This change removes AST_CEL_BRIDGE_UPDATE since it should no longer be used because masquerade situations are now accounted for in other ways. This also refactors usage of AST_CEL_FORWARD to be produced by a Dial message which has been extended with a "forward" field. (closes issue ASTERISK-21566) Review: https://reviewboard.asterisk.org/r/2635/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392829 65c4cc65-6c06-0410-ace0-fbb531ad65f3
3214 lines
118 KiB
C
3214 lines
118 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2012, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*! \file
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*
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* \brief dial() & retrydial() - Trivial application to dial a channel and send an URL on answer
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \ingroup applications
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <sys/time.h>
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#include <sys/signal.h>
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#include <sys/stat.h>
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#include <netinet/in.h>
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#include "asterisk/paths.h" /* use ast_config_AST_DATA_DIR */
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#include "asterisk/lock.h"
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#include "asterisk/file.h"
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#include "asterisk/channel.h"
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#include "asterisk/pbx.h"
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#include "asterisk/module.h"
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#include "asterisk/translate.h"
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#include "asterisk/say.h"
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#include "asterisk/config.h"
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#include "asterisk/features.h"
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#include "asterisk/musiconhold.h"
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#include "asterisk/callerid.h"
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#include "asterisk/utils.h"
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#include "asterisk/app.h"
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#include "asterisk/causes.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/manager.h"
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#include "asterisk/privacy.h"
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#include "asterisk/stringfields.h"
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#include "asterisk/global_datastores.h"
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#include "asterisk/dsp.h"
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#include "asterisk/cel.h"
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#include "asterisk/aoc.h"
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#include "asterisk/ccss.h"
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#include "asterisk/indications.h"
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#include "asterisk/framehook.h"
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#include "asterisk/dial.h"
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#include "asterisk/stasis_channels.h"
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#include "asterisk/bridging.h"
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#include "asterisk/features_config.h"
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/*** DOCUMENTATION
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<application name="Dial" language="en_US">
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<synopsis>
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Attempt to connect to another device or endpoint and bridge the call.
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</synopsis>
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<syntax>
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<parameter name="Technology/Resource" required="true" argsep="&">
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<argument name="Technology/Resource" required="true">
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<para>Specification of the device(s) to dial. These must be in the format of
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<literal>Technology/Resource</literal>, where <replaceable>Technology</replaceable>
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represents a particular channel driver, and <replaceable>Resource</replaceable>
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represents a resource available to that particular channel driver.</para>
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</argument>
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<argument name="Technology2/Resource2" required="false" multiple="true">
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<para>Optional extra devices to dial in parallel</para>
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<para>If you need more then one enter them as
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Technology2/Resource2&Technology3/Resourse3&.....</para>
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</argument>
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</parameter>
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<parameter name="timeout" required="false">
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<para>Specifies the number of seconds we attempt to dial the specified devices</para>
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<para>If not specified, this defaults to 136 years.</para>
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</parameter>
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<parameter name="options" required="false">
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<optionlist>
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<option name="A">
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<argument name="x" required="true">
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<para>The file to play to the called party</para>
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</argument>
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<para>Play an announcement to the called party, where <replaceable>x</replaceable> is the prompt to be played</para>
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</option>
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<option name="a">
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<para>Immediately answer the calling channel when the called channel answers in
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all cases. Normally, the calling channel is answered when the called channel
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answers, but when options such as A() and M() are used, the calling channel is
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not answered until all actions on the called channel (such as playing an
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announcement) are completed. This option can be used to answer the calling
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channel before doing anything on the called channel. You will rarely need to use
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this option, the default behavior is adequate in most cases.</para>
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</option>
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<option name="b" argsep="^">
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<para>Before initiating an outgoing call, Gosub to the specified
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location using the newly created channel. The Gosub will be
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executed for each destination channel.</para>
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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<argument name="priority" required="true" hasparams="optional" argsep="^">
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<argument name="arg1" multiple="true" required="true" />
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<argument name="argN" />
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</argument>
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</option>
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<option name="B" argsep="^">
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<para>Before initiating the outgoing call(s), Gosub to the specified
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location using the current channel.</para>
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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<argument name="priority" required="true" hasparams="optional" argsep="^">
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<argument name="arg1" multiple="true" required="true" />
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<argument name="argN" />
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</argument>
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</option>
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<option name="C">
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<para>Reset the call detail record (CDR) for this call.</para>
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</option>
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<option name="c">
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<para>If the Dial() application cancels this call, always set HANGUPCAUSE to 'answered elsewhere'</para>
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</option>
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<option name="d">
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<para>Allow the calling user to dial a 1 digit extension while waiting for
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a call to be answered. Exit to that extension if it exists in the
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current context, or the context defined in the <variable>EXITCONTEXT</variable> variable,
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if it exists.</para>
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<note>
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<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
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connected. If you wish to use this option with these phones, you
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can use the <literal>Answer</literal> application before dialing.</para>
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</note>
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</option>
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<option name="D" argsep=":">
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<argument name="called" />
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<argument name="calling" />
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<argument name="progress" />
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<para>Send the specified DTMF strings <emphasis>after</emphasis> the called
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party has answered, but before the call gets bridged. The
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<replaceable>called</replaceable> DTMF string is sent to the called party, and the
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<replaceable>calling</replaceable> DTMF string is sent to the calling party. Both arguments
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can be used alone. If <replaceable>progress</replaceable> is specified, its DTMF is sent
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to the called party immediately after receiving a PROGRESS message.</para>
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<para>See SendDTMF for valid digits.</para>
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</option>
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<option name="e">
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<para>Execute the <literal>h</literal> extension for peer after the call ends</para>
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</option>
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<option name="f">
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<argument name="x" required="false" />
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<para>If <replaceable>x</replaceable> is not provided, force the CallerID sent on a call-forward or
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deflection to the dialplan extension of this Dial() using a dialplan <literal>hint</literal>.
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For example, some PSTNs do not allow CallerID to be set to anything
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other than the numbers assigned to you.
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If <replaceable>x</replaceable> is provided, force the CallerID sent to <replaceable>x</replaceable>.</para>
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</option>
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<option name="F" argsep="^">
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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<argument name="priority" required="true" />
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<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party
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to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
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<note>
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<para>Any channel variables you want the called channel to inherit from the caller channel must be
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prefixed with one or two underbars ('_').</para>
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</note>
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</option>
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<option name="F">
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<para>When the caller hangs up, transfer the <emphasis>called</emphasis> party to the next priority of the current extension
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and <emphasis>start</emphasis> execution at that location.</para>
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<note>
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<para>Any channel variables you want the called channel to inherit from the caller channel must be
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prefixed with one or two underbars ('_').</para>
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</note>
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<note>
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<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
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</note>
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</option>
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<option name="g">
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<para>Proceed with dialplan execution at the next priority in the current extension if the
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destination channel hangs up.</para>
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</option>
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<option name="G" argsep="^">
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<argument name="context" required="false" />
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<argument name="exten" required="false" />
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<argument name="priority" required="true" />
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<para>If the call is answered, transfer the calling party to
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the specified <replaceable>priority</replaceable> and the called party to the specified
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<replaceable>priority</replaceable> plus one.</para>
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<note>
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<para>You cannot use any additional action post answer options in conjunction with this option.</para>
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</note>
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</option>
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<option name="h">
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<para>Allow the called party to hang up by sending the DTMF sequence
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defined for disconnect in <filename>features.conf</filename>.</para>
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</option>
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<option name="H">
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<para>Allow the calling party to hang up by sending the DTMF sequence
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defined for disconnect in <filename>features.conf</filename>.</para>
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<note>
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<para>Many SIP and ISDN phones cannot send DTMF digits until the call is
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connected. If you wish to allow DTMF disconnect before the dialed
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party answers with these phones, you can use the <literal>Answer</literal>
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application before dialing.</para>
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</note>
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</option>
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<option name="i">
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<para>Asterisk will ignore any forwarding requests it may receive on this dial attempt.</para>
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</option>
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<option name="I">
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<para>Asterisk will ignore any connected line update requests or any redirecting party
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update requests it may receive on this dial attempt.</para>
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</option>
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<option name="k">
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<para>Allow the called party to enable parking of the call by sending
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the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
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</option>
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<option name="K">
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<para>Allow the calling party to enable parking of the call by sending
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the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
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</option>
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<option name="L" argsep=":">
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<argument name="x" required="true">
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<para>Maximum call time, in milliseconds</para>
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</argument>
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<argument name="y">
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<para>Warning time, in milliseconds</para>
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</argument>
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<argument name="z">
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<para>Repeat time, in milliseconds</para>
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</argument>
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<para>Limit the call to <replaceable>x</replaceable> milliseconds. Play a warning when <replaceable>y</replaceable> milliseconds are
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left. Repeat the warning every <replaceable>z</replaceable> milliseconds until time expires.</para>
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<para>This option is affected by the following variables:</para>
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<variablelist>
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<variable name="LIMIT_PLAYAUDIO_CALLER">
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<value name="yes" default="true" />
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<value name="no" />
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<para>If set, this variable causes Asterisk to play the prompts to the caller.</para>
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</variable>
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<variable name="LIMIT_PLAYAUDIO_CALLEE">
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<value name="yes" />
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<value name="no" default="true"/>
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<para>If set, this variable causes Asterisk to play the prompts to the callee.</para>
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</variable>
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<variable name="LIMIT_TIMEOUT_FILE">
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<value name="filename"/>
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<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the timeout is reached.
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If not set, the time remaining will be announced.</para>
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</variable>
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<variable name="LIMIT_CONNECT_FILE">
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<value name="filename"/>
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<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play when the call begins.
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If not set, the time remaining will be announced.</para>
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</variable>
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<variable name="LIMIT_WARNING_FILE">
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<value name="filename"/>
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<para>If specified, <replaceable>filename</replaceable> specifies the sound prompt to play as
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a warning when time <replaceable>x</replaceable> is reached. If not set, the time remaining will be announced.</para>
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</variable>
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</variablelist>
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</option>
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<option name="m">
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<argument name="class" required="false"/>
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<para>Provide hold music to the calling party until a requested
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channel answers. A specific music on hold <replaceable>class</replaceable>
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(as defined in <filename>musiconhold.conf</filename>) can be specified.</para>
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</option>
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<option name="M" argsep="^">
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<argument name="macro" required="true">
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<para>Name of the macro that should be executed.</para>
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</argument>
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<argument name="arg" multiple="true">
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<para>Macro arguments</para>
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</argument>
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<para>Execute the specified <replaceable>macro</replaceable> for the <emphasis>called</emphasis> channel
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before connecting to the calling channel. Arguments can be specified to the Macro
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using <literal>^</literal> as a delimiter. The macro can set the variable
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<variable>MACRO_RESULT</variable> to specify the following actions after the macro is
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finished executing:</para>
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<variablelist>
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<variable name="MACRO_RESULT">
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<para>If set, this action will be taken after the macro finished executing.</para>
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<value name="ABORT">
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Hangup both legs of the call
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</value>
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<value name="CONGESTION">
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Behave as if line congestion was encountered
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</value>
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<value name="BUSY">
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Behave as if a busy signal was encountered
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</value>
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<value name="CONTINUE">
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Hangup the called party and allow the calling party to continue dialplan execution at the next priority
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</value>
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<value name="GOTO:[[<context>^]<exten>^]<priority>">
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Transfer the call to the specified destination.
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</value>
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</variable>
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</variablelist>
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<note>
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<para>You cannot use any additional action post answer options in conjunction
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with this option. Also, pbx services are run on the peer (called) channel,
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so you will not be able to set timeouts via the TIMEOUT() function in this macro.</para>
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</note>
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<warning><para>Be aware of the limitations that macros have, specifically with regards to use of
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the <literal>WaitExten</literal> application. For more information, see the documentation for
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Macro()</para></warning>
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</option>
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<option name="n">
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<argument name="delete">
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<para>With <replaceable>delete</replaceable> either not specified or set to <literal>0</literal>,
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the recorded introduction will not be deleted if the caller hangs up while the remote party has not
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yet answered.</para>
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<para>With <replaceable>delete</replaceable> set to <literal>1</literal>, the introduction will
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always be deleted.</para>
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</argument>
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<para>This option is a modifier for the call screening/privacy mode. (See the
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<literal>p</literal> and <literal>P</literal> options.) It specifies
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that no introductions are to be saved in the <directory>priv-callerintros</directory>
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directory.</para>
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</option>
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<option name="N">
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<para>This option is a modifier for the call screening/privacy mode. It specifies
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that if Caller*ID is present, do not screen the call.</para>
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</option>
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<option name="o">
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<argument name="x" required="false" />
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<para>If <replaceable>x</replaceable> is not provided, specify that the CallerID that was present on the
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<emphasis>calling</emphasis> channel be stored as the CallerID on the <emphasis>called</emphasis> channel.
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This was the behavior of Asterisk 1.0 and earlier.
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If <replaceable>x</replaceable> is provided, specify the CallerID stored on the <emphasis>called</emphasis> channel.
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Note that o(${CALLERID(all)}) is similar to option o without the parameter.</para>
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</option>
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<option name="O">
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<argument name="mode">
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<para>With <replaceable>mode</replaceable> either not specified or set to <literal>1</literal>,
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the originator hanging up will cause the phone to ring back immediately.</para>
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<para>With <replaceable>mode</replaceable> set to <literal>2</literal>, when the operator
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flashes the trunk, it will ring their phone back.</para>
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</argument>
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<para>Enables <emphasis>operator services</emphasis> mode. This option only
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works when bridging a DAHDI channel to another DAHDI channel
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only. if specified on non-DAHDI interfaces, it will be ignored.
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When the destination answers (presumably an operator services
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station), the originator no longer has control of their line.
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They may hang up, but the switch will not release their line
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until the destination party (the operator) hangs up.</para>
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</option>
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<option name="p">
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<para>This option enables screening mode. This is basically Privacy mode
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without memory.</para>
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</option>
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<option name="P">
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<argument name="x" />
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<para>Enable privacy mode. Use <replaceable>x</replaceable> as the family/key in the AstDB database if
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it is provided. The current extension is used if a database family/key is not specified.</para>
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</option>
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<option name="r">
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<para>Default: Indicate ringing to the calling party, even if the called party isn't actually ringing. Pass no audio to the calling
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party until the called channel has answered.</para>
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<argument name="tone" required="false">
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<para>Indicate progress to calling party. Send audio 'tone' from indications.conf</para>
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</argument>
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</option>
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<option name="S">
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<argument name="x" required="true" />
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<para>Hang up the call <replaceable>x</replaceable> seconds <emphasis>after</emphasis> the called party has
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answered the call.</para>
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</option>
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<option name="s">
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<argument name="x" required="true" />
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<para>Force the outgoing callerid tag parameter to be set to the string <replaceable>x</replaceable>.</para>
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<para>Works with the f option.</para>
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</option>
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<option name="t">
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<para>Allow the called party to transfer the calling party by sending the
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DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
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transfers initiated by other methods.</para>
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</option>
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<option name="T">
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<para>Allow the calling party to transfer the called party by sending the
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DTMF sequence defined in <filename>features.conf</filename>. This setting does not perform policy enforcement on
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transfers initiated by other methods.</para>
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</option>
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<option name="U" argsep="^">
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<argument name="x" required="true">
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<para>Name of the subroutine to execute via Gosub</para>
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</argument>
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<argument name="arg" multiple="true" required="false">
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<para>Arguments for the Gosub routine</para>
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</argument>
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<para>Execute via Gosub the routine <replaceable>x</replaceable> for the <emphasis>called</emphasis> channel before connecting
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to the calling channel. Arguments can be specified to the Gosub
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using <literal>^</literal> as a delimiter. The Gosub routine can set the variable
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<variable>GOSUB_RESULT</variable> to specify the following actions after the Gosub returns.</para>
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<variablelist>
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<variable name="GOSUB_RESULT">
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<value name="ABORT">
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Hangup both legs of the call.
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</value>
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<value name="CONGESTION">
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Behave as if line congestion was encountered.
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</value>
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<value name="BUSY">
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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>
|
|
<note>
|
|
<para>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 TIMEOUT() function in this routine.</para>
|
|
</note>
|
|
</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 f 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 executing 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 Set(GROUP()=...).
|
|
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 Set(GROUP()=...). Unlike <variable>OUTBOUND_GROUP</variable>,
|
|
however, the variable will be unset after use.</para>
|
|
|
|
<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="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>
|
|
</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>
|
|
</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)
|
|
|
|
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,
|
|
/* 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('r', OPT_RINGBACK, OPT_ARG_RINGBACK),
|
|
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;
|
|
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);
|
|
}
|
|
|
|
static void hanguptree(struct dial_head *out_chans, struct ast_channel *exception, int answered_elsewhere)
|
|
{
|
|
/* 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 (answered_elsewhere) {
|
|
/* This is for the channel drivers */
|
|
ast_channel_hangupcause_set(outgoing->chan, AST_CAUSE_ANSWERED_ELSEWHERE);
|
|
}
|
|
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];
|
|
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(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 {
|
|
/* Setup parameters */
|
|
c = o->chan = ast_request(tech, ast_channel_nativeformats(in), in, stuff, &cause);
|
|
if (c) {
|
|
if (single && !caller_entertained) {
|
|
ast_channel_make_compatible(o->chan, in);
|
|
}
|
|
ast_channel_lock_both(in, o->chan);
|
|
ast_channel_inherit_variables(in, o->chan);
|
|
ast_channel_datastore_inherit(in, 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_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_accountcode_set(c, ast_channel_accountcode(in));
|
|
|
|
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_clear_flag64(o, DIAL_STILLGOING);
|
|
ast_hangup(original);
|
|
ast_hangup(c);
|
|
c = o->chan = NULL;
|
|
num->nochan++;
|
|
} else {
|
|
ast_channel_lock_both(c, in);
|
|
ast_channel_publish_dial(in, c, stuff, NULL);
|
|
ast_channel_unlock(in);
|
|
ast_channel_unlock(c);
|
|
|
|
ast_channel_lock_both(original, in);
|
|
ast_channel_publish_dial_forward(in, original, NULL, "CANCEL",
|
|
ast_channel_call_forward(c));
|
|
ast_channel_unlock(in);
|
|
ast_channel_unlock(original);
|
|
|
|
/* 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);
|
|
}
|
|
}
|
|
|
|
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_party_connected_line connected_caller;
|
|
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();
|
|
|
|
ast_party_connected_line_init(&connected_caller);
|
|
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(outgoing->chan, in) < 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)) {
|
|
ast_channel_lock(outgoing->chan);
|
|
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(outgoing->chan));
|
|
ast_channel_unlock(outgoing->chan);
|
|
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
|
|
if (ast_channel_connected_line_sub(outgoing->chan, in, &connected_caller, 0) &&
|
|
ast_channel_connected_line_macro(outgoing->chan, in, &connected_caller, 1, 0)) {
|
|
ast_channel_update_connected_line(in, &connected_caller, NULL);
|
|
}
|
|
ast_party_connected_line_free(&connected_caller);
|
|
}
|
|
}
|
|
|
|
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 (!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)) {
|
|
ast_channel_lock(c);
|
|
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
|
|
ast_channel_unlock(c);
|
|
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
|
|
if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
|
|
ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
|
|
ast_channel_update_connected_line(in, &connected_caller, NULL);
|
|
}
|
|
ast_party_connected_line_free(&connected_caller);
|
|
}
|
|
}
|
|
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;
|
|
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, to,
|
|
forced_clid, stored_clid);
|
|
|
|
if (single && o->chan
|
|
&& !ast_test_flag64(o, OPT_IGNORE_CONNECTEDLINE)
|
|
&& !ast_test_flag64(o, DIAL_CALLERID_ABSENT)) {
|
|
ast_channel_lock(o->chan);
|
|
ast_connected_line_copy_from_caller(&connected_caller,
|
|
ast_channel_caller(o->chan));
|
|
ast_channel_unlock(o->chan);
|
|
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
|
|
if (ast_channel_connected_line_sub(o->chan, in, &connected_caller, 0) &&
|
|
ast_channel_connected_line_macro(o->chan, in, &connected_caller, 1, 0)) {
|
|
ast_channel_update_connected_line(in, &connected_caller, NULL);
|
|
}
|
|
ast_party_connected_line_free(&connected_caller);
|
|
}
|
|
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 (!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)) {
|
|
ast_channel_lock(c);
|
|
ast_connected_line_copy_from_caller(&connected_caller, ast_channel_caller(c));
|
|
ast_channel_unlock(c);
|
|
connected_caller.source = AST_CONNECTED_LINE_UPDATE_SOURCE_ANSWER;
|
|
if (ast_channel_connected_line_sub(c, in, &connected_caller, 0) &&
|
|
ast_channel_connected_line_macro(c, in, &connected_caller, 1, 0)) {
|
|
ast_channel_update_connected_line(in, &connected_caller, NULL);
|
|
}
|
|
ast_party_connected_line_free(&connected_caller);
|
|
}
|
|
}
|
|
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;
|
|
ast_channel_publish_dial(in, peer, NULL, "ANSWER");
|
|
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_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: %d\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_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: %d\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_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_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_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_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':
|
|
/* XXX should we set status to DENY ? */
|
|
if (ast_test_flag64(opts, OPT_PRIVACY))
|
|
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 {
|
|
/* hang up on the callee -- he didn't want to talk anyway! */
|
|
ast_autoservice_chan_hangup_peer(chan, peer);
|
|
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);
|
|
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_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_after_bridge_set_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_after_bridge_set_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];
|
|
struct ast_datastore *datastore = NULL;
|
|
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;
|
|
|
|
/* Reset all DIAL variables back to blank, to prevent confusion (in case we don't reset all of them). */
|
|
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", "");
|
|
|
|
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;
|
|
}
|
|
|
|
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;
|
|
/* find if we already dialed this interface */
|
|
struct ast_dialed_interface *di;
|
|
AST_LIST_HEAD(,ast_dialed_interface) *dialed_interfaces;
|
|
|
|
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);
|
|
ast_set2_flag64(tmp, args.url, DIAL_NOFORWARDHTML);
|
|
}
|
|
|
|
/* Request the peer */
|
|
|
|
ast_channel_lock(chan);
|
|
datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL);
|
|
/*
|
|
* 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));
|
|
ast_channel_unlock(chan);
|
|
|
|
if (datastore)
|
|
dialed_interfaces = datastore->data;
|
|
else {
|
|
if (!(datastore = ast_datastore_alloc(&dialed_interface_info, NULL))) {
|
|
ast_log(LOG_WARNING, "Unable to create channel datastore for dialed interfaces. Aborting!\n");
|
|
chanlist_free(tmp);
|
|
goto out;
|
|
}
|
|
datastore->inheritance = DATASTORE_INHERIT_FOREVER;
|
|
|
|
if (!(dialed_interfaces = ast_calloc(1, sizeof(*dialed_interfaces)))) {
|
|
ast_datastore_free(datastore);
|
|
chanlist_free(tmp);
|
|
goto out;
|
|
}
|
|
|
|
datastore->data = dialed_interfaces;
|
|
AST_LIST_HEAD_INIT(dialed_interfaces);
|
|
|
|
ast_channel_lock(chan);
|
|
ast_channel_datastore_add(chan, datastore);
|
|
ast_channel_unlock(chan);
|
|
}
|
|
|
|
AST_LIST_LOCK(dialed_interfaces);
|
|
AST_LIST_TRAVERSE(dialed_interfaces, di, list) {
|
|
if (!strcasecmp(di->interface, tmp->interface)) {
|
|
ast_log(LOG_WARNING, "Skipping dialing interface '%s' again since it has already been dialed\n",
|
|
di->interface);
|
|
break;
|
|
}
|
|
}
|
|
AST_LIST_UNLOCK(dialed_interfaces);
|
|
if (di) {
|
|
fulldial++;
|
|
chanlist_free(tmp);
|
|
continue;
|
|
}
|
|
|
|
/* It is always ok to dial a Local interface. We only keep track of
|
|
* which "real" interfaces have been dialed. The Local channel will
|
|
* inherit this list so that if it ends up dialing a real interface,
|
|
* it won't call one that has already been called. */
|
|
if (strcasecmp(tmp->tech, "Local")) {
|
|
if (!(di = ast_calloc(1, sizeof(*di) + strlen(tmp->interface)))) {
|
|
chanlist_free(tmp);
|
|
goto out;
|
|
}
|
|
strcpy(di->interface, tmp->interface);
|
|
|
|
AST_LIST_LOCK(dialed_interfaces);
|
|
AST_LIST_INSERT_TAIL(dialed_interfaces, di, list);
|
|
AST_LIST_UNLOCK(dialed_interfaces);
|
|
}
|
|
|
|
tc = ast_request(tmp->tech, ast_channel_nativeformats(chan), chan, tmp->number, &cause);
|
|
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);
|
|
}
|
|
pbx_builtin_setvar_helper(tc, "DIALEDPEERNUMBER", tmp->number);
|
|
|
|
ast_channel_lock_both(tc, chan);
|
|
|
|
/* 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_channel_appl_set(tc, "AppDial");
|
|
ast_channel_data_set(tc, "(Outgoing Line)");
|
|
ast_channel_publish_snapshot(tc);
|
|
|
|
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;
|
|
|
|
if (!ast_strlen_zero(ast_channel_accountcode(chan))) {
|
|
ast_channel_accountcode_set(tc, ast_channel_accountcode(chan));
|
|
}
|
|
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_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)) {
|
|
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);
|
|
|
|
/* The ast_channel_datastore_remove() function could fail here if the
|
|
* datastore was moved to another channel during a masquerade. If this is
|
|
* the case, don't free the datastore here because later, when the channel
|
|
* to which the datastore was moved hangs up, it will attempt to free this
|
|
* datastore again, causing a crash
|
|
*/
|
|
ast_channel_lock(chan);
|
|
datastore = ast_channel_datastore_find(chan, &dialed_interface_info, NULL); /* make sure we weren't cleaned up already */
|
|
if (datastore && !ast_channel_datastore_remove(chan, datastore)) {
|
|
ast_datastore_free(datastore);
|
|
}
|
|
ast_channel_unlock(chan);
|
|
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;
|
|
|
|
if (ast_test_flag64(&opts, OPT_CALLER_ANSWER))
|
|
ast_answer(chan);
|
|
|
|
strcpy(pa.status, "ANSWER");
|
|
pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
|
|
/* 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. */
|
|
hanguptree(&out_chans, peer, 1);
|
|
/* If appropriate, log that we have a destination channel and set the answer time */
|
|
if (ast_channel_name(peer))
|
|
pbx_builtin_setvar_helper(chan, "DIALEDPEERNAME", ast_channel_name(peer));
|
|
|
|
ast_channel_lock(peer);
|
|
number = pbx_builtin_getvar_helper(peer, "DIALEDPEERNUMBER");
|
|
if (ast_strlen_zero(number)) {
|
|
number = NULL;
|
|
} else {
|
|
number = ast_strdupa(number);
|
|
}
|
|
ast_channel_unlock(peer);
|
|
pbx_builtin_setvar_helper(chan, "DIALEDPEERNUMBER", number);
|
|
|
|
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)) {
|
|
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 announcment while monitoring the caller for a hangup */
|
|
|
|
/* 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_set_flag(ast_channel_flags(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_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;
|
|
default:
|
|
break;
|
|
}
|
|
break;
|
|
default:
|
|
/* Ignore all others */
|
|
break;
|
|
}
|
|
ast_frfree(fr);
|
|
}
|
|
ast_sched_runq(ast_channel_sched(peer));
|
|
}
|
|
ast_clear_flag(ast_channel_flags(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_clear_flag(ast_channel_flags(chan), AST_FLAG_OUTGOING);
|
|
ast_clear_flag(ast_channel_flags(peer), 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_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);
|
|
if (ast_pbx_start(peer)) {
|
|
ast_autoservice_chan_hangup_peer(chan, peer);
|
|
}
|
|
hanguptree(&out_chans, NULL, ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0);
|
|
if (continue_exec)
|
|
*continue_exec = 1;
|
|
res = 0;
|
|
goto done;
|
|
}
|
|
|
|
if (ast_test_flag64(&opts, OPT_CALLEE_MACRO) && !ast_strlen_zero(opt_args[OPT_ARG_CALLEE_MACRO])) {
|
|
const char *macro_result_peer;
|
|
|
|
/* 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]);
|
|
res = ast_app_exec_macro(chan, peer, opt_args[OPT_ARG_CALLEE_MACRO]);
|
|
|
|
ast_channel_lock(peer);
|
|
|
|
if (!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);
|
|
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);
|
|
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);
|
|
res = -1;
|
|
} else if (!strcasecmp(macro_result, "ABORT")) {
|
|
/* Hangup both ends unless the caller has the g flag */
|
|
res = -1;
|
|
} else if (!strncasecmp(macro_result, "GOTO:", 5)) {
|
|
macro_transfer_dest = macro_result + 5;
|
|
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);
|
|
}
|
|
}
|
|
} else {
|
|
ast_channel_unlock(peer);
|
|
}
|
|
}
|
|
|
|
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 res9 = -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) {
|
|
res9 = 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 (!res9 && (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);
|
|
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);
|
|
res = -1;
|
|
} else if (!strcasecmp(gosub_result, "CONTINUE")) {
|
|
/* Hangup peer and continue with the next extension priority. */
|
|
ast_set_flag64(peerflags, OPT_GO_ON);
|
|
res = -1;
|
|
} else if (!strcasecmp(gosub_result, "ABORT")) {
|
|
/* Hangup both ends unless the caller has the g flag */
|
|
res = -1;
|
|
} else if (!strncasecmp(gosub_result, "GOTO:", 5)) {
|
|
gosub_transfer_dest = gosub_result + 5;
|
|
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);
|
|
}
|
|
}
|
|
} else {
|
|
ast_channel_unlock(peer);
|
|
ast_channel_unlock(chan);
|
|
}
|
|
}
|
|
|
|
if (!res) {
|
|
if (!ast_tvzero(calldurationlimit)) {
|
|
struct timeval whentohangup = ast_tvadd(ast_tvnow(), calldurationlimit);
|
|
ast_channel_whentohangup_set(peer, &whentohangup);
|
|
}
|
|
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_after_bridge_goto_setup(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);
|
|
}
|
|
/* BUGBUG bridge needs to set hangup cause on chan when peer breaks the bridge. */
|
|
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);
|
|
hanguptree(&out_chans, NULL, ast_channel_hangupcause(chan)==AST_CAUSE_ANSWERED_ELSEWHERE || ast_test_flag64(&opts, OPT_CANCEL_ELSEWHERE) ? 1 : 0 ); /* forward 'answered elsewhere' if we received it */
|
|
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");
|