2001-12-27 11:07:33 +00:00
|
|
|
/*
|
2005-09-14 20:46:50 +00:00
|
|
|
* Asterisk -- An open source telephony toolkit.
|
2001-12-27 11:07:33 +00:00
|
|
|
*
|
2012-04-28 00:58:54 +00:00
|
|
|
* Copyright (C) 1999 - 2012, Digium, Inc.
|
|
|
|
* Copyright (C) 2012, Russell Bryant
|
2001-12-27 11:07:33 +00:00
|
|
|
*
|
2004-08-31 13:32:11 +00:00
|
|
|
* Mark Spencer <markster@digium.com>
|
2001-12-27 11:07:33 +00:00
|
|
|
*
|
2005-09-14 20:46:50 +00:00
|
|
|
* See http://www.asterisk.org for more information about
|
|
|
|
* the Asterisk project. Please do not directly contact
|
|
|
|
* any of the maintainers of this project for assistance;
|
|
|
|
* the project provides a web site, mailing lists and IRC
|
|
|
|
* channels for your use.
|
|
|
|
*
|
2001-12-27 11:07:33 +00:00
|
|
|
* This program is free software, distributed under the terms of
|
2005-09-14 20:46:50 +00:00
|
|
|
* the GNU General Public License Version 2. See the LICENSE file
|
|
|
|
* at the top of the source tree.
|
|
|
|
*/
|
|
|
|
|
2005-10-24 20:12:06 +00:00
|
|
|
/*! \file
|
2005-09-14 20:46:50 +00:00
|
|
|
*
|
2006-01-19 22:09:18 +00:00
|
|
|
* \brief Routines implementing call features as call pickup, parking and transfer
|
2005-12-30 21:18:06 +00:00
|
|
|
*
|
2012-03-22 19:51:16 +00:00
|
|
|
* \author Mark Spencer <markster@digium.com>
|
2001-12-27 11:07:33 +00:00
|
|
|
*/
|
|
|
|
|
2012-10-18 14:17:40 +00:00
|
|
|
/*! \li \ref features.c uses the configuration file \ref features.conf
|
|
|
|
* \addtogroup configuration_file Configuration Files
|
|
|
|
*/
|
|
|
|
|
|
|
|
/*!
|
|
|
|
* \page features.conf features.conf
|
|
|
|
* \verbinclude features.conf.sample
|
|
|
|
*/
|
|
|
|
|
2011-07-14 20:28:54 +00:00
|
|
|
/*** MODULEINFO
|
|
|
|
<support_level>core</support_level>
|
|
|
|
***/
|
|
|
|
|
2006-06-07 18:54:56 +00:00
|
|
|
#include "asterisk.h"
|
|
|
|
|
2008-01-23 23:09:11 +00:00
|
|
|
#include "asterisk/_private.h"
|
|
|
|
|
2005-06-06 22:12:19 +00:00
|
|
|
#include <pthread.h>
|
2010-03-20 12:03:07 +00:00
|
|
|
#include <signal.h>
|
2005-06-06 22:12:19 +00:00
|
|
|
#include <sys/time.h>
|
2016-06-03 05:57:02 +00:00
|
|
|
#include <signal.h>
|
2005-06-06 22:12:19 +00:00
|
|
|
#include <netinet/in.h>
|
|
|
|
|
2005-04-21 06:02:45 +00:00
|
|
|
#include "asterisk/lock.h"
|
|
|
|
#include "asterisk/file.h"
|
|
|
|
#include "asterisk/channel.h"
|
|
|
|
#include "asterisk/pbx.h"
|
2005-06-23 22:12:01 +00:00
|
|
|
#include "asterisk/causes.h"
|
2005-04-21 06:02:45 +00:00
|
|
|
#include "asterisk/module.h"
|
|
|
|
#include "asterisk/translate.h"
|
|
|
|
#include "asterisk/app.h"
|
|
|
|
#include "asterisk/say.h"
|
|
|
|
#include "asterisk/features.h"
|
|
|
|
#include "asterisk/musiconhold.h"
|
|
|
|
#include "asterisk/config.h"
|
|
|
|
#include "asterisk/cli.h"
|
|
|
|
#include "asterisk/manager.h"
|
|
|
|
#include "asterisk/utils.h"
|
METERMAIDS:
-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 16:43:21 +00:00
|
|
|
#include "asterisk/devicestate.h"
|
2007-11-30 21:19:57 +00:00
|
|
|
#include "asterisk/audiohook.h"
|
2008-03-01 01:30:37 +00:00
|
|
|
#include "asterisk/global_datastores.h"
|
2008-04-21 23:42:45 +00:00
|
|
|
#include "asterisk/astobj2.h"
|
2010-03-10 20:51:23 +00:00
|
|
|
#include "asterisk/test.h"
|
2013-07-25 04:06:32 +00:00
|
|
|
#include "asterisk/bridge.h"
|
2013-08-02 02:32:44 +00:00
|
|
|
#include "asterisk/bridge_features.h"
|
2013-07-25 04:06:32 +00:00
|
|
|
#include "asterisk/bridge_basic.h"
|
|
|
|
#include "asterisk/bridge_after.h"
|
2013-06-28 19:19:15 +00:00
|
|
|
#include "asterisk/stasis.h"
|
|
|
|
#include "asterisk/stasis_channels.h"
|
2013-06-06 21:40:35 +00:00
|
|
|
#include "asterisk/features_config.h"
|
2015-04-15 15:38:02 +00:00
|
|
|
#include "asterisk/max_forwards.h"
|
2013-06-06 21:40:35 +00:00
|
|
|
|
2008-11-01 21:10:07 +00:00
|
|
|
/*** DOCUMENTATION
|
|
|
|
<application name="Bridge" language="en_US">
|
|
|
|
<synopsis>
|
|
|
|
Bridge two channels.
|
|
|
|
</synopsis>
|
|
|
|
<syntax>
|
|
|
|
<parameter name="channel" required="true">
|
|
|
|
<para>The current channel is bridged to the specified <replaceable>channel</replaceable>.</para>
|
|
|
|
</parameter>
|
|
|
|
<parameter name="options">
|
|
|
|
<optionlist>
|
|
|
|
<option name="p">
|
|
|
|
<para>Play a courtesy tone to <replaceable>channel</replaceable>.</para>
|
|
|
|
</option>
|
2012-03-22 21:25:22 +00:00
|
|
|
<option name="F" argsep="^">
|
|
|
|
<argument name="context" required="false" />
|
|
|
|
<argument name="exten" required="false" />
|
|
|
|
<argument name="priority" required="true" />
|
|
|
|
<para>When the bridger hangs up, transfer the <emphasis>bridged</emphasis> party
|
|
|
|
to the specified destination and <emphasis>start</emphasis> execution at that location.</para>
|
|
|
|
<note>
|
|
|
|
<para>Any channel variables you want the called channel to inherit from the caller channel must be
|
|
|
|
prefixed with one or two underbars ('_').</para>
|
|
|
|
</note>
|
|
|
|
<note>
|
|
|
|
<para>This option will override the 'x' option</para>
|
|
|
|
</note>
|
|
|
|
</option>
|
|
|
|
<option name="F">
|
|
|
|
<para>When the bridger hangs up, transfer the <emphasis>bridged</emphasis> party
|
|
|
|
to the next priority of the current extension and <emphasis>start</emphasis> execution
|
|
|
|
at that location.</para>
|
|
|
|
<note>
|
|
|
|
<para>Any channel variables you want the called channel to inherit from the caller channel must be
|
|
|
|
prefixed with one or two underbars ('_').</para>
|
|
|
|
</note>
|
|
|
|
<note>
|
|
|
|
<para>Using this option from a Macro() or GoSub() might not make sense as there would be no return points.</para>
|
|
|
|
</note>
|
|
|
|
<note>
|
|
|
|
<para>This option will override the 'x' option</para>
|
|
|
|
</note>
|
|
|
|
</option>
|
|
|
|
|
2009-09-24 20:29:51 +00:00
|
|
|
<option name="h">
|
|
|
|
<para>Allow the called party to hang up by sending the
|
|
|
|
<replaceable>*</replaceable> DTMF digit.</para>
|
|
|
|
</option>
|
|
|
|
<option name="H">
|
|
|
|
<para>Allow the calling party to hang up by pressing the
|
|
|
|
<replaceable>*</replaceable> DTMF digit.</para>
|
|
|
|
</option>
|
|
|
|
<option name="k">
|
|
|
|
<para>Allow the called party to enable parking of the call by sending
|
2011-08-16 17:23:08 +00:00
|
|
|
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="K">
|
|
|
|
<para>Allow the calling party to enable parking of the call by sending
|
2011-08-16 17:23:08 +00:00
|
|
|
the DTMF sequence defined for call parking in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="L(x[:y][:z])">
|
|
|
|
<para>Limit the call to <replaceable>x</replaceable> ms. Play a warning
|
|
|
|
when <replaceable>y</replaceable> ms are left. Repeat the warning every
|
|
|
|
<replaceable>z</replaceable> ms. The following special variables can be
|
|
|
|
used with this option:</para>
|
|
|
|
<variablelist>
|
|
|
|
<variable name="LIMIT_PLAYAUDIO_CALLER">
|
|
|
|
<para>Play sounds to the caller. yes|no (default yes)</para>
|
|
|
|
</variable>
|
2012-03-22 19:51:16 +00:00
|
|
|
<variable name="LIMIT_PLAYAUDIO_CALLEE">
|
2009-09-24 20:29:51 +00:00
|
|
|
<para>Play sounds to the callee. yes|no</para>
|
|
|
|
</variable>
|
|
|
|
<variable name="LIMIT_TIMEOUT_FILE">
|
|
|
|
<para>File to play when time is up.</para>
|
|
|
|
</variable>
|
|
|
|
<variable name="LIMIT_CONNECT_FILE">
|
|
|
|
<para>File to play when call begins.</para>
|
|
|
|
</variable>
|
|
|
|
<variable name="LIMIT_WARNING_FILE">
|
|
|
|
<para>File to play as warning if <replaceable>y</replaceable> is
|
|
|
|
defined. The default is to say the time remaining.</para>
|
|
|
|
</variable>
|
|
|
|
</variablelist>
|
|
|
|
</option>
|
|
|
|
<option name="S(x)">
|
|
|
|
<para>Hang up the call after <replaceable>x</replaceable> seconds *after* the called party has answered the call.</para>
|
|
|
|
</option>
|
|
|
|
<option name="t">
|
|
|
|
<para>Allow the called party to transfer the calling party by sending the
|
2011-08-16 17:23:08 +00:00
|
|
|
DTMF sequence defined in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="T">
|
|
|
|
<para>Allow the calling party to transfer the called party by sending the
|
2011-08-16 17:23:08 +00:00
|
|
|
DTMF sequence defined in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="w">
|
|
|
|
<para>Allow the called party to enable recording of the call by sending
|
2011-08-16 17:23:08 +00:00
|
|
|
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="W">
|
|
|
|
<para>Allow the calling party to enable recording of the call by sending
|
2011-08-16 17:23:08 +00:00
|
|
|
the DTMF sequence defined for one-touch recording in <filename>features.conf</filename>.</para>
|
2009-09-24 20:29:51 +00:00
|
|
|
</option>
|
|
|
|
<option name="x">
|
|
|
|
<para>Cause the called party to be hung up after the bridge, instead of being
|
|
|
|
restarted in the dialplan.</para>
|
|
|
|
</option>
|
2008-11-01 21:10:07 +00:00
|
|
|
</optionlist>
|
|
|
|
</parameter>
|
|
|
|
</syntax>
|
|
|
|
<description>
|
|
|
|
<para>Allows the ability to bridge two channels via the dialplan.</para>
|
|
|
|
<para>This application sets the following channel variable upon completion:</para>
|
|
|
|
<variablelist>
|
|
|
|
<variable name="BRIDGERESULT">
|
|
|
|
<para>The result of the bridge attempt as a text string.</para>
|
|
|
|
<value name="SUCCESS" />
|
|
|
|
<value name="FAILURE" />
|
|
|
|
<value name="LOOP" />
|
|
|
|
<value name="NONEXISTENT" />
|
|
|
|
<value name="INCOMPATIBLE" />
|
|
|
|
</variable>
|
|
|
|
</variablelist>
|
|
|
|
</description>
|
2016-08-14 01:15:58 +00:00
|
|
|
<see-also>
|
|
|
|
<ref type="manager">Bridge</ref>
|
|
|
|
<ref type="managerEvent">BridgeCreate</ref>
|
|
|
|
<ref type="managerEvent">BridgeEnter</ref>
|
|
|
|
</see-also>
|
2008-11-01 21:10:07 +00:00
|
|
|
</application>
|
2009-05-22 17:52:35 +00:00
|
|
|
<manager name="Bridge" language="en_US">
|
|
|
|
<synopsis>
|
|
|
|
Bridge two channels already in the PBX.
|
|
|
|
</synopsis>
|
|
|
|
<syntax>
|
|
|
|
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
|
|
|
|
<parameter name="Channel1" required="true">
|
|
|
|
<para>Channel to Bridge to Channel2.</para>
|
|
|
|
</parameter>
|
|
|
|
<parameter name="Channel2" required="true">
|
|
|
|
<para>Channel to Bridge to Channel1.</para>
|
|
|
|
</parameter>
|
|
|
|
<parameter name="Tone">
|
|
|
|
<para>Play courtesy tone to Channel 2.</para>
|
|
|
|
<enumlist>
|
|
|
|
<enum name="no" />
|
2013-05-28 14:45:31 +00:00
|
|
|
<enum name="Channel1" />
|
|
|
|
<enum name="Channel2" />
|
|
|
|
<enum name="Both" />
|
2009-05-22 17:52:35 +00:00
|
|
|
</enumlist>
|
|
|
|
</parameter>
|
|
|
|
</syntax>
|
|
|
|
<description>
|
|
|
|
<para>Bridge together two channels already in the PBX.</para>
|
|
|
|
</description>
|
2016-08-14 01:15:58 +00:00
|
|
|
<see-also>
|
|
|
|
<ref type="application">Bridge</ref>
|
|
|
|
<ref type="managerEvent">BridgeCreate</ref>
|
|
|
|
<ref type="managerEvent">BridgeEnter</ref>
|
|
|
|
<ref type="manager">BridgeDestroy</ref>
|
|
|
|
<ref type="manager">BridgeInfo</ref>
|
|
|
|
<ref type="manager">BridgeKick</ref>
|
|
|
|
<ref type="manager">BridgeList</ref>
|
|
|
|
</see-also>
|
2009-05-22 17:52:35 +00:00
|
|
|
</manager>
|
2008-11-01 21:10:07 +00:00
|
|
|
***/
|
|
|
|
|
Merged revisions 310902 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8
................
r310902 | twilson | 2011-03-16 12:19:57 -0500 (Wed, 16 Mar 2011) | 43 lines
Merged revisions 310889 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
................
r310889 | twilson | 2011-03-16 12:03:27 -0500 (Wed, 16 Mar 2011) | 36 lines
Merged revisions 310888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r310888 | twilson | 2011-03-16 11:58:42 -0500 (Wed, 16 Mar 2011) | 29 lines
Don't delay DTMF in core bridge while listening for DTMF features
This patch is mostly the work of Olle Johansson. I did some cleanup and
added the silence generating code if transmit_silence is set.
When a channel listens for DTMF in the core bridge, the outbound DTMF is not
sent until we have received DTMF_END. For a long DTMF, this is a disaster. We
send 4 seconds of DTMF to Asterisk, which sends no audio for those 4 seconds.
Some products see this delay and the time skew on RTP packets that results and
start ignoring the audio that is sent afterward.
With this change, the DTMF_BEGIN frame is inspected and checked. If it matches
a feature code, we wait for DTMF_END and activate the feature as before. If
transmit_silence=yes in asterisk.conf, silence is sent if we paritally match a
multi-digit feature. If it doesn't match a feature, the frame is forwarded
along with the DTMF_END without delay. By doing it this way, DTMF is not delayed.
(closes issue #15642)
Reported by: jasonshugart
Patches:
issue_15652_dtmf_ast-1.4.patch.txt uploaded by twilson (license 396)
Tested by: globalnetinc, jde
(closes issue #16625)
Reported by: sharvanek
Review: https://reviewboard.asterisk.org/r/1092/
Review: https://reviewboard.asterisk.org/r/1125/
........
................
................
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@310941 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-03-16 17:29:16 +00:00
|
|
|
typedef enum {
|
|
|
|
FEATURE_INTERPRET_DETECT, /* Used by ast_feature_detect */
|
|
|
|
FEATURE_INTERPRET_DO, /* Used by feature_interpret */
|
|
|
|
FEATURE_INTERPRET_CHECK, /* Used by feature_check */
|
|
|
|
} feature_interpret_op;
|
|
|
|
|
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 21:29:12 +00:00
|
|
|
struct ast_dial_features {
|
2012-04-25 01:26:44 +00:00
|
|
|
/*! Channel's feature flags. */
|
|
|
|
struct ast_flags my_features;
|
|
|
|
/*! Bridge peer's feature flags. */
|
|
|
|
struct ast_flags peer_features;
|
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 21:29:12 +00:00
|
|
|
};
|
2001-12-27 11:07:33 +00:00
|
|
|
|
Merged revisions 172517 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r172517 | twilson | 2009-01-30 11:47:41 -0600 (Fri, 30 Jan 2009) | 37 lines
Fix feature inheritance with builtin features
When using builtin features like parking and transfers, the AST_FEATURE_* flags
would not be set correctly for all instances when either performing a builtin
attended transfer, or parking a call and getting the timeout callback. Also,
there was no way on a per-call basis to specify what features someone should
have on picking up a parked call (since that doesn't involve the Dial() command).
There was a global option for setting whether or not all users who pickup a
parked call should have AST_FEATURE_REDIRECT set, but nothing for DISCONNECT,
AUTOMON, or PARKCALL.
This patch:
1) adds the BRIDGE_FEATURES dialplan variable which can be set either in the
dialplan or with setvar in channels that support it. This variable can be set
to any combination of 't', 'k', 'w', and 'h' (case insensitive matching of the
equivalent dial options), to set what features should be activated on this
channel. The patch moves the setting of the features datastores into the
bridging code instead of app_dial to help facilitate this.
2) adds global options parkedcallparking, parkedcallhangup, and
parkedcallrecording to be similar to the parkedcalltransfers option for
globally setting features.
3) has builtin_atxfer call builtin_parkcall if being transfered to the parking
extension since tracking everything through multiple masquerades, etc. is
difficult and error-prone
4) attempts to fix all cases of return calls from parking and completed builtin
transfers not having the correct permissions
(closes issue #14274)
Reported by: aragon
Patches:
fix_feature_inheritence.diff.txt uploaded by otherwiseguy (license 396)
Tested by: aragon, otherwiseguy
Review http://reviewboard.digium.com/r/138/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-30 21:29:12 +00:00
|
|
|
static void *dial_features_duplicate(void *data)
|
|
|
|
{
|
|
|
|
struct ast_dial_features *df = data, *df_copy;
|
2012-03-22 19:51:16 +00:00
|
|
|
|
|
|
|
if (!(df_copy = ast_calloc(1, sizeof(*df)))) {
|
|
|
|
return NULL;
|
|
|
|
}
|
|
|
|
|
|
|
|
memcpy(df_copy, df, sizeof(*df));
|
|
|
|
|
|
|
|
return df_copy;
|
2009-06-15 17:34:30 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
static const struct ast_datastore_info dial_features_info = {
|
2012-03-22 19:51:16 +00:00
|
|
|
.type = "dial-features",
|
2013-04-10 23:08:02 +00:00
|
|
|
.destroy = ast_free_ptr,
|
2012-03-22 19:51:16 +00:00
|
|
|
.duplicate = dial_features_duplicate,
|
2011-06-09 16:47:07 +00:00
|
|
|
};
|
2012-03-22 19:51:16 +00:00
|
|
|
|
2012-04-25 01:26:44 +00:00
|
|
|
/*!
|
|
|
|
* \internal
|
|
|
|
* \brief Set the features datastore if it doesn't exist.
|
|
|
|
*
|
|
|
|
* \param chan Channel to add features datastore
|
|
|
|
* \param my_features The channel's feature flags
|
|
|
|
* \param peer_features The channel's bridge peer feature flags
|
|
|
|
*
|
|
|
|
* \retval TRUE if features datastore already existed.
|
|
|
|
*/
|
|
|
|
static int add_features_datastore(struct ast_channel *chan, const struct ast_flags *my_features, const struct ast_flags *peer_features)
|
|
|
|
{
|
|
|
|
struct ast_datastore *datastore;
|
|
|
|
struct ast_dial_features *dialfeatures;
|
|
|
|
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
datastore = ast_channel_datastore_find(chan, &dial_features_info, NULL);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
if (datastore) {
|
|
|
|
/* Already exists. */
|
|
|
|
return 1;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Create a new datastore with specified feature flags. */
|
|
|
|
datastore = ast_datastore_alloc(&dial_features_info, NULL);
|
|
|
|
if (!datastore) {
|
|
|
|
ast_log(LOG_WARNING, "Unable to create channel features datastore.\n");
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
dialfeatures = ast_calloc(1, sizeof(*dialfeatures));
|
|
|
|
if (!dialfeatures) {
|
|
|
|
ast_log(LOG_WARNING, "Unable to allocate memory for feature flags.\n");
|
|
|
|
ast_datastore_free(datastore);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
ast_copy_flags(&dialfeatures->my_features, my_features, AST_FLAGS_ALL);
|
|
|
|
ast_copy_flags(&dialfeatures->peer_features, peer_features, AST_FLAGS_ALL);
|
|
|
|
datastore->inheritance = DATASTORE_INHERIT_FOREVER;
|
|
|
|
datastore->data = dialfeatures;
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
ast_channel_datastore_add(chan, datastore);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2012-03-22 19:51:16 +00:00
|
|
|
struct ast_bridge_thread_obj
|
2005-01-05 19:56:47 +00:00
|
|
|
{
|
|
|
|
struct ast_bridge_config bconfig;
|
|
|
|
struct ast_channel *chan;
|
|
|
|
struct ast_channel *peer;
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
unsigned int return_to_pbx:1;
|
2005-01-05 19:56:47 +00:00
|
|
|
};
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
static void set_config_flags(struct ast_channel *chan, struct ast_bridge_config *config)
|
2005-08-23 14:40:03 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_clear_flag(config, AST_FLAGS_ALL);
|
2005-08-23 14:40:03 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_test_flag(&config->features_caller, AST_FEATURE_DTMF_MASK)) {
|
|
|
|
ast_set_flag(config, AST_BRIDGE_DTMF_CHANNEL_0);
|
2008-12-11 17:06:16 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_test_flag(&config->features_callee, AST_FEATURE_DTMF_MASK)) {
|
|
|
|
ast_set_flag(config, AST_BRIDGE_DTMF_CHANNEL_1);
|
2005-08-23 02:22:33 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (!(ast_test_flag(config, AST_BRIDGE_DTMF_CHANNEL_0) && ast_test_flag(config, AST_BRIDGE_DTMF_CHANNEL_1))) {
|
|
|
|
RAII_VAR(struct ao2_container *, applicationmap, NULL, ao2_cleanup);
|
2005-08-23 02:22:33 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
applicationmap = ast_get_chan_applicationmap(chan);
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
|
|
|
if (!applicationmap) {
|
|
|
|
return;
|
2007-05-31 18:21:47 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/* If an applicationmap exists for this channel at all, then the channel needs the DTMF flag set */
|
|
|
|
ast_set_flag(config, AST_BRIDGE_DTMF_CHANNEL_0);
|
2007-05-31 18:21:47 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
void ast_channel_log(char *title, struct ast_channel *chan);
|
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
void ast_channel_log(char *title, struct ast_channel *chan) /* for debug, this is handy enough to justify keeping it in the source */
|
|
|
|
{
|
|
|
|
ast_log(LOG_NOTICE, "______ %s (%lx)______\n", title, (unsigned long) chan);
|
|
|
|
ast_log(LOG_NOTICE, "CHAN: name: %s; appl: %s; data: %s; contxt: %s; exten: %s; pri: %d;\n",
|
|
|
|
ast_channel_name(chan), ast_channel_appl(chan), ast_channel_data(chan),
|
|
|
|
ast_channel_context(chan), ast_channel_exten(chan), ast_channel_priority(chan));
|
|
|
|
ast_log(LOG_NOTICE, "CHAN: acctcode: %s; dialcontext: %s; amaflags: %x; maccontxt: %s; macexten: %s; macpri: %d;\n",
|
|
|
|
ast_channel_accountcode(chan), ast_channel_dialcontext(chan), ast_channel_amaflags(chan),
|
|
|
|
ast_channel_macrocontext(chan), ast_channel_macroexten(chan), ast_channel_macropriority(chan));
|
|
|
|
ast_log(LOG_NOTICE, "CHAN: masq: %p; masqr: %p; uniqueID: %s; linkedID:%s\n",
|
|
|
|
ast_channel_masq(chan), ast_channel_masqr(chan),
|
|
|
|
ast_channel_uniqueid(chan), ast_channel_linkedid(chan));
|
|
|
|
if (ast_channel_masqr(chan)) {
|
|
|
|
ast_log(LOG_NOTICE, "CHAN: masquerading as: %s; cdr: %p;\n",
|
|
|
|
ast_channel_name(ast_channel_masqr(chan)), ast_channel_cdr(ast_channel_masqr(chan)));
|
2012-12-12 04:43:18 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_log(LOG_NOTICE, "===== done ====\n");
|
Merged revisions 183126 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r183126 | dvossel | 2009-03-19 11:15:16 -0500 (Thu, 19 Mar 2009) | 17 lines
Allow disconnect feature before a call is bridged
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
(closes issue #11583)
Reported by: sobomax
Patches:
patch-apps__app_dial.c uploaded by sobomax (license 359)
11583.latest-patch uploaded by murf (license 17)
detect_disconnect.diff uploaded by dvossel (license 671)
Tested by: sobomax, dvossel
Review: http://reviewboard.digium.com/r/195/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183172 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-19 16:28:33 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
static void set_bridge_features_on_config(struct ast_bridge_config *config, const char *features)
|
2005-01-04 04:01:40 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
const char *feature;
|
2007-05-08 16:31:16 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_strlen_zero(features)) {
|
|
|
|
return;
|
2005-01-04 04:01:40 +00:00
|
|
|
}
|
2005-09-07 21:36:30 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
for (feature = features; *feature; feature++) {
|
|
|
|
struct ast_flags *party;
|
2010-07-09 21:57:21 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (isupper(*feature)) {
|
|
|
|
party = &config->features_caller;
|
|
|
|
} else {
|
|
|
|
party = &config->features_callee;
|
2005-09-07 21:36:30 +00:00
|
|
|
}
|
2005-01-04 04:01:40 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
switch (tolower(*feature)) {
|
|
|
|
case 't' :
|
|
|
|
ast_set_flag(party, AST_FEATURE_REDIRECT);
|
|
|
|
break;
|
|
|
|
case 'k' :
|
|
|
|
ast_set_flag(party, AST_FEATURE_PARKCALL);
|
|
|
|
break;
|
|
|
|
case 'h' :
|
|
|
|
ast_set_flag(party, AST_FEATURE_DISCONNECT);
|
|
|
|
break;
|
|
|
|
case 'w' :
|
|
|
|
ast_set_flag(party, AST_FEATURE_AUTOMON);
|
|
|
|
break;
|
|
|
|
case 'x' :
|
|
|
|
ast_set_flag(party, AST_FEATURE_AUTOMIXMON);
|
|
|
|
break;
|
|
|
|
default :
|
|
|
|
ast_log(LOG_WARNING, "Skipping unknown feature code '%c'\n", *feature);
|
2011-08-16 17:23:08 +00:00
|
|
|
break;
|
2008-04-21 23:42:45 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
static void add_features_datastores(struct ast_channel *caller, struct ast_channel *callee, struct ast_bridge_config *config)
|
2008-04-21 23:42:45 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
if (add_features_datastore(caller, &config->features_caller, &config->features_callee)) {
|
|
|
|
/*
|
|
|
|
* If we don't return here, then when we do a builtin_atxfer we
|
|
|
|
* will copy the disconnect flags over from the atxfer to the
|
|
|
|
* callee (Party C).
|
|
|
|
*/
|
|
|
|
return;
|
2001-12-27 11:07:33 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
|
|
|
|
add_features_datastore(callee, &config->features_callee, &config->features_caller);
|
2001-12-27 11:07:33 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
static void bridge_config_set_limits_warning_values(struct ast_bridge_config *config, struct ast_bridge_features_limits *limits)
|
2011-08-16 17:23:08 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
if (config->end_sound) {
|
|
|
|
ast_string_field_set(limits, duration_sound, config->end_sound);
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (config->warning_sound) {
|
|
|
|
ast_string_field_set(limits, warning_sound, config->warning_sound);
|
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (config->start_sound) {
|
|
|
|
ast_string_field_set(limits, connect_sound, config->start_sound);
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2010-09-15 19:23:56 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
limits->frequency = config->warning_freq;
|
|
|
|
limits->warning = config->play_warning;
|
|
|
|
}
|
2008-04-21 23:42:45 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/*!
|
|
|
|
* \internal brief Setup limit hook structures on calls that need limits
|
|
|
|
*
|
|
|
|
* \param config ast_bridge_config which provides the limit data
|
|
|
|
* \param caller_limits pointer to an ast_bridge_features_limits struct which will store the caller side limits
|
|
|
|
* \param callee_limits pointer to an ast_bridge_features_limits struct which will store the callee side limits
|
|
|
|
*/
|
|
|
|
static void bridge_config_set_limits(struct ast_bridge_config *config, struct ast_bridge_features_limits *caller_limits, struct ast_bridge_features_limits *callee_limits)
|
|
|
|
{
|
|
|
|
if (ast_test_flag(&config->features_caller, AST_FEATURE_PLAY_WARNING)) {
|
|
|
|
bridge_config_set_limits_warning_values(config, caller_limits);
|
2008-04-21 23:42:45 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_test_flag(&config->features_callee, AST_FEATURE_PLAY_WARNING)) {
|
|
|
|
bridge_config_set_limits_warning_values(config, callee_limits);
|
2008-04-21 23:42:45 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
caller_limits->duration = config->timelimit;
|
|
|
|
callee_limits->duration = config->timelimit;
|
2008-04-21 23:42:45 +00:00
|
|
|
}
|
|
|
|
|
2011-08-16 17:23:08 +00:00
|
|
|
/*!
|
|
|
|
* \internal
|
2013-06-06 21:40:35 +00:00
|
|
|
* \brief Check if Monitor needs to be started on a channel.
|
|
|
|
* \since 12.0.0
|
2011-08-16 17:23:08 +00:00
|
|
|
*
|
2013-06-06 21:40:35 +00:00
|
|
|
* \param chan The bridge considers this channel the caller.
|
|
|
|
* \param peer The bridge considers this channel the callee.
|
2011-08-16 17:23:08 +00:00
|
|
|
*
|
|
|
|
* \return Nothing
|
|
|
|
*/
|
2013-06-06 21:40:35 +00:00
|
|
|
static void bridge_check_monitor(struct ast_channel *chan, struct ast_channel *peer)
|
2011-08-16 17:23:08 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
const char *value;
|
|
|
|
const char *monitor_args = NULL;
|
|
|
|
struct ast_channel *monitor_chan = NULL;
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
value = pbx_builtin_getvar_helper(chan, "AUTO_MONITOR");
|
|
|
|
if (!ast_strlen_zero(value)) {
|
|
|
|
monitor_args = ast_strdupa(value);
|
|
|
|
monitor_chan = chan;
|
|
|
|
}
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
if (!monitor_chan) {
|
|
|
|
ast_channel_lock(peer);
|
|
|
|
value = pbx_builtin_getvar_helper(peer, "AUTO_MONITOR");
|
|
|
|
if (!ast_strlen_zero(value)) {
|
|
|
|
monitor_args = ast_strdupa(value);
|
|
|
|
monitor_chan = peer;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_unlock(peer);
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
if (monitor_chan) {
|
|
|
|
struct ast_app *monitor_app;
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
monitor_app = pbx_findapp("Monitor");
|
|
|
|
if (monitor_app) {
|
|
|
|
pbx_exec(monitor_chan, monitor_app, monitor_args);
|
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/*!
|
|
|
|
* \internal
|
|
|
|
* \brief Send the peer channel on its way on bridge start failure.
|
|
|
|
* \since 12.0.0
|
|
|
|
*
|
|
|
|
* \param chan Chan to put into autoservice.
|
|
|
|
* \param peer Chan to send to after bridge goto or run hangup handlers and hangup.
|
|
|
|
*
|
|
|
|
* \return Nothing
|
|
|
|
*/
|
|
|
|
static void bridge_failed_peer_goto(struct ast_channel *chan, struct ast_channel *peer)
|
|
|
|
{
|
2013-07-25 02:20:23 +00:00
|
|
|
if (ast_bridge_setup_after_goto(peer)
|
2013-06-06 21:40:35 +00:00
|
|
|
|| ast_pbx_start(peer)) {
|
|
|
|
ast_autoservice_chan_hangup_peer(chan, peer);
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
static int pre_bridge_setup(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config,
|
|
|
|
struct ast_bridge_features *chan_features, struct ast_bridge_features *peer_features)
|
|
|
|
{
|
|
|
|
int res;
|
|
|
|
|
|
|
|
set_bridge_features_on_config(config, pbx_builtin_getvar_helper(chan, "BRIDGE_FEATURES"));
|
|
|
|
add_features_datastores(chan, peer, config);
|
|
|
|
|
|
|
|
/*
|
|
|
|
* This is an interesting case. One example is if a ringing
|
|
|
|
* channel gets redirected to an extension that picks up a
|
|
|
|
* parked call. This will make sure that the call taken out of
|
|
|
|
* parking gets told that the channel it just got bridged to is
|
|
|
|
* still ringing.
|
|
|
|
*/
|
|
|
|
if (ast_channel_state(chan) == AST_STATE_RINGING
|
|
|
|
&& ast_channel_visible_indication(peer) != AST_CONTROL_RINGING) {
|
|
|
|
ast_indicate(peer, AST_CONTROL_RINGING);
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
bridge_check_monitor(chan, peer);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
set_config_flags(chan, config);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/* Answer if need be */
|
|
|
|
if (ast_channel_state(chan) != AST_STATE_UP) {
|
2013-06-17 03:00:38 +00:00
|
|
|
if (ast_raw_answer(chan)) {
|
2013-06-06 21:40:35 +00:00
|
|
|
return -1;
|
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
#ifdef FOR_DEBUG
|
|
|
|
/* show the two channels and cdrs involved in the bridge for debug & devel purposes */
|
|
|
|
ast_channel_log("Pre-bridge CHAN Channel info", chan);
|
|
|
|
ast_channel_log("Pre-bridge PEER Channel info", peer);
|
|
|
|
#endif
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
res = 0;
|
|
|
|
ast_channel_lock(chan);
|
2015-04-15 15:38:02 +00:00
|
|
|
ast_max_forwards_reset(chan);
|
2013-08-05 20:18:54 +00:00
|
|
|
res |= ast_bridge_features_ds_append(chan, &config->features_caller);
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_unlock(chan);
|
|
|
|
ast_channel_lock(peer);
|
2015-04-15 15:38:02 +00:00
|
|
|
ast_max_forwards_reset(peer);
|
2013-08-05 20:18:54 +00:00
|
|
|
res |= ast_bridge_features_ds_append(peer, &config->features_callee);
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_unlock(peer);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (res) {
|
|
|
|
return -1;
|
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (config->timelimit) {
|
|
|
|
struct ast_bridge_features_limits call_duration_limits_chan;
|
|
|
|
struct ast_bridge_features_limits call_duration_limits_peer;
|
|
|
|
int abandon_call = 0; /* TRUE if set limits fails so we can abandon the call. */
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_bridge_features_limits_construct(&call_duration_limits_chan)) {
|
|
|
|
ast_log(LOG_ERROR, "Could not construct caller duration limits. Bridge canceled.\n");
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
return -1;
|
2011-08-09 23:17:13 +00:00
|
|
|
}
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_bridge_features_limits_construct(&call_duration_limits_peer)) {
|
|
|
|
ast_log(LOG_ERROR, "Could not construct callee duration limits. Bridge canceled.\n");
|
|
|
|
ast_bridge_features_limits_destroy(&call_duration_limits_chan);
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
return -1;
|
2008-04-21 23:42:45 +00:00
|
|
|
}
|
2011-08-09 23:17:13 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
bridge_config_set_limits(config, &call_duration_limits_chan, &call_duration_limits_peer);
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (ast_bridge_features_set_limits(chan_features, &call_duration_limits_chan, 0)) {
|
|
|
|
abandon_call = 1;
|
|
|
|
}
|
|
|
|
if (ast_bridge_features_set_limits(peer_features, &call_duration_limits_peer, 0)) {
|
|
|
|
abandon_call = 1;
|
2011-08-09 23:17:13 +00:00
|
|
|
}
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/* At this point we are done with the limits structs since they have been copied to the individual feature sets. */
|
|
|
|
ast_bridge_features_limits_destroy(&call_duration_limits_chan);
|
|
|
|
ast_bridge_features_limits_destroy(&call_duration_limits_peer);
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (abandon_call) {
|
|
|
|
ast_log(LOG_ERROR, "Could not set duration limits on one or more sides of the call. Bridge canceled.\n");
|
|
|
|
return -1;
|
2008-01-23 23:09:11 +00:00
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2011-08-16 17:23:08 +00:00
|
|
|
return 0;
|
|
|
|
}
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-08-22 18:52:41 +00:00
|
|
|
int ast_bridge_call_with_flags(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config, unsigned int flags)
|
2011-08-16 17:23:08 +00:00
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
int res;
|
|
|
|
struct ast_bridge *bridge;
|
|
|
|
struct ast_bridge_features chan_features;
|
|
|
|
struct ast_bridge_features *peer_features;
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/* Setup features. */
|
|
|
|
res = ast_bridge_features_init(&chan_features);
|
|
|
|
peer_features = ast_bridge_features_new();
|
|
|
|
if (res || !peer_features) {
|
|
|
|
ast_bridge_features_destroy(peer_features);
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
bridge_failed_peer_goto(chan, peer);
|
|
|
|
return -1;
|
2008-01-23 23:09:11 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
|
|
|
|
if (pre_bridge_setup(chan, peer, config, &chan_features, peer_features)) {
|
|
|
|
ast_bridge_features_destroy(peer_features);
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
bridge_failed_peer_goto(chan, peer);
|
|
|
|
return -1;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
|
|
|
|
/* Create bridge */
|
|
|
|
bridge = ast_bridge_basic_new();
|
|
|
|
if (!bridge) {
|
|
|
|
ast_bridge_features_destroy(peer_features);
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
bridge_failed_peer_goto(chan, peer);
|
|
|
|
return -1;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2013-08-22 18:52:41 +00:00
|
|
|
ast_bridge_basic_set_flags(bridge, flags);
|
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/* Put peer into the bridge */
|
2013-09-13 22:19:23 +00:00
|
|
|
if (ast_bridge_impart(bridge, peer, NULL, peer_features,
|
|
|
|
AST_BRIDGE_IMPART_CHAN_INDEPENDENT | AST_BRIDGE_IMPART_INHIBIT_JOIN_COLP)) {
|
2013-08-22 21:09:52 +00:00
|
|
|
ast_bridge_destroy(bridge, 0);
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
bridge_failed_peer_goto(chan, peer);
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
/* Join bridge */
|
2013-09-13 22:19:23 +00:00
|
|
|
ast_bridge_join(bridge, chan, NULL, &chan_features, NULL,
|
|
|
|
AST_BRIDGE_JOIN_PASS_REFERENCE | AST_BRIDGE_JOIN_INHIBIT_JOIN_COLP);
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
/*
|
|
|
|
* If the bridge was broken for a hangup that isn't real, then
|
|
|
|
* don't run the h extension, because the channel isn't really
|
|
|
|
* hung up. This should really only happen with
|
|
|
|
* AST_SOFTHANGUP_ASYNCGOTO.
|
|
|
|
*/
|
|
|
|
res = -1;
|
|
|
|
ast_channel_lock(chan);
|
|
|
|
if (ast_channel_softhangup_internal_flag(chan) & AST_SOFTHANGUP_ASYNCGOTO) {
|
|
|
|
res = 0;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_channel_unlock(chan);
|
2008-01-23 23:09:11 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
2003-07-02 14:06:12 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (res && config->end_bridge_callback) {
|
|
|
|
config->end_bridge_callback(config->end_bridge_callback_data);
|
2008-03-04 23:04:29 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
|
|
|
|
return res;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2007-06-06 14:45:29 +00:00
|
|
|
|
2013-08-22 18:52:41 +00:00
|
|
|
/*!
|
|
|
|
* \brief bridge the call and set CDR
|
|
|
|
*
|
|
|
|
* \param chan The bridge considers this channel the caller.
|
|
|
|
* \param peer The bridge considers this channel the callee.
|
|
|
|
* \param config Configuration for this bridge.
|
|
|
|
*
|
|
|
|
* Set start time, check for two channels,check if monitor on
|
|
|
|
* check for feature activation, create new CDR
|
|
|
|
* \retval res on success.
|
|
|
|
* \retval -1 on failure to bridge.
|
|
|
|
*/
|
|
|
|
int ast_bridge_call(struct ast_channel *chan, struct ast_channel *peer, struct ast_bridge_config *config)
|
|
|
|
{
|
|
|
|
return ast_bridge_call_with_flags(chan, peer, config, 0);
|
|
|
|
}
|
|
|
|
|
2013-05-28 14:45:31 +00:00
|
|
|
enum play_tone_action {
|
|
|
|
PLAYTONE_NONE = 0,
|
|
|
|
PLAYTONE_CHANNEL1 = (1 << 0),
|
|
|
|
PLAYTONE_CHANNEL2 = (1 << 1),
|
|
|
|
PLAYTONE_BOTH = PLAYTONE_CHANNEL1 | PLAYTONE_CHANNEL2,
|
|
|
|
};
|
|
|
|
|
|
|
|
static enum play_tone_action parse_playtone(const char *playtone_val)
|
|
|
|
{
|
|
|
|
if (ast_strlen_zero(playtone_val) || ast_false(playtone_val)) {
|
|
|
|
return PLAYTONE_NONE;
|
|
|
|
} if (!strcasecmp(playtone_val, "channel1")) {
|
|
|
|
return PLAYTONE_CHANNEL1;
|
|
|
|
} else if (!strcasecmp(playtone_val, "channel2") || ast_true(playtone_val)) {
|
|
|
|
return PLAYTONE_CHANNEL2;
|
|
|
|
} else if (!strcasecmp(playtone_val, "both")) {
|
|
|
|
return PLAYTONE_BOTH;
|
|
|
|
} else {
|
|
|
|
/* Invalid input. Assume none */
|
|
|
|
return PLAYTONE_NONE;
|
|
|
|
}
|
|
|
|
}
|
|
|
|
|
2011-08-16 17:23:08 +00:00
|
|
|
/*!
|
|
|
|
* \brief Bridge channels together
|
|
|
|
* \param s
|
|
|
|
* \param m
|
2012-03-22 19:51:16 +00:00
|
|
|
*
|
|
|
|
* Make sure valid channels were specified,
|
2011-08-16 17:23:08 +00:00
|
|
|
* send errors if any of the channels could not be found/locked, answer channels if needed,
|
2012-03-22 19:51:16 +00:00
|
|
|
* create the placeholder channels and grab the other channels
|
|
|
|
* make the channels compatible, send error if we fail doing so
|
2011-08-16 17:23:08 +00:00
|
|
|
* setup the bridge thread object and start the bridge.
|
2012-03-22 19:51:16 +00:00
|
|
|
*
|
2012-06-23 00:29:18 +00:00
|
|
|
* \retval 0
|
2011-08-16 17:23:08 +00:00
|
|
|
*/
|
|
|
|
static int action_bridge(struct mansession *s, const struct message *m)
|
|
|
|
{
|
|
|
|
const char *channela = astman_get_header(m, "Channel1");
|
|
|
|
const char *channelb = astman_get_header(m, "Channel2");
|
2013-05-28 14:45:31 +00:00
|
|
|
enum play_tone_action playtone = parse_playtone(astman_get_header(m, "Tone"));
|
|
|
|
RAII_VAR(struct ast_channel *, chana, NULL, ao2_cleanup);
|
|
|
|
RAII_VAR(struct ast_channel *, chanb, NULL, ao2_cleanup);
|
|
|
|
const char *chana_exten;
|
|
|
|
const char *chana_context;
|
|
|
|
int chana_priority;
|
|
|
|
const char *chanb_exten;
|
|
|
|
const char *chanb_context;
|
|
|
|
int chanb_priority;
|
|
|
|
struct ast_bridge *bridge;
|
2012-06-23 00:29:18 +00:00
|
|
|
char buf[256];
|
2013-06-25 22:28:22 +00:00
|
|
|
RAII_VAR(struct ast_features_xfer_config *, xfer_cfg_a, NULL, ao2_cleanup);
|
|
|
|
RAII_VAR(struct ast_features_xfer_config *, xfer_cfg_b, NULL, ao2_cleanup);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
|
|
|
/* make sure valid channels were specified */
|
|
|
|
if (ast_strlen_zero(channela) || ast_strlen_zero(channelb)) {
|
|
|
|
astman_send_error(s, m, "Missing channel parameter in request");
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
Fix two race conditions and ref counting issue when joining a bridge
These problems were all caught by a test in the Asterisk Test Suite that
originated some Local channels and attempted to move the ;2 half of the Local
channel into a bridge using the Bridge AMI action.
(1) When originating a channel, the Newchannel event is emitted quickly;
however, the ;2 channel will not have a pbx thread assigned to it until
after the outbound 'dialing' for the ;1 is complete. Thus, there is a period
of time where the outside world "knows" of the channel's existence and can
influence it but Asterisk has not yet started the dialplan execution thread.
If a Bridge AMI action is taken on the channel, the channel appears to be a
Dialed channel with no PBX thread; hence, the channel will be imparted into
the Bridge by first 'yanking' the channel. At the same time, a race condition
can occur after the yank (but before entering the bridge) when ;1 answers
and starts a PBX on the ;2. The end result currently is an assertion failure
in the Bridging API, as a channel with a PBX is imparted into the Bridge.
There's no way to prevent AMI from attempting to Bridge a channel
immediately after creation; likewise, holding the channel lock through the
entire Dial operation is unwise (and impossible). Instead of treating the
presence of a PBX thread as an error, we simply bail out of the adding the
channel to the bridge through ast_bridge_impart. The Bridge action will
then fail - but we avoid a situation where the channel is both executing
a PBX thread and simultaneously being given a separate thread in the
bridging system (which would be a "bad thing"). Since imparting a channel
with a PBX *can* occur and is not a programming error, the asserts have been
removed.
(2) When the first condition occurs, we have to take one of two actions: either
hangup the yanked channel as it did not enter the bridge, or deref it
because we don't own it. We can determine if we own it or not by testing
for the presence of the PBX thread. If we hung it up directly, we'd crash.
(3) bridge_find_channel does not increase the reference count of the
ast_bridge_channel object. The RAII_VAR usage in ast_bridge_add_channel
thus created a ticking time bomb in whatever bridge the channel moved into,
as the destructor for the ast_bridge_channel object would be called.
Review: https://reviewboard.asterisk.org/r/2741/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-12 15:59:19 +00:00
|
|
|
ast_debug(1, "Performing Bridge action on %s and %s\n", channela, channelb);
|
|
|
|
|
2011-08-16 17:23:08 +00:00
|
|
|
/* Start with chana */
|
|
|
|
chana = ast_channel_get_by_name_prefix(channela, strlen(channela));
|
|
|
|
if (!chana) {
|
2013-05-28 14:45:31 +00:00
|
|
|
snprintf(buf, sizeof(buf), "Channel1 does not exist: %s", channela);
|
2011-08-16 17:23:08 +00:00
|
|
|
astman_send_error(s, m, buf);
|
|
|
|
return 0;
|
|
|
|
}
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_channel_lock(chana);
|
2016-06-30 20:17:02 +00:00
|
|
|
xfer_cfg_a = ast_get_chan_features_xfer_config(chana);
|
2013-05-28 14:45:31 +00:00
|
|
|
chana_exten = ast_strdupa(ast_channel_exten(chana));
|
|
|
|
chana_context = ast_strdupa(ast_channel_context(chana));
|
|
|
|
chana_priority = ast_channel_priority(chana);
|
2017-04-30 21:40:16 +00:00
|
|
|
if (ast_test_flag(ast_channel_flags(chana), AST_FLAG_IN_AUTOLOOP)) {
|
2013-05-28 14:45:31 +00:00
|
|
|
chana_priority++;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_channel_unlock(chana);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
|
|
|
chanb = ast_channel_get_by_name_prefix(channelb, strlen(channelb));
|
|
|
|
if (!chanb) {
|
2013-05-28 14:45:31 +00:00
|
|
|
snprintf(buf, sizeof(buf), "Channel2 does not exist: %s", channelb);
|
2011-08-16 17:23:08 +00:00
|
|
|
astman_send_error(s, m, buf);
|
|
|
|
return 0;
|
|
|
|
}
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_channel_lock(chanb);
|
2016-06-30 20:17:02 +00:00
|
|
|
xfer_cfg_b = ast_get_chan_features_xfer_config(chanb);
|
2013-05-28 14:45:31 +00:00
|
|
|
chanb_exten = ast_strdupa(ast_channel_exten(chanb));
|
|
|
|
chanb_context = ast_strdupa(ast_channel_context(chanb));
|
|
|
|
chanb_priority = ast_channel_priority(chanb);
|
2017-04-30 21:40:16 +00:00
|
|
|
if (ast_test_flag(ast_channel_flags(chanb), AST_FLAG_IN_AUTOLOOP)) {
|
2013-05-28 14:45:31 +00:00
|
|
|
chanb_priority++;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_channel_unlock(chanb);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2013-05-28 14:45:31 +00:00
|
|
|
bridge = ast_bridge_basic_new();
|
|
|
|
if (!bridge) {
|
|
|
|
astman_send_error(s, m, "Unable to create bridge\n");
|
2012-06-23 00:29:18 +00:00
|
|
|
return 0;
|
|
|
|
}
|
2011-08-16 17:23:08 +00:00
|
|
|
|
2017-04-30 21:40:16 +00:00
|
|
|
ast_bridge_set_after_goto(chana, chana_context, chana_exten, chana_priority);
|
2013-06-25 22:28:22 +00:00
|
|
|
if (ast_bridge_add_channel(bridge, chana, NULL, playtone & PLAYTONE_CHANNEL1, xfer_cfg_a ? xfer_cfg_a->xfersound : NULL)) {
|
2013-05-28 14:45:31 +00:00
|
|
|
snprintf(buf, sizeof(buf), "Unable to add Channel1 to bridge: %s", ast_channel_name(chana));
|
|
|
|
astman_send_error(s, m, buf);
|
2013-08-22 21:09:52 +00:00
|
|
|
ast_bridge_destroy(bridge, 0);
|
2012-06-23 00:29:18 +00:00
|
|
|
return 0;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2017-04-30 21:40:16 +00:00
|
|
|
ast_bridge_set_after_goto(chanb, chanb_context, chanb_exten, chanb_priority);
|
2013-06-25 22:28:22 +00:00
|
|
|
if (ast_bridge_add_channel(bridge, chanb, NULL, playtone & PLAYTONE_CHANNEL2, xfer_cfg_b ? xfer_cfg_b->xfersound : NULL)) {
|
2013-05-28 14:45:31 +00:00
|
|
|
snprintf(buf, sizeof(buf), "Unable to add Channel2 to bridge: %s", ast_channel_name(chanb));
|
|
|
|
astman_send_error(s, m, buf);
|
2013-08-22 21:09:52 +00:00
|
|
|
ast_bridge_destroy(bridge, 0);
|
2012-06-23 00:29:18 +00:00
|
|
|
return 0;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
|
|
|
|
2013-05-28 14:45:31 +00:00
|
|
|
astman_send_ack(s, m, "Channels have been bridged");
|
2014-11-12 20:40:59 +00:00
|
|
|
ao2_cleanup(bridge);
|
2011-08-16 17:23:08 +00:00
|
|
|
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
static char *app_bridge = "Bridge";
|
|
|
|
|
|
|
|
enum {
|
|
|
|
BRIDGE_OPT_PLAYTONE = (1 << 0),
|
2009-09-24 20:29:51 +00:00
|
|
|
OPT_CALLEE_HANGUP = (1 << 1),
|
|
|
|
OPT_CALLER_HANGUP = (1 << 2),
|
|
|
|
OPT_DURATION_LIMIT = (1 << 3),
|
|
|
|
OPT_DURATION_STOP = (1 << 4),
|
|
|
|
OPT_CALLEE_TRANSFER = (1 << 5),
|
|
|
|
OPT_CALLER_TRANSFER = (1 << 6),
|
|
|
|
OPT_CALLEE_MONITOR = (1 << 7),
|
|
|
|
OPT_CALLER_MONITOR = (1 << 8),
|
|
|
|
OPT_CALLEE_PARK = (1 << 9),
|
|
|
|
OPT_CALLER_PARK = (1 << 10),
|
|
|
|
OPT_CALLEE_KILL = (1 << 11),
|
2012-03-22 21:25:22 +00:00
|
|
|
OPT_CALLEE_GO_ON = (1 << 12),
|
2009-09-24 20:29:51 +00:00
|
|
|
};
|
2012-03-22 19:51:16 +00:00
|
|
|
|
2009-09-24 20:29:51 +00:00
|
|
|
enum {
|
|
|
|
OPT_ARG_DURATION_LIMIT = 0,
|
|
|
|
OPT_ARG_DURATION_STOP,
|
2012-03-22 21:25:22 +00:00
|
|
|
OPT_ARG_CALLEE_GO_ON,
|
2009-09-24 20:29:51 +00:00
|
|
|
/* note: this entry _MUST_ be the last one in the enum */
|
|
|
|
OPT_ARG_ARRAY_SIZE,
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
};
|
|
|
|
|
|
|
|
AST_APP_OPTIONS(bridge_exec_options, BEGIN_OPTIONS
|
2009-09-24 20:29:51 +00:00
|
|
|
AST_APP_OPTION('p', BRIDGE_OPT_PLAYTONE),
|
2012-03-22 21:25:22 +00:00
|
|
|
AST_APP_OPTION_ARG('F', OPT_CALLEE_GO_ON, OPT_ARG_CALLEE_GO_ON),
|
2009-09-24 20:29:51 +00:00
|
|
|
AST_APP_OPTION('h', OPT_CALLEE_HANGUP),
|
|
|
|
AST_APP_OPTION('H', OPT_CALLER_HANGUP),
|
|
|
|
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('S', OPT_DURATION_STOP, OPT_ARG_DURATION_STOP),
|
|
|
|
AST_APP_OPTION('t', OPT_CALLEE_TRANSFER),
|
|
|
|
AST_APP_OPTION('T', OPT_CALLER_TRANSFER),
|
|
|
|
AST_APP_OPTION('w', OPT_CALLEE_MONITOR),
|
|
|
|
AST_APP_OPTION('W', OPT_CALLER_MONITOR),
|
|
|
|
AST_APP_OPTION('x', OPT_CALLEE_KILL),
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
END_OPTIONS );
|
|
|
|
|
2009-09-24 20:29:51 +00:00
|
|
|
int ast_bridge_timelimit(struct ast_channel *chan, struct ast_bridge_config *config,
|
|
|
|
char *parse, struct timeval *calldurationlimit)
|
|
|
|
{
|
|
|
|
char *stringp = ast_strdupa(parse);
|
|
|
|
char *limit_str, *warning_str, *warnfreq_str;
|
|
|
|
const char *var;
|
|
|
|
int play_to_caller = 0, play_to_callee = 0;
|
|
|
|
int delta;
|
|
|
|
|
|
|
|
limit_str = strsep(&stringp, ":");
|
|
|
|
warning_str = strsep(&stringp, ":");
|
|
|
|
warnfreq_str = strsep(&stringp, ":");
|
|
|
|
|
|
|
|
config->timelimit = atol(limit_str);
|
|
|
|
if (warning_str)
|
|
|
|
config->play_warning = atol(warning_str);
|
|
|
|
if (warnfreq_str)
|
|
|
|
config->warning_freq = atol(warnfreq_str);
|
|
|
|
|
|
|
|
if (!config->timelimit) {
|
|
|
|
ast_log(LOG_WARNING, "Bridge does not accept L(%s), hanging up.\n", limit_str);
|
|
|
|
config->timelimit = config->play_warning = config->warning_freq = 0;
|
|
|
|
config->warning_sound = NULL;
|
|
|
|
return -1; /* error */
|
|
|
|
} else if ( (delta = config->play_warning - config->timelimit) > 0) {
|
|
|
|
int w = config->warning_freq;
|
|
|
|
|
2011-08-09 23:17:13 +00:00
|
|
|
/*
|
|
|
|
* If the first warning is requested _after_ the entire call
|
|
|
|
* would end, and no warning frequency is requested, then turn
|
|
|
|
* off the warning. If a warning frequency is requested, reduce
|
|
|
|
* the 'first warning' time by that frequency until it falls
|
|
|
|
* within the call's total time limit.
|
|
|
|
*
|
|
|
|
* Graphically:
|
|
|
|
* timelim->| delta |<-playwarning
|
|
|
|
* 0__________________|_________________|
|
|
|
|
* | w | | | |
|
|
|
|
*
|
|
|
|
* so the number of intervals to cut is 1+(delta-1)/w
|
|
|
|
*/
|
2009-09-24 20:29:51 +00:00
|
|
|
if (w == 0) {
|
|
|
|
config->play_warning = 0;
|
|
|
|
} else {
|
|
|
|
config->play_warning -= w * ( 1 + (delta-1)/w );
|
|
|
|
if (config->play_warning < 1)
|
|
|
|
config->play_warning = config->warning_freq = 0;
|
|
|
|
}
|
|
|
|
}
|
2012-03-22 19:51:16 +00:00
|
|
|
|
2009-09-24 20:29:51 +00:00
|
|
|
ast_channel_lock(chan);
|
|
|
|
|
|
|
|
var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLER");
|
|
|
|
play_to_caller = var ? ast_true(var) : 1;
|
|
|
|
|
|
|
|
var = pbx_builtin_getvar_helper(chan, "LIMIT_PLAYAUDIO_CALLEE");
|
|
|
|
play_to_callee = var ? ast_true(var) : 0;
|
|
|
|
|
|
|
|
if (!play_to_caller && !play_to_callee)
|
|
|
|
play_to_caller = 1;
|
|
|
|
|
|
|
|
var = pbx_builtin_getvar_helper(chan, "LIMIT_WARNING_FILE");
|
|
|
|
config->warning_sound = !ast_strlen_zero(var) ? ast_strdup(var) : ast_strdup("timeleft");
|
|
|
|
|
|
|
|
/* The code looking at config wants a NULL, not just "", to decide
|
|
|
|
* that the message should not be played, so we replace "" with NULL.
|
|
|
|
* Note, pbx_builtin_getvar_helper _can_ return NULL if the variable is
|
|
|
|
* not found.
|
|
|
|
*/
|
|
|
|
|
|
|
|
var = pbx_builtin_getvar_helper(chan, "LIMIT_TIMEOUT_FILE");
|
|
|
|
config->end_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
|
|
|
|
|
|
|
|
var = pbx_builtin_getvar_helper(chan, "LIMIT_CONNECT_FILE");
|
|
|
|
config->start_sound = !ast_strlen_zero(var) ? ast_strdup(var) : NULL;
|
|
|
|
|
|
|
|
ast_channel_unlock(chan);
|
|
|
|
|
|
|
|
/* undo effect of S(x) in case they are both used */
|
|
|
|
calldurationlimit->tv_sec = 0;
|
|
|
|
calldurationlimit->tv_usec = 0;
|
|
|
|
|
|
|
|
/* more efficient to do it like S(x) does since no advanced opts */
|
|
|
|
if (!config->play_warning && !config->start_sound && !config->end_sound && config->timelimit) {
|
|
|
|
calldurationlimit->tv_sec = config->timelimit / 1000;
|
|
|
|
calldurationlimit->tv_usec = (config->timelimit % 1000) * 1000;
|
|
|
|
ast_verb(3, "Setting call duration limit to %.3lf seconds.\n",
|
|
|
|
calldurationlimit->tv_sec + calldurationlimit->tv_usec / 1000000.0);
|
2013-05-21 18:00:22 +00:00
|
|
|
play_to_caller = 0;
|
|
|
|
play_to_callee = 0;
|
|
|
|
config->timelimit = 0;
|
|
|
|
config->play_warning = 0;
|
|
|
|
config->warning_freq = 0;
|
2009-09-24 20:29:51 +00:00
|
|
|
} else {
|
2010-01-18 22:31:25 +00:00
|
|
|
ast_verb(4, "Limit Data for this call:\n");
|
|
|
|
ast_verb(4, "timelimit = %ld ms (%.3lf s)\n", config->timelimit, config->timelimit / 1000.0);
|
|
|
|
ast_verb(4, "play_warning = %ld ms (%.3lf s)\n", config->play_warning, config->play_warning / 1000.0);
|
2009-09-24 20:29:51 +00:00
|
|
|
ast_verb(4, "play_to_caller = %s\n", play_to_caller ? "yes" : "no");
|
|
|
|
ast_verb(4, "play_to_callee = %s\n", play_to_callee ? "yes" : "no");
|
2010-01-18 22:31:25 +00:00
|
|
|
ast_verb(4, "warning_freq = %ld ms (%.3lf s)\n", config->warning_freq, config->warning_freq / 1000.0);
|
2009-09-24 20:29:51 +00:00
|
|
|
ast_verb(4, "start_sound = %s\n", S_OR(config->start_sound, ""));
|
|
|
|
ast_verb(4, "warning_sound = %s\n", config->warning_sound);
|
|
|
|
ast_verb(4, "end_sound = %s\n", S_OR(config->end_sound, ""));
|
|
|
|
}
|
|
|
|
if (play_to_caller)
|
|
|
|
ast_set_flag(&(config->features_caller), AST_FEATURE_PLAY_WARNING);
|
|
|
|
if (play_to_callee)
|
|
|
|
ast_set_flag(&(config->features_callee), AST_FEATURE_PLAY_WARNING);
|
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
|
|
|
|
2007-08-07 23:04:01 +00:00
|
|
|
/*!
|
|
|
|
* \brief Bridge channels
|
|
|
|
* \param chan
|
2007-09-05 16:31:39 +00:00
|
|
|
* \param data channel to bridge with.
|
2012-03-22 19:51:16 +00:00
|
|
|
*
|
2007-08-07 23:04:01 +00:00
|
|
|
* Split data, check we aren't bridging with ourself, check valid channel,
|
|
|
|
* answer call if not already, check compatible channels, setup bridge config
|
2014-01-17 17:16:14 +00:00
|
|
|
* now bridge call, if transferred party hangs up return to PBX extension.
|
2011-05-20 17:04:53 +00:00
|
|
|
*/
|
2009-05-21 21:13:09 +00:00
|
|
|
static int bridge_exec(struct ast_channel *chan, const char *data)
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
{
|
2014-10-06 15:41:32 +00:00
|
|
|
struct ast_channel *current_dest_chan;
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
char *tmp_data = NULL;
|
|
|
|
struct ast_flags opts = { 0, };
|
|
|
|
struct ast_bridge_config bconfig = { { 0, }, };
|
2009-09-24 20:29:51 +00:00
|
|
|
char *opt_args[OPT_ARG_ARRAY_SIZE];
|
|
|
|
struct timeval calldurationlimit = { 0, };
|
2013-05-21 18:00:22 +00:00
|
|
|
const char *context;
|
|
|
|
const char *extension;
|
|
|
|
int priority;
|
2014-10-06 15:41:32 +00:00
|
|
|
int bridge_add_failed;
|
2013-05-28 14:45:31 +00:00
|
|
|
struct ast_bridge_features chan_features;
|
|
|
|
struct ast_bridge_features *peer_features;
|
|
|
|
struct ast_bridge *bridge;
|
2014-10-06 15:41:32 +00:00
|
|
|
struct ast_features_xfer_config *xfer_cfg;
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
|
|
|
AST_DECLARE_APP_ARGS(args,
|
|
|
|
AST_APP_ARG(dest_chan);
|
|
|
|
AST_APP_ARG(options);
|
|
|
|
);
|
2012-03-22 19:51:16 +00:00
|
|
|
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
if (ast_strlen_zero(data)) {
|
|
|
|
ast_log(LOG_WARNING, "Bridge require at least 1 argument specifying the other end of the bridge\n");
|
|
|
|
return -1;
|
|
|
|
}
|
|
|
|
|
|
|
|
tmp_data = ast_strdupa(data);
|
|
|
|
AST_STANDARD_APP_ARGS(args, tmp_data);
|
|
|
|
if (!ast_strlen_zero(args.options))
|
2009-09-24 20:29:51 +00:00
|
|
|
ast_app_parse_options(bridge_exec_options, &opts, opt_args, args.options);
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
|
|
|
/* make sure we have a valid end point */
|
2014-10-06 15:41:32 +00:00
|
|
|
current_dest_chan = ast_channel_get_by_name_prefix(args.dest_chan,
|
|
|
|
strlen(args.dest_chan));
|
|
|
|
if (!current_dest_chan) {
|
2012-06-23 00:29:18 +00:00
|
|
|
ast_log(LOG_WARNING, "Bridge failed because channel %s does not exist\n",
|
|
|
|
args.dest_chan);
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2013-05-28 14:45:31 +00:00
|
|
|
/* avoid bridge with ourselves */
|
|
|
|
if (chan == current_dest_chan) {
|
2014-10-06 15:41:32 +00:00
|
|
|
ast_channel_unref(current_dest_chan);
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_log(LOG_WARNING, "Unable to bridge channel %s with itself\n", ast_channel_name(chan));
|
2012-06-23 00:29:18 +00:00
|
|
|
return 0;
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
}
|
|
|
|
|
2012-06-23 00:29:18 +00:00
|
|
|
if (ast_test_flag(&opts, OPT_DURATION_LIMIT)
|
|
|
|
&& !ast_strlen_zero(opt_args[OPT_ARG_DURATION_LIMIT])
|
|
|
|
&& ast_bridge_timelimit(chan, &bconfig, opt_args[OPT_ARG_DURATION_LIMIT], &calldurationlimit)) {
|
|
|
|
pbx_builtin_setvar_helper(chan, "BRIDGERESULT", "FAILURE");
|
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
|
2009-09-24 20:29:51 +00:00
|
|
|
if (ast_test_flag(&opts, OPT_CALLEE_TRANSFER))
|
|
|
|
ast_set_flag(&(bconfig.features_callee), AST_FEATURE_REDIRECT);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLER_TRANSFER))
|
|
|
|
ast_set_flag(&(bconfig.features_caller), AST_FEATURE_REDIRECT);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLEE_HANGUP))
|
|
|
|
ast_set_flag(&(bconfig.features_callee), AST_FEATURE_DISCONNECT);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLER_HANGUP))
|
|
|
|
ast_set_flag(&(bconfig.features_caller), AST_FEATURE_DISCONNECT);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLEE_MONITOR))
|
|
|
|
ast_set_flag(&(bconfig.features_callee), AST_FEATURE_AUTOMON);
|
2011-02-04 16:55:39 +00:00
|
|
|
if (ast_test_flag(&opts, OPT_CALLER_MONITOR))
|
2009-09-24 20:29:51 +00:00
|
|
|
ast_set_flag(&(bconfig.features_caller), AST_FEATURE_AUTOMON);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLEE_PARK))
|
|
|
|
ast_set_flag(&(bconfig.features_callee), AST_FEATURE_PARKCALL);
|
|
|
|
if (ast_test_flag(&opts, OPT_CALLER_PARK))
|
|
|
|
ast_set_flag(&(bconfig.features_caller), AST_FEATURE_PARKCALL);
|
Convert the ast_channel data structure over to the astobj2 framework.
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24 14:04:26 +00:00
|
|
|
|
2013-05-21 18:00:22 +00:00
|
|
|
/* Setup after bridge goto location. */
|
|
|
|
if (ast_test_flag(&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);
|
2013-07-25 02:20:23 +00:00
|
|
|
ast_bridge_set_after_go_on(current_dest_chan, context, extension, priority,
|
2013-05-21 18:00:22 +00:00
|
|
|
opt_args[OPT_ARG_CALLEE_GO_ON]);
|
|
|
|
} else if (!ast_test_flag(&opts, OPT_CALLEE_KILL)) {
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_channel_lock(current_dest_chan);
|
|
|
|
context = ast_strdupa(ast_channel_context(current_dest_chan));
|
|
|
|
extension = ast_strdupa(ast_channel_exten(current_dest_chan));
|
|
|
|
priority = ast_channel_priority(current_dest_chan);
|
|
|
|
ast_channel_unlock(current_dest_chan);
|
2015-01-09 21:45:10 +00:00
|
|
|
ast_bridge_set_after_go_on(current_dest_chan, context, extension, priority, NULL);
|
2013-05-28 14:45:31 +00:00
|
|
|
}
|
|
|
|
|
|
|
|
if (ast_bridge_features_init(&chan_features)) {
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
goto done;
|
2013-05-21 18:00:22 +00:00
|
|
|
}
|
|
|
|
|
2013-05-28 14:45:31 +00:00
|
|
|
peer_features = ast_bridge_features_new();
|
|
|
|
if (!peer_features) {
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
|
|
|
|
if (pre_bridge_setup(chan, current_dest_chan, &bconfig, &chan_features, peer_features)) {
|
|
|
|
ast_bridge_features_destroy(peer_features);
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
|
|
|
|
bridge = ast_bridge_basic_new();
|
|
|
|
if (!bridge) {
|
|
|
|
ast_bridge_features_destroy(peer_features);
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
|
2016-06-30 20:17:02 +00:00
|
|
|
ast_channel_lock(current_dest_chan);
|
2013-06-25 22:28:22 +00:00
|
|
|
xfer_cfg = ast_get_chan_features_xfer_config(current_dest_chan);
|
2016-06-30 20:17:02 +00:00
|
|
|
ast_channel_unlock(current_dest_chan);
|
2014-10-06 15:41:32 +00:00
|
|
|
bridge_add_failed = ast_bridge_add_channel(bridge, current_dest_chan, peer_features,
|
|
|
|
ast_test_flag(&opts, BRIDGE_OPT_PLAYTONE),
|
|
|
|
xfer_cfg ? xfer_cfg->xfersound : NULL);
|
|
|
|
ao2_cleanup(xfer_cfg);
|
|
|
|
if (bridge_add_failed) {
|
2013-05-28 14:45:31 +00:00
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
2013-08-22 21:09:52 +00:00
|
|
|
ast_bridge_destroy(bridge, 0);
|
2013-05-28 14:45:31 +00:00
|
|
|
goto done;
|
|
|
|
}
|
|
|
|
|
2014-10-06 15:41:32 +00:00
|
|
|
/* Don't keep the channel ref in case it was not already in a bridge. */
|
|
|
|
current_dest_chan = ast_channel_unref(current_dest_chan);
|
|
|
|
|
2013-09-13 22:19:23 +00:00
|
|
|
ast_bridge_join(bridge, chan, NULL, &chan_features, NULL,
|
|
|
|
AST_BRIDGE_JOIN_PASS_REFERENCE);
|
2013-05-28 14:45:31 +00:00
|
|
|
|
|
|
|
ast_bridge_features_cleanup(&chan_features);
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
2012-06-23 00:29:18 +00:00
|
|
|
/* The bridge has ended, set BRIDGERESULT to SUCCESS. */
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
pbx_builtin_setvar_helper(chan, "BRIDGERESULT", "SUCCESS");
|
2009-09-24 20:29:51 +00:00
|
|
|
done:
|
2012-06-23 00:29:18 +00:00
|
|
|
ast_free((char *) bconfig.warning_sound);
|
|
|
|
ast_free((char *) bconfig.end_sound);
|
|
|
|
ast_free((char *) bconfig.start_sound);
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
2014-10-06 15:41:32 +00:00
|
|
|
ast_channel_cleanup(current_dest_chan);
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
return 0;
|
|
|
|
}
|
|
|
|
|
2013-08-16 17:33:21 +00:00
|
|
|
/*!
|
|
|
|
* \internal
|
|
|
|
* \brief Clean up resources on Asterisk shutdown
|
|
|
|
*/
|
2012-10-02 01:47:16 +00:00
|
|
|
static void features_shutdown(void)
|
|
|
|
{
|
2013-06-06 21:40:35 +00:00
|
|
|
ast_features_config_shutdown();
|
|
|
|
|
2012-10-02 01:47:16 +00:00
|
|
|
ast_manager_unregister("Bridge");
|
2013-05-21 18:00:22 +00:00
|
|
|
|
2012-10-02 01:47:16 +00:00
|
|
|
ast_unregister_application(app_bridge);
|
|
|
|
|
|
|
|
}
|
|
|
|
|
2011-08-16 17:23:08 +00:00
|
|
|
int ast_features_init(void)
|
|
|
|
{
|
|
|
|
int res;
|
Merge changes from team/russell/issue_5841:
This patch adds a "Bridge" Manager action, as well as a "Bridge" dialplan
application. The manager action will allow you to steal two active channels
in the system and bridge them together. Then, the one that did not hang up
will continue in the dialplan. Using the application will bridge the calling
channel to an arbitrary channel in the system. Whichever channel does not
hang up here will continue in the dialplan, as well.
This patch has been touched by a bunch of people over the course of a couple
years. Please forgive me if I have missed your name in the history of things.
The most recent patch came from issue #5841, but there is also a reference to
an earlier version of this patch from issue #4297. The people involved in writing
and/or reviewing the code include at least: twisted, mflorrel, heath1444, davetroy,
tim_ringenbach, moy, tmancill, serge-v, and me. There are also positive test
reports from many people.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61281 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 20:44:44 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
res = ast_features_config_init();
|
2011-08-16 17:23:08 +00:00
|
|
|
if (res) {
|
2005-01-10 04:03:30 +00:00
|
|
|
return res;
|
2011-08-16 17:23:08 +00:00
|
|
|
}
|
2013-06-06 21:40:35 +00:00
|
|
|
res |= ast_register_application2(app_bridge, bridge_exec, NULL, NULL, NULL);
|
|
|
|
res |= ast_manager_register_xml_core("Bridge", EVENT_FLAG_CALL, action_bridge);
|
METERMAIDS:
-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-06-26 16:43:21 +00:00
|
|
|
|
2013-06-06 21:40:35 +00:00
|
|
|
if (res) {
|
|
|
|
features_shutdown();
|
|
|
|
} else {
|
2015-03-26 22:24:26 +00:00
|
|
|
ast_register_cleanup(features_shutdown);
|
2013-06-06 21:40:35 +00:00
|
|
|
}
|
2012-10-02 01:47:16 +00:00
|
|
|
|
2001-12-27 11:07:33 +00:00
|
|
|
return res;
|
|
|
|
}
|