_RINGBACK) || ast_test_flag64(outgoing, OPT_RING_WITH_EARLY_MEDIA)) {
- if (!ast_strlen_zero(opt_args[OPT_ARG_RINGBACK])) {
- if (dial_handle_playtones(chan, opt_args[OPT_ARG_RINGBACK])){
- ast_indicate(chan, AST_CONTROL_RINGING);
- sentringing++;
- } else {
- ast_indicate(chan, AST_CONTROL_PROGRESS);
- }
- } else {
- ast_indicate(chan, AST_CONTROL_RINGING);
- sentringing++;
- }
- }
- }
- peer = wait_for_answer(chan, &out_chans, &to, peerflags, opt_args, &pa, &num, &result,
- dtmf_progress, ignore_cc, &forced_clid, &stored_clid);
- if (!peer) {
- if (result) {
- res = result;
- } else if (to) { /* Musta gotten hung up */
- res = -1;
- } else { /* Nobody answered, next please? */
- res = 0;
- }
- } else {
- const char *number;
- const char *name;
- int dial_end_raised = 0;
- int cause = -1;
- if (ast_test_flag64(&opts, OPT_CALLER_ANSWER)) {
- ast_answer(chan);
- }
- /* Ah ha! Someone answered within the desired timeframe. Of course after this
- we will always return with -1 so that it is hung up properly after the
- conversation. */
- if (ast_test_flag64(&opts, OPT_HANGUPCAUSE)
- && !ast_strlen_zero(opt_args[OPT_ARG_HANGUPCAUSE])) {
- cause = ast_str2cause(opt_args[OPT_ARG_HANGUPCAUSE]);
- if (cause <= 0) {
- if (!strcasecmp(opt_args[OPT_ARG_HANGUPCAUSE], "NONE")) {
- cause = 0;
- } else if (sscanf(opt_args[OPT_ARG_HANGUPCAUSE], "%30d", &cause) != 1
- || cause < 0) {
- ast_log(LOG_WARNING, "Invalid cause given to Dial(...Q(<cause>)): \"%s\"\n",
- opt_args[OPT_ARG_HANGUPCAUSE]);
- cause = -1;
- }
- }
- }
- hanguptree(&out_chans, peer, cause >= 0 ? cause : AST_CAUSE_ANSWERED_ELSEWHERE);
- /* If appropriate, log that we have a destination channel and set the answer time */
- ast_channel_lock(peer);
- name = ast_strdupa(ast_channel_name(peer));
- number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
- if (ast_strlen_zero(number)) {
- number = NULL;
- } else {
- number = ast_strdupa(number);
- }
- ast_channel_unlock(peer);
- ast_channel_lock(chan);
- ast_channel_stage_snapshot(chan);
- strcpy(pa.status, "ANSWER");
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", name);
- pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
- ast_channel_stage_snapshot_done(chan);
- ast_channel_unlock(chan);
- if (!ast_strlen_zero(args.url) && ast_channel_supports_html(peer) ) {
- ast_debug(1, "app_dial: sendurl=%s.\n", args.url);
- ast_channel_sendurl( peer, args.url );
- }
- if ( (ast_test_flag64(&opts, OPT_PRIVACY) || ast_test_flag64(&opts, OPT_SCREENING)) && pa.privdb_val == AST_PRIVACY_UNKNOWN) {
- if (do_privacy(chan, peer, &opts, opt_args, &pa)) {
- ast_channel_publish_dial(chan, peer, NULL, pa.status);
- /* hang up on the callee -- he didn't want to talk anyway! */
- ast_autoservice_chan_hangup_peer(chan, peer);
- res = 0;
- goto out;
- }
- }
- if (!ast_test_flag64(&opts, OPT_ANNOUNCE) || ast_strlen_zero(opt_args[OPT_ARG_ANNOUNCE])) {
- res = 0;
- } else {
- int digit = 0;
- struct ast_channel *chans[2];
- struct ast_channel *active_chan;
- chans[0] = chan;
- chans[1] = peer;
- /* we need to stream the announcement to the called party when the OPT_ARG_ANNOUNCE (-A) is setted */
- /* stream the file */
- res = ast_streamfile(peer, opt_args[OPT_ARG_ANNOUNCE], ast_channel_language(peer));
- if (res) {
- res = 0;
- ast_log(LOG_ERROR, "error streaming file '%s' to callee\n", opt_args[OPT_ARG_ANNOUNCE]);
- }
- ast_channel_set_flag(peer, AST_FLAG_END_DTMF_ONLY);
- while (ast_channel_stream(peer)) {
- int ms;
- ms = ast_sched_wait(ast_channel_sched(peer));
- if (ms < 0 && !ast_channel_timingfunc(peer)) {
- ast_stopstream(peer);
- break;
- }
- if (ms < 0)
- ms = 1000;
- active_chan = ast_waitfor_n(chans, 2, &ms);
- if (active_chan) {
- struct ast_channel *other_chan;
- struct ast_frame *fr = ast_read(active_chan);
- if (!fr) {
- ast_autoservice_chan_hangup_peer(chan, peer);
- res = -1;
- goto done;
- }
- switch (fr->frametype) {
- case AST_FRAME_DTMF_END:
- digit = fr->subclass.integer;
- if (active_chan == peer && strchr(AST_DIGIT_ANY, res)) {
- ast_stopstream(peer);
- res = ast_senddigit(chan, digit, 0);
- }
- break;
- case AST_FRAME_CONTROL:
- switch (fr->subclass.integer) {
- case AST_CONTROL_HANGUP:
- ast_frfree(fr);
- ast_autoservice_chan_hangup_peer(chan, peer);
- res = -1;
- goto done;
- case AST_CONTROL_CONNECTED_LINE:
- /* Pass COLP update to the other channel. */
- if (active_chan == chan) {
- other_chan = peer;
- } else {
- other_chan = chan;
- }
- if (ast_channel_connected_line_sub(active_chan, other_chan, fr, 1)
- && ast_channel_connected_line_macro(active_chan,
- other_chan, fr, other_chan == chan, 1)) {
- ast_indicate_data(other_chan, fr->subclass.integer,
- fr->data.ptr, fr->datalen);
- }
- break;
- default:
- break;
- }
- break;
- default:
- /* Ignore all others */
- break;
- }
- ast_frfree(fr);
- }
- ast_sched_runq(ast_channel_sched(peer));
- }
- ast_channel_clear_flag(peer, AST_FLAG_END_DTMF_ONLY);
- }
- if (chan && peer && ast_test_flag64(&opts, OPT_GOTO) && !ast_strlen_zero(opt_args[OPT_ARG_GOTO])) {
- /* chan and peer are going into the PBX; as such neither are considered
- * outgoing channels any longer */
- ast_channel_clear_flag(chan, AST_FLAG_OUTGOING);
- ast_replace_subargument_delimiter(opt_args[OPT_ARG_GOTO]);
- ast_parseable_goto(chan, opt_args[OPT_ARG_GOTO]);
- /* peer goes to the same context and extension as chan, so just copy info from chan*/
- ast_channel_lock(peer);
- ast_channel_stage_snapshot(peer);
- ast_clear_flag(ast_channel_flags(peer), AST_FLAG_OUTGOING);
- ast_channel_context_set(peer, ast_channel_context(chan));
- ast_channel_exten_set(peer, ast_channel_exten(chan));
- ast_channel_priority_set(peer, ast_channel_priority(chan) + 2);
- ast_channel_stage_snapshot_done(peer);
- ast_channel_unlock(peer);
- if (ast_pbx_start(peer)) {
- ast_autoservice_chan_hangup_peer(chan, peer);
- }
- if (continue_exec)
- *continue_exec = 1;
- res = 0;
- ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
- goto done;
- }
- if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
- const char *macro_result_peer;
- int macro_res;
- /* Set peer->exten and peer->context so that MACRO_EXTEN and MACRO_CONTEXT get set */
- ast_channel_lock_both(chan, peer);
- ast_channel_context_set(peer, ast_channel_context(chan));
- ast_channel_exten_set(peer, ast_channel_exten(chan));
- ast_channel_unlock(peer);
- ast_channel_unlock(chan);
- ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_MACRO]);
- macro_res = ast_app_exec_macro(chan, peer, opt_args[OPT_ARG_CALLEE_MACRO]);
- ast_channel_lock(peer);
- if (!macro_res && (macro_result_peer = pbx_builtin_getvar_helper(peer, "MACRO_RESULT"))) {
- char *macro_result = ast_strdupa(macro_result_peer);
- char *macro_transfer_dest;
- ast_channel_unlock(peer);
- if (!strcasecmp(macro_result, "BUSY")) {
- ast_copy_string(pa.status, macro_result, sizeof(pa.status));
- ast_set_flag64(peerflags, OPT_GO_ON);
- macro_res = -1;
- } else if (!strcasecmp(macro_result, "CONGESTION") || !strcasecmp(macro_result, "CHANUNAVAIL")) {
- ast_copy_string(pa.status, macro_result, sizeof(pa.status));
- ast_set_flag64(peerflags, OPT_GO_ON);
- macro_res = -1;
- } else if (!strcasecmp(macro_result, "CONTINUE")) {
- /* hangup peer and keep chan alive assuming the macro has changed
- the context / exten / priority or perhaps
- the next priority in the current exten is desired.
- */
- ast_set_flag64(peerflags, OPT_GO_ON);
- macro_res = -1;
- } else if (!strcasecmp(macro_result, "ABORT")) {
- /* Hangup both ends unless the caller has the g flag */
- macro_res = -1;
- } else if (!strncasecmp(macro_result, "GOTO:", 5)) {
- macro_transfer_dest = macro_result + 5;
- macro_res = -1;
- /* perform a transfer to a new extension */
- if (strchr(macro_transfer_dest, '^')) { /* context^exten^priority*/
- ast_replace_subargument_delimiter(macro_transfer_dest);
- }
- if (!ast_parseable_goto(chan, macro_transfer_dest)) {
- ast_set_flag64(peerflags, OPT_GO_ON);
- }
- }
- if (macro_res && !dial_end_raised) {
- ast_channel_publish_dial(chan, peer, NULL, macro_result);
- dial_end_raised = 1;
- }
- } else {
- ast_channel_unlock(peer);
- }
- res = macro_res;
- }
- if (ast_test_flag64(&opts, OPT_CALLEE_GOSUB) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_GOSUB])) {
- const char *gosub_result_peer;
- char *gosub_argstart;
- char *gosub_args = NULL;
- int gosub_res = -1;
- ast_replace_subargument_delimiter(opt_args[OPT_ARG_CALLEE_GOSUB]);
- gosub_argstart = strchr(opt_args[OPT_ARG_CALLEE_GOSUB], ',');
- if (gosub_argstart) {
- const char *what_is_s = "s";
- *gosub_argstart = 0;
- if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
- ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
- what_is_s = "~~s~~";
- }
- if (ast_asprintf(&gosub_args, "%s,%s,1(%s)", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s, gosub_argstart + 1) < 0) {
- gosub_args = NULL;
- }
- *gosub_argstart = ',';
- } else {
- const char *what_is_s = "s";
- if (!ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "s", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL)) &&
- ast_exists_extension(peer, opt_args[OPT_ARG_CALLEE_GOSUB], "~~s~~", 1, S_COR(ast_channel_caller(peer)->id.number.valid, ast_channel_caller(peer)->id.number.str, NULL))) {
- what_is_s = "~~s~~";
- }
- if (ast_asprintf(&gosub_args, "%s,%s,1", opt_args[OPT_ARG_CALLEE_GOSUB], what_is_s) < 0) {
- gosub_args = NULL;
- }
- }
- if (gosub_args) {
- gosub_res = ast_app_exec_sub(chan, peer, gosub_args, 0);
- ast_free(gosub_args);
- } else {
- ast_log(LOG_ERROR, "Could not Allocate string for Gosub arguments -- Gosub Call Aborted!\n");
- }
- ast_channel_lock_both(chan, peer);
- if (!gosub_res && (gosub_result_peer = pbx_builtin_getvar_helper(peer, "GOSUB_RESULT"))) {
- char *gosub_transfer_dest;
- char *gosub_result = ast_strdupa(gosub_result_peer);
- const char *gosub_retval = pbx_builtin_getvar_helper(peer, "GOSUB_RETVAL");
- /* Inherit return value from the peer, so it can be used in the master */
- if (gosub_retval) {
- pbx_builtin_setvar_helper(chan, "GOSUB_RETVAL", gosub_retval);
- }
- ast_channel_unlock(peer);
- ast_channel_unlock(chan);
- if (!strcasecmp(gosub_result, "BUSY")) {
- ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
- ast_set_flag64(peerflags, OPT_GO_ON);
- gosub_res = -1;
- } else if (!strcasecmp(gosub_result, "CONGESTION") || !strcasecmp(gosub_result, "CHANUNAVAIL")) {
- ast_copy_string(pa.status, gosub_result, sizeof(pa.status));
- ast_set_flag64(peerflags, OPT_GO_ON);
- gosub_res = -1;
- } else if (!strcasecmp(gosub_result, "CONTINUE")) {
- /* Hangup peer and continue with the next extension priority. */
- ast_set_flag64(peerflags, OPT_GO_ON);
- gosub_res = -1;
- } else if (!strcasecmp(gosub_result, "ABORT")) {
- /* Hangup both ends unless the caller has the g flag */
- gosub_res = -1;
- } else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
- gosub_transfer_dest = gosub_result + 5;
- gosub_res = -1;
- /* perform a transfer to a new extension */
- if (strchr(gosub_transfer_dest, '^')) { /* context^exten^priority*/
- ast_replace_subargument_delimiter(gosub_transfer_dest);
- }
- if (!ast_parseable_goto(chan, gosub_transfer_dest)) {
- ast_set_flag64(peerflags, OPT_GO_ON);
- }
- }
- if (gosub_res) {
- res = gosub_res;
- if (!dial_end_raised) {
- ast_channel_publish_dial(chan, peer, NULL, gosub_result);
- dial_end_raised = 1;
- }
- }
- } else {
- ast_channel_unlock(peer);
- ast_channel_unlock(chan);
- }
- }
- if (!res) {
- /* None of the Dial options changed our status; inform
- * everyone that this channel answered
- */
- if (!dial_end_raised) {
- ast_channel_publish_dial(chan, peer, NULL, "ANSWER");
- dial_end_raised = 1;
- }
- if (!ast_tvzero(calldurationlimit)) {
- struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
- ast_channel_lock(peer);
- ast_channel_whentohangup_set(peer, &whentohangup);
- ast_channel_unlock(peer);
- }
- if (!ast_strlen_zero(dtmfcalled)) {
- ast_verb(3, "Sending DTMF '%s' to the called party.\n", dtmfcalled);
- res = ast_dtmf_stream(peer, chan, dtmfcalled, 250, 0);
- }
- if (!ast_strlen_zero(dtmfcalling)) {
- ast_verb(3, "Sending DTMF '%s' to the calling party.\n", dtmfcalling);
- res = ast_dtmf_stream(chan, peer, dtmfcalling, 250, 0);
- }
- }
- if (res) { /* some error */
- if (!ast_check_hangup(chan) && ast_check_hangup(peer)) {
- ast_channel_hangupcause_set(chan, ast_channel_hangupcause(peer));
- }
- setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
- if (ast_bridge_setup_after_goto(peer)
- || ast_pbx_start(peer)) {
- ast_autoservice_chan_hangup_peer(chan, peer);
- }
- res = -1;
- } else {
- if (ast_test_flag64(peerflags, OPT_CALLEE_TRANSFER))
- ast_set_flag(&(config.features_callee), AST_FEATURE_REDIRECT);
- if (ast_test_flag64(peerflags, OPT_CALLER_TRANSFER))
- ast_set_flag(&(config.features_caller), AST_FEATURE_REDIRECT);
- if (ast_test_flag64(peerflags, OPT_CALLEE_HANGUP))
- ast_set_flag(&(config.features_callee), AST_FEATURE_DISCONNECT);
- if (ast_test_flag64(peerflags, OPT_CALLER_HANGUP))
- ast_set_flag(&(config.features_caller), AST_FEATURE_DISCONNECT);
- if (ast_test_flag64(peerflags, OPT_CALLEE_MONITOR))
- ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMON);
- if (ast_test_flag64(peerflags, OPT_CALLER_MONITOR))
- ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMON);
- if (ast_test_flag64(peerflags, OPT_CALLEE_PARK))
- ast_set_flag(&(config.features_callee), AST_FEATURE_PARKCALL);
- if (ast_test_flag64(peerflags, OPT_CALLER_PARK))
- ast_set_flag(&(config.features_caller), AST_FEATURE_PARKCALL);
- if (ast_test_flag64(peerflags, OPT_CALLEE_MIXMONITOR))
- ast_set_flag(&(config.features_callee), AST_FEATURE_AUTOMIXMON);
- if (ast_test_flag64(peerflags, OPT_CALLER_MIXMONITOR))
- ast_set_flag(&(config.features_caller), AST_FEATURE_AUTOMIXMON);
- config.end_bridge_callback = end_bridge_callback;
- config.end_bridge_callback_data = chan;
- config.end_bridge_callback_data_fixup = end_bridge_callback_data_fixup;
- if (moh) {
- moh = 0;
- ast_moh_stop(chan);
- } else if (sentringing) {
- sentringing = 0;
- ast_indicate(chan, -1);
- }
- /* Be sure no generators are left on it and reset the visible indication */
- ast_deactivate_generator(chan);
- ast_channel_visible_indication_set(chan, 0);
- /* Make sure channels are compatible */
- res = ast_channel_make_compatible(chan, peer);
- if (res < 0) {
- ast_log(LOG_WARNING, "Had to drop call because I couldn't make %s compatible with %s\n", ast_channel_name(chan), ast_channel_name(peer));
- ast_autoservice_chan_hangup_peer(chan, peer);
- res = -1;
- goto done;
- }
- if (opermode) {
- struct oprmode oprmode;
- oprmode.peer = peer;
- oprmode.mode = opermode;
- ast_channel_setoption(chan, AST_OPTION_OPRMODE, &oprmode, sizeof(oprmode), 0);
- }
- setup_peer_after_bridge_goto(chan, peer, &opts, opt_args);
- res = ast_bridge_call(chan, peer, &config);
- }
- }
- out:
- if (moh) {
- moh = 0;
- ast_moh_stop(chan);
- } else if (sentringing) {
- sentringing = 0;
- ast_indicate(chan, -1);
- }
- if (delprivintro && ast_fileexists(pa.privintro, NULL, NULL) > 0) {
- ast_filedelete(pa.privintro, NULL);
- if (ast_fileexists(pa.privintro, NULL, NULL) > 0) {
- ast_log(LOG_NOTICE, "privacy: ast_filedelete didn't do its job on %s\n", pa.privintro);
- } else {
- ast_verb(3, "Successfully deleted %s intro file\n", pa.privintro);
- }
- }
- ast_channel_early_bridge(chan, NULL);
- /* forward 'answered elsewhere' if we received it */
- hanguptree(&out_chans, NULL,
- ast_channel_hangupcause(chan) == AST_CAUSE_ANSWERED_ELSEWHERE
- || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE)
- ? AST_CAUSE_ANSWERED_ELSEWHERE : -1);
- pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
- ast_debug(1, "Exiting with DIALSTATUS=%s.\n", pa.status);
- if ((ast_test_flag64(peerflags, OPT_GO_ON)) && !ast_check_hangup(chan) && (res != AST_PBX_INCOMPLETE)) {
- if (!ast_tvzero(calldurationlimit))
- memset(ast_channel_whentohangup(chan), 0, sizeof(*ast_channel_whentohangup(chan)));
- res = 0;
- }
- done:
- if (config.warning_sound) {
- ast_free((char *)config.warning_sound);
- }
- if (config.end_sound) {
- ast_free((char *)config.end_sound);
- }
- if (config.start_sound) {
- ast_free((char *)config.start_sound);
- }
- ast_ignore_cc(chan);
- return res;
- }
- static int dial_exec(struct ast_channel *chan, const char *data)
- {
- struct ast_flags64 peerflags;
- memset(&peerflags, 0, sizeof(peerflags));
- return dial_exec_full(chan, data, &peerflags, NULL);
- }
- static int retrydial_exec(struct ast_channel *chan, const char *data)
- {
- char *parse;
- const char *context = NULL;
- int sleepms = 0, loops = 0, res = -1;
- struct ast_flags64 peerflags = { 0, };
- AST_DECLARE_APP_ARGS(args,
- AST_APP_ARG(announce);
- AST_APP_ARG(sleep);
- AST_APP_ARG(retries);
- AST_APP_ARG(dialdata);
- );
- if (ast_strlen_zero(data)) {
- ast_log(LOG_WARNING, "RetryDial requires an argument!\n");
- return -1;
- }
- parse = ast_strdupa(data);
- AST_STANDARD_APP_ARGS(args, parse);
- if (!ast_strlen_zero(args.sleep) && (sleepms = atoi(args.sleep)))
- sleepms *= 1000;
- if (!ast_strlen_zero(args.retries)) {
- loops = atoi(args.retries);
- }
- if (!args.dialdata) {
- ast_log(LOG_ERROR, "%s requires a 4th argument (dialdata)\n", rapp);
- goto done;
- }
- if (sleepms < 1000)
- sleepms = 10000;
- if (!loops)
- loops = -1; /* run forever */
- ast_channel_lock(chan);
- context = pbx_builtin_getvar_helper(chan, "EXITCONTEXT");
- context = !ast_strlen_zero(context) ? ast_strdupa(context) : NULL;
- ast_channel_unlock(chan);
- res = 0;
- while (loops) {
- int continue_exec;
- ast_channel_data_set(chan, "Retrying");
- if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
- ast_moh_stop(chan);
- res = dial_exec_full(chan, args.dialdata, &peerflags, &continue_exec);
- if (continue_exec)
- break;
- if (res == 0) {
- if (ast_test_flag64(&peerflags, OPT_DTMF_EXIT)) {
- if (!ast_strlen_zero(args.announce)) {
- if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
- if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
- ast_waitstream(chan, AST_DIGIT_ANY);
- } else
- ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
- }
- if (!res && sleepms) {
- if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
- ast_moh_start(chan, NULL, NULL);
- res = ast_waitfordigit(chan, sleepms);
- }
- } else {
- if (!ast_strlen_zero(args.announce)) {
- if (ast_fileexists(args.announce, NULL, ast_channel_language(chan)) > 0) {
- if (!(res = ast_streamfile(chan, args.announce, ast_channel_language(chan))))
- res = ast_waitstream(chan, "");
- } else
- ast_log(LOG_WARNING, "Announce file \"%s\" specified in Retrydial does not exist\n", args.announce);
- }
- if (sleepms) {
- if (!ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
- ast_moh_start(chan, NULL, NULL);
- if (!res)
- res = ast_waitfordigit(chan, sleepms);
- }
- }
- }
- if (res < 0 || res == AST_PBX_INCOMPLETE) {
- break;
- } else if (res > 0) { /* Trying to send the call elsewhere (1 digit ext) */
- if (onedigit_goto(chan, context, (char) res, 1)) {
- res = 0;
- break;
- }
- }
- loops--;
- }
- if (loops == 0)
- res = 0;
- else if (res == 1)
- res = 0;
- if (ast_test_flag(ast_channel_flags(chan), AST_FLAG_MOH))
- ast_moh_stop(chan);
- done:
- return res;
- }
- static int unload_module(void)
- {
- int res;
- res = ast_unregister_application(app);
- res |= ast_unregister_application(rapp);
- return res;
- }
- static int load_module(void)
- {
- int res;
- res = ast_register_application_xml(app, dial_exec);
- res |= ast_register_application_xml(rapp, retrydial_exec);
- return res;
- }
- AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Dialing Application");
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