asterisk/apps/app_mixmonitor.c

1801 lines
58 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2005, Anthony Minessale II
* Copyright (C) 2005 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
* Kevin P. Fleming <kpfleming@digium.com>
*
* Based on app_muxmon.c provided by
* Anthony Minessale II <anthmct@yahoo.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief MixMonitor() - Record a call and mix the audio during the recording
* \ingroup applications
*
* \author Mark Spencer <markster@digium.com>
* \author Kevin P. Fleming <kpfleming@digium.com>
*
* \note Based on app_muxmon.c provided by
* Anthony Minessale II <anthmct@yahoo.com>
*/
/*** MODULEINFO
<use type="module">func_periodic_hook</use>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/paths.h" /* use ast_config_AST_MONITOR_DIR */
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
#include "asterisk/stringfields.h"
#include "asterisk/file.h"
#include "asterisk/audiohook.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/channel.h"
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
#include "asterisk/autochan.h"
#include "asterisk/manager.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
#include "asterisk/callerid.h"
#include "asterisk/mod_format.h"
#include "asterisk/linkedlists.h"
#include "asterisk/test.h"
#include "asterisk/mixmonitor.h"
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
#include "asterisk/format_cache.h"
#include "asterisk/beep.h"
/*** DOCUMENTATION
<application name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</synopsis>
<syntax>
<parameter name="file" required="true" argsep=".">
<argument name="filename" required="true">
<para>If <replaceable>filename</replaceable> is an absolute path, uses that path, otherwise
creates the file in the configured monitoring directory from <filename>asterisk.conf.</filename></para>
</argument>
<argument name="extension" required="true" />
</parameter>
<parameter name="options">
<optionlist>
<option name="a">
<para>Append to the file instead of overwriting it.</para>
</option>
<option name="b">
<para>Only save audio to the file while the channel is bridged.</para>
<note><para>If you utilize this option inside a Local channel, you must make sure the Local
channel is not optimized away. To do this, be sure to call your Local channel with the
<literal>/n</literal> option. For example: Dial(Local/start@mycontext/n)</para></note>
</option>
<option name="B">
<para>Play a periodic beep while this call is being recorded.</para>
<argument name="interval"><para>Interval, in seconds. Default is 15.</para></argument>
</option>
<option name="c">
<para>Use the real Caller ID from the channel for the voicemail Caller ID.</para>
<para>By default, the Connected Line is used. If you want the channel caller's
real number, you may need to specify this option.</para>
</option>
<option name="d">
<para>Delete the recording file as soon as MixMonitor is done with it.</para>
<para>For example, if you use the m option to dispatch the recording to a voicemail box,
you can specify this option to delete the original copy of it afterwards.</para>
</option>
<option name="v">
<para>Adjust the <emphasis>heard</emphasis> volume by a factor of <replaceable>x</replaceable>
(range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="V">
<para>Adjust the <emphasis>spoken</emphasis> volume by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="W">
<para>Adjust both, <emphasis>heard and spoken</emphasis> volumes by a factor
of <replaceable>x</replaceable> (range <literal>-4</literal> to <literal>4</literal>)</para>
<argument name="x" required="true" />
</option>
<option name="r">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>receive</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="t">
<argument name="file" required="true" />
<para>Use the specified file to record the <emphasis>transmit</emphasis> audio feed.
Like with the basic filename argument, if an absolute path isn't given, it will create
the file in the configured monitoring directory.</para>
</option>
<option name="n">
<para>When the <replaceable>r</replaceable> or <replaceable>t</replaceable> option is
used, MixMonitor will insert silence into the specified files to maintain
synchronization between them. Use this option to disable that behavior.</para>
</option>
<option name="i">
<argument name="chanvar" required="true" />
<para>Stores the MixMonitor's ID on this channel variable.</para>
</option>
<option name="p">
<para>Play a beep on the channel that starts the recording.</para>
</option>
<option name="P">
<para>Play a beep on the channel that stops the recording.</para>
</option>
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
<option name="m">
<argument name="mailbox" required="true" />
<para>Create a copy of the recording as a voicemail in the indicated <emphasis>mailbox</emphasis>(es)
separated by commas eg. m(1111@default,2222@default,...). Folders can be optionally specified using
the syntax: mailbox@context/folder</para>
</option>
</optionlist>
</parameter>
<parameter name="command">
<para>Will be executed when the recording is over.</para>
<para>Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.</para>
<para>All variables will be evaluated at the time MixMonitor is called.</para>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para></warning>
</parameter>
</syntax>
<description>
<para>Records the audio on the current channel to the specified file.</para>
<para>This application does not automatically answer and should be preceeded by
an application such as Answer or Progress().</para>
<note><para>MixMonitor runs as an audiohook.</para></note>
<note><para>If a filename passed to MixMonitor ends with
<literal>.wav49</literal>, Asterisk will silently convert the extension to
<literal>.WAV</literal> for legacy reasons. <variable>MIXMONITOR_FILENAME</variable>
will contain the actual filename that Asterisk is writing to, not necessarily the
value that was passed in.</para></note>
<variablelist>
<variable name="MIXMONITOR_FILENAME">
<para>Will contain the filename used to record.</para>
</variable>
</variablelist>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of ANY of the application's
parameters. You risk a command injection attack executing arbitrary commands
if the untrusted strings aren't filtered to remove dangerous characters. See
function <variable>FILTER()</variable>.</para></warning>
</description>
<see-also>
<ref type="application">StopMixMonitor</ref>
<ref type="function">AUDIOHOOK_INHERIT</ref>
</see-also>
</application>
<application name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor, and free the recording's file handle.
</synopsis>
<syntax>
<parameter name="MixMonitorID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
MixMonitor.</para>
</parameter>
</syntax>
<description>
<para>Stops the audio recording that was started with a call to <literal>MixMonitor()</literal>
on the current channel.</para>
</description>
<see-also>
<ref type="application">MixMonitor</ref>
</see-also>
</application>
<manager name="MixMonitorMute" language="en_US">
<synopsis>
Mute / unMute a Mixmonitor recording.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to mute.</para>
</parameter>
<parameter name="Direction">
<para>Which part of the recording to mute: read, write or both (from channel, to channel or both channels).</para>
</parameter>
<parameter name="State">
<para>Turn mute on or off : 1 to turn on, 0 to turn off.</para>
</parameter>
</syntax>
<description>
<para>This action may be used to mute a MixMonitor recording.</para>
</description>
</manager>
<manager name="MixMonitor" language="en_US">
<synopsis>
Record a call and mix the audio during the recording. Use of StopMixMonitor is required
to guarantee the audio file is available for processing during dialplan execution.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Used to specify the channel to record.</para>
</parameter>
<parameter name="File">
<para>Is the name of the file created in the monitor spool directory.
Defaults to the same name as the channel (with slashes replaced with dashes).
This argument is optional if you specify to record unidirectional audio with
either the r(filename) or t(filename) options in the options field. If
neither MIXMONITOR_FILENAME or this parameter is set, the mixed stream won't
be recorded.</para>
</parameter>
<parameter name="options">
<para>Options that apply to the MixMonitor in the same way as they
would apply if invoked from the MixMonitor application. For a list of
available options, see the documentation for the mixmonitor application. </para>
</parameter>
<parameter name="Command">
<para>Will be executed when the recording is over.
Any strings matching <literal>^{X}</literal> will be unescaped to <variable>X</variable>.
All variables will be evaluated at the time MixMonitor is called.</para>
<warning><para>Do not use untrusted strings such as <variable>CALLERID(num)</variable>
or <variable>CALLERID(name)</variable> as part of the command parameters. You
risk a command injection attack executing arbitrary commands if the untrusted
strings aren't filtered to remove dangerous characters. See function
<variable>FILTER()</variable>.</para></warning>
</parameter>
</syntax>
<description>
<para>This action records the audio on the current channel to the specified file.</para>
<variablelist>
<variable name="MIXMONITOR_FILENAME">
<para>Will contain the filename used to record the mixed stream.</para>
</variable>
</variablelist>
</description>
</manager>
<manager name="StopMixMonitor" language="en_US">
<synopsis>
Stop recording a call through MixMonitor, and free the recording's file handle.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The name of the channel monitored.</para>
</parameter>
<parameter name="MixMonitorID" required="false">
<para>If a valid ID is provided, then this command will stop only that specific
MixMonitor.</para>
</parameter>
</syntax>
<description>
<para>This action stops the audio recording that was started with the <literal>MixMonitor</literal>
action on the current channel.</para>
</description>
</manager>
<function name="MIXMONITOR" language="en_US">
<synopsis>
Retrieve data pertaining to specific instances of MixMonitor on a channel.
</synopsis>
<syntax>
<parameter name="id" required="true">
<para>The unique ID of the MixMonitor instance. The unique ID can be retrieved through the channel
variable used as an argument to the <replaceable>i</replaceable> option to MixMonitor.</para>
</parameter>
<parameter name="key" required="true">
<para>The piece of data to retrieve from the MixMonitor.</para>
<enumlist>
<enum name="filename" />
</enumlist>
</parameter>
</syntax>
</function>
<managerEvent language="en_US" name="MixMonitorStart">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when monitoring has started on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MixMonitorStop</ref>
<ref type="application">MixMonitor</ref>
<ref type="manager">MixMonitor</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MixMonitorStop">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when monitoring has stopped on a channel.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">MixMonitorStart</ref>
<ref type="application">StopMixMonitor</ref>
<ref type="manager">StopMixMonitor</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="MixMonitorMute">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when monitoring is muted or unmuted on a channel.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Direction">
<para>Which part of the recording was muted or unmuted: read, write or both
(from channel, to channel or both directions).</para>
</parameter>
<parameter name="State">
<para>If the monitoring was muted or unmuted: 1 when muted, 0 when unmuted.</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">MixMonitorMute</ref>
</see-also>
</managerEventInstance>
</managerEvent>
***/
#define get_volfactor(x) x ? ((x > 0) ? (1 << x) : ((1 << abs(x)) * -1)) : 0
static const char * const app = "MixMonitor";
static const char * const stop_app = "StopMixMonitor";
static const char * const mixmonitor_spy_type = "MixMonitor";
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \internal
* \brief This struct is a list item holds data needed to find a vm_recipient within voicemail
*/
struct vm_recipient {
char mailbox[AST_MAX_CONTEXT];
char context[AST_MAX_EXTENSION];
char folder[80];
AST_LIST_ENTRY(vm_recipient) list;
};
struct mixmonitor {
struct ast_audiohook audiohook;
char *filename;
char *filename_read;
char *filename_write;
char *post_process;
char *name;
ast_callid callid;
unsigned int flags;
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
struct ast_autochan *autochan;
struct mixmonitor_ds *mixmonitor_ds;
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/* the below string fields describe data used for creating voicemails from the recording */
AST_DECLARE_STRING_FIELDS(
AST_STRING_FIELD(call_context);
AST_STRING_FIELD(call_extension);
AST_STRING_FIELD(call_callerchan);
AST_STRING_FIELD(call_callerid);
);
int call_priority;
/* FUTURE DEVELOPMENT NOTICE
* recipient_list will need locks if we make it editable after the monitor is started */
AST_LIST_HEAD_NOLOCK(, vm_recipient) recipient_list;
};
enum mixmonitor_flags {
MUXFLAG_APPEND = (1 << 1),
MUXFLAG_BRIDGED = (1 << 2),
MUXFLAG_VOLUME = (1 << 3),
MUXFLAG_READVOLUME = (1 << 4),
MUXFLAG_WRITEVOLUME = (1 << 5),
MUXFLAG_READ = (1 << 6),
MUXFLAG_WRITE = (1 << 7),
MUXFLAG_COMBINED = (1 << 8),
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
MUXFLAG_UID = (1 << 9),
MUXFLAG_VMRECIPIENTS = (1 << 10),
MUXFLAG_BEEP = (1 << 11),
MUXFLAG_BEEP_START = (1 << 12),
MUXFLAG_BEEP_STOP = (1 << 13),
MUXFLAG_DEPRECATED_RWSYNC = (1 << 14),
MUXFLAG_NO_RWSYNC = (1 << 15),
MUXFLAG_AUTO_DELETE = (1 << 16),
MUXFLAG_REAL_CALLERID = (1 << 17),
};
enum mixmonitor_args {
OPT_ARG_READVOLUME = 0,
OPT_ARG_WRITEVOLUME,
OPT_ARG_VOLUME,
OPT_ARG_WRITENAME,
OPT_ARG_READNAME,
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
OPT_ARG_UID,
OPT_ARG_VMRECIPIENTS,
OPT_ARG_BEEP_INTERVAL,
OPT_ARG_DEPRECATED_RWSYNC,
OPT_ARG_NO_RWSYNC,
OPT_ARG_ARRAY_SIZE, /* Always last element of the enum */
};
AST_APP_OPTIONS(mixmonitor_opts, {
AST_APP_OPTION('a', MUXFLAG_APPEND),
AST_APP_OPTION('b', MUXFLAG_BRIDGED),
AST_APP_OPTION_ARG('B', MUXFLAG_BEEP, OPT_ARG_BEEP_INTERVAL),
AST_APP_OPTION('c', MUXFLAG_REAL_CALLERID),
AST_APP_OPTION('d', MUXFLAG_AUTO_DELETE),
AST_APP_OPTION('p', MUXFLAG_BEEP_START),
AST_APP_OPTION('P', MUXFLAG_BEEP_STOP),
AST_APP_OPTION_ARG('v', MUXFLAG_READVOLUME, OPT_ARG_READVOLUME),
AST_APP_OPTION_ARG('V', MUXFLAG_WRITEVOLUME, OPT_ARG_WRITEVOLUME),
AST_APP_OPTION_ARG('W', MUXFLAG_VOLUME, OPT_ARG_VOLUME),
AST_APP_OPTION_ARG('r', MUXFLAG_READ, OPT_ARG_READNAME),
AST_APP_OPTION_ARG('t', MUXFLAG_WRITE, OPT_ARG_WRITENAME),
AST_APP_OPTION_ARG('i', MUXFLAG_UID, OPT_ARG_UID),
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
AST_APP_OPTION_ARG('m', MUXFLAG_VMRECIPIENTS, OPT_ARG_VMRECIPIENTS),
AST_APP_OPTION_ARG('S', MUXFLAG_DEPRECATED_RWSYNC, OPT_ARG_DEPRECATED_RWSYNC),
AST_APP_OPTION_ARG('n', MUXFLAG_NO_RWSYNC, OPT_ARG_NO_RWSYNC),
});
struct mixmonitor_ds {
unsigned int destruction_ok;
ast_cond_t destruction_condition;
ast_mutex_t lock;
/* The filestream is held in the datastore so it can be stopped
* immediately during stop_mixmonitor or channel destruction. */
int fs_quit;
struct ast_filestream *fs;
struct ast_filestream *fs_read;
struct ast_filestream *fs_write;
struct ast_audiohook *audiohook;
unsigned int samp_rate;
char *filename;
char *beep_id;
};
/*!
* \internal
* \pre mixmonitor_ds must be locked before calling this function
*/
static void mixmonitor_ds_close_fs(struct mixmonitor_ds *mixmonitor_ds)
{
unsigned char quitting = 0;
if (mixmonitor_ds->fs) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs);
mixmonitor_ds->fs = NULL;
ast_verb(2, "MixMonitor close filestream (mixed)\n");
}
if (mixmonitor_ds->fs_read) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs_read);
mixmonitor_ds->fs_read = NULL;
ast_verb(2, "MixMonitor close filestream (read)\n");
}
if (mixmonitor_ds->fs_write) {
quitting = 1;
ast_closestream(mixmonitor_ds->fs_write);
mixmonitor_ds->fs_write = NULL;
ast_verb(2, "MixMonitor close filestream (write)\n");
}
if (quitting) {
mixmonitor_ds->fs_quit = 1;
}
}
static void mixmonitor_ds_destroy(void *data)
{
struct mixmonitor_ds *mixmonitor_ds = data;
ast_mutex_lock(&mixmonitor_ds->lock);
mixmonitor_ds->audiohook = NULL;
mixmonitor_ds->destruction_ok = 1;
ast_free(mixmonitor_ds->filename);
ast_free(mixmonitor_ds->beep_id);
ast_cond_signal(&mixmonitor_ds->destruction_condition);
ast_mutex_unlock(&mixmonitor_ds->lock);
}
static const struct ast_datastore_info mixmonitor_ds_info = {
.type = "mixmonitor",
.destroy = mixmonitor_ds_destroy,
};
static void destroy_monitor_audiohook(struct mixmonitor *mixmonitor)
{
if (mixmonitor->mixmonitor_ds) {
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
mixmonitor->mixmonitor_ds->audiohook = NULL;
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
}
/* kill the audiohook.*/
ast_audiohook_lock(&mixmonitor->audiohook);
ast_audiohook_detach(&mixmonitor->audiohook);
ast_audiohook_unlock(&mixmonitor->audiohook);
ast_audiohook_destroy(&mixmonitor->audiohook);
}
static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
if (!chan) {
return -1;
}
return ast_audiohook_attach(chan, audiohook);
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \internal
* \brief adds recipients to a mixmonitor's recipient list
* \param mixmonitor mixmonitor being affected
* \param vm_recipients string containing the desired recipients to add
*/
static void add_vm_recipients_from_string(struct mixmonitor *mixmonitor, const char *vm_recipients)
{
/* recipients are in a single string with a format format resembling "mailbox@context/INBOX,mailbox2@context2,mailbox3@context3/Work" */
char *cur_mailbox = ast_strdupa(vm_recipients);
char *cur_context;
char *cur_folder;
char *next;
int elements_processed = 0;
while (!ast_strlen_zero(cur_mailbox)) {
ast_debug(3, "attempting to add next element %d from %s\n", elements_processed, cur_mailbox);
if ((next = strchr(cur_mailbox, ',')) || (next = strchr(cur_mailbox, '&'))) {
*(next++) = '\0';
}
if ((cur_folder = strchr(cur_mailbox, '/'))) {
*(cur_folder++) = '\0';
} else {
cur_folder = "INBOX";
}
if ((cur_context = strchr(cur_mailbox, '@'))) {
*(cur_context++) = '\0';
} else {
cur_context = "default";
}
if (!ast_strlen_zero(cur_mailbox) && !ast_strlen_zero(cur_context)) {
struct vm_recipient *recipient;
if (!(recipient = ast_malloc(sizeof(*recipient)))) {
ast_log(LOG_ERROR, "Failed to allocate recipient. Aborting function.\n");
return;
}
ast_copy_string(recipient->context, cur_context, sizeof(recipient->context));
ast_copy_string(recipient->mailbox, cur_mailbox, sizeof(recipient->mailbox));
ast_copy_string(recipient->folder, cur_folder, sizeof(recipient->folder));
/* Add to list */
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
ast_verb(4, "Adding %s@%s to recipient list\n", recipient->mailbox, recipient->context);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
AST_LIST_INSERT_HEAD(&mixmonitor->recipient_list, recipient, list);
} else {
ast_log(LOG_ERROR, "Failed to properly parse extension and/or context from element %d of recipient string: %s\n", elements_processed, vm_recipients);
}
cur_mailbox = next;
elements_processed++;
}
}
static void clear_mixmonitor_recipient_list(struct mixmonitor *mixmonitor)
{
struct vm_recipient *current;
while ((current = AST_LIST_REMOVE_HEAD(&mixmonitor->recipient_list, list))) {
/* Clear list element data */
ast_free(current);
}
}
#define SAMPLES_PER_FRAME 160
static void mixmonitor_free(struct mixmonitor *mixmonitor)
{
if (mixmonitor) {
if (mixmonitor->mixmonitor_ds) {
ast_mutex_destroy(&mixmonitor->mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor->mixmonitor_ds->destruction_condition);
ast_free(mixmonitor->mixmonitor_ds);
}
ast_free(mixmonitor->name);
ast_free(mixmonitor->post_process);
ast_free(mixmonitor->filename);
ast_free(mixmonitor->filename_write);
ast_free(mixmonitor->filename_read);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/* Free everything in the recipient list */
clear_mixmonitor_recipient_list(mixmonitor);
/* clean stringfields */
ast_string_field_free_memory(mixmonitor);
ast_free(mixmonitor);
}
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/*!
* \internal
* \brief Copies the mixmonitor to all voicemail recipients
* \param mixmonitor The mixmonitor that needs to forward its file to recipients
* \param ext Format of the file that was saved
* \param filename
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
*/
static void copy_to_voicemail(struct mixmonitor *mixmonitor, const char *ext, const char *filename)
{
struct vm_recipient *recipient = NULL;
struct ast_vm_recording_data recording_data;
if (ast_string_field_init(&recording_data, 512)) {
ast_log(LOG_ERROR, "Failed to string_field_init, skipping copy_to_voicemail\n");
return;
}
/* Copy strings to stringfields that will be used for all recipients */
ast_string_field_set(&recording_data, recording_file, filename);
ast_string_field_set(&recording_data, recording_ext, ext);
ast_string_field_set(&recording_data, call_context, mixmonitor->call_context);
ast_string_field_set(&recording_data, call_extension, mixmonitor->call_extension);
ast_string_field_set(&recording_data, call_callerchan, mixmonitor->call_callerchan);
ast_string_field_set(&recording_data, call_callerid, mixmonitor->call_callerid);
/* and call_priority gets copied too */
recording_data.call_priority = mixmonitor->call_priority;
AST_LIST_TRAVERSE(&mixmonitor->recipient_list, recipient, list) {
/* context, mailbox, and folder need to be set per recipient */
ast_string_field_set(&recording_data, context, recipient->context);
ast_string_field_set(&recording_data, mailbox, recipient->mailbox);
ast_string_field_set(&recording_data, folder, recipient->folder);
ast_verb(4, "MixMonitor attempting to send voicemail copy to %s@%s\n", recording_data.mailbox,
recording_data.context);
ast_app_copy_recording_to_vm(&recording_data);
}
/* Free the string fields for recording_data before exiting the function. */
ast_string_field_free_memory(&recording_data);
}
static void mixmonitor_save_prep(struct mixmonitor *mixmonitor, char *filename, struct ast_filestream **fs, unsigned int *oflags, int *errflag, char **ext)
{
/* Initialize the file if not already done so */
char *last_slash = NULL;
if (!ast_strlen_zero(filename)) {
if (!*fs && !*errflag && !mixmonitor->mixmonitor_ds->fs_quit) {
*oflags = O_CREAT | O_WRONLY;
*oflags |= ast_test_flag(mixmonitor, MUXFLAG_APPEND) ? O_APPEND : O_TRUNC;
last_slash = strrchr(filename, '/');
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if ((*ext = strrchr(filename, '.')) && (*ext > last_slash)) {
**ext = '\0';
*ext = *ext + 1;
} else {
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
*ext = "raw";
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (!(*fs = ast_writefile(filename, *ext, NULL, *oflags, 0, 0666))) {
ast_log(LOG_ERROR, "Cannot open %s.%s\n", filename, *ext);
*errflag = 1;
} else {
struct ast_filestream *tmp = *fs;
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
mixmonitor->mixmonitor_ds->samp_rate = MAX(mixmonitor->mixmonitor_ds->samp_rate, ast_format_get_sample_rate(tmp->fmt->format));
}
}
}
}
static int mixmonitor_autochan_is_bridged(struct ast_autochan *autochan)
{
int is_bridged;
ast_autochan_channel_lock(autochan);
is_bridged = ast_channel_is_bridged(autochan->chan);
ast_autochan_channel_unlock(autochan);
return is_bridged;
}
static void *mixmonitor_thread(void *obj)
{
struct mixmonitor *mixmonitor = obj;
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
char *fs_ext = "";
char *fs_read_ext = "";
char *fs_write_ext = "";
struct ast_filestream **fs = NULL;
struct ast_filestream **fs_read = NULL;
struct ast_filestream **fs_write = NULL;
unsigned int oflags;
int errflag = 0;
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
struct ast_format *format_slin;
/* Keep callid association before any log messages */
if (mixmonitor->callid) {
ast_callid_threadassoc_add(mixmonitor->callid);
}
ast_verb(2, "Begin MixMonitor Recording %s\n", mixmonitor->name);
fs = &mixmonitor->mixmonitor_ds->fs;
fs_read = &mixmonitor->mixmonitor_ds->fs_read;
fs_write = &mixmonitor->mixmonitor_ds->fs_write;
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
mixmonitor_save_prep(mixmonitor, mixmonitor->filename, fs, &oflags, &errflag, &fs_ext);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_read, fs_read, &oflags, &errflag, &fs_read_ext);
mixmonitor_save_prep(mixmonitor, mixmonitor->filename_write, fs_write, &oflags, &errflag, &fs_write_ext);
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
format_slin = ast_format_cache_get_slin_by_rate(mixmonitor->mixmonitor_ds->samp_rate);
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* The audiohook must enter and exit the loop locked */
ast_audiohook_lock(&mixmonitor->audiohook);
while (mixmonitor->audiohook.status == AST_AUDIOHOOK_STATUS_RUNNING && !mixmonitor->mixmonitor_ds->fs_quit) {
struct ast_frame *fr = NULL;
struct ast_frame *fr_read = NULL;
struct ast_frame *fr_write = NULL;
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
if (!(fr = ast_audiohook_read_frame_all(&mixmonitor->audiohook, SAMPLES_PER_FRAME, format_slin,
&fr_read, &fr_write))) {
ast_audiohook_trigger_wait(&mixmonitor->audiohook);
if (mixmonitor->audiohook.status != AST_AUDIOHOOK_STATUS_RUNNING) {
break;
}
continue;
}
/* audiohook lock is not required for the next block.
* Unlock it, but remember to lock it before looping or exiting */
ast_audiohook_unlock(&mixmonitor->audiohook);
if (!ast_test_flag(mixmonitor, MUXFLAG_BRIDGED)
|| mixmonitor_autochan_is_bridged(mixmonitor->autochan)) {
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
/* Write out the frame(s) */
if ((*fs_read) && (fr_read)) {
struct ast_frame *cur;
for (cur = fr_read; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs_read, cur);
}
}
if ((*fs_write) && (fr_write)) {
struct ast_frame *cur;
for (cur = fr_write; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs_write, cur);
}
}
if ((*fs) && (fr)) {
struct ast_frame *cur;
for (cur = fr; cur; cur = AST_LIST_NEXT(cur, frame_list)) {
ast_writestream(*fs, cur);
}
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
}
/* All done! free it. */
if (fr) {
ast_frame_free(fr, 0);
}
if (fr_read) {
ast_frame_free(fr_read, 0);
}
if (fr_write) {
ast_frame_free(fr_write, 0);
}
fr = NULL;
fr_write = NULL;
fr_read = NULL;
ast_audiohook_lock(&mixmonitor->audiohook);
}
ast_audiohook_unlock(&mixmonitor->audiohook);
if (ast_test_flag(mixmonitor, MUXFLAG_BEEP_STOP)) {
ast_autochan_channel_lock(mixmonitor->autochan);
ast_stream_and_wait(mixmonitor->autochan->chan, "beep", "");
ast_autochan_channel_unlock(mixmonitor->autochan);
}
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
ast_autochan_destroy(mixmonitor->autochan);
/* Datastore cleanup. close the filestream and wait for ds destruction */
ast_mutex_lock(&mixmonitor->mixmonitor_ds->lock);
mixmonitor_ds_close_fs(mixmonitor->mixmonitor_ds);
if (!mixmonitor->mixmonitor_ds->destruction_ok) {
ast_cond_wait(&mixmonitor->mixmonitor_ds->destruction_condition, &mixmonitor->mixmonitor_ds->lock);
}
ast_mutex_unlock(&mixmonitor->mixmonitor_ds->lock);
/* kill the audiohook */
destroy_monitor_audiohook(mixmonitor);
if (mixmonitor->post_process) {
ast_verb(2, "Executing [%s]\n", mixmonitor->post_process);
ast_safe_system(mixmonitor->post_process);
}
ast_verb(2, "End MixMonitor Recording %s\n", mixmonitor->name);
ast_test_suite_event_notify("MIXMONITOR_END", "File: %s\r\n", mixmonitor->filename);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (!AST_LIST_EMPTY(&mixmonitor->recipient_list)) {
if (ast_strlen_zero(fs_ext)) {
ast_log(LOG_ERROR, "No file extension set for Mixmonitor %s. Skipping copy to voicemail.\n",
mixmonitor -> name);
} else {
ast_verb(3, "Copying recordings for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
copy_to_voicemail(mixmonitor, fs_ext, mixmonitor->filename);
}
if (!ast_strlen_zero(fs_read_ext)) {
ast_verb(3, "Copying read recording for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
copy_to_voicemail(mixmonitor, fs_read_ext, mixmonitor->filename_read);
}
if (!ast_strlen_zero(fs_write_ext)) {
ast_verb(3, "Copying write recording for Mixmonitor %s to voicemail recipients\n", mixmonitor->name);
copy_to_voicemail(mixmonitor, fs_write_ext, mixmonitor->filename_write);
}
} else {
ast_debug(3, "No recipients to forward monitor to, moving on.\n");
}
if (ast_test_flag(mixmonitor, MUXFLAG_AUTO_DELETE)) {
ast_debug(3, "Deleting our copies of recording files\n");
if (!ast_strlen_zero(fs_ext)) {
ast_filedelete(mixmonitor->filename, fs_ext);
}
if (!ast_strlen_zero(fs_read_ext)) {
ast_filedelete(mixmonitor->filename_read, fs_ext);
}
if (!ast_strlen_zero(fs_write_ext)) {
ast_filedelete(mixmonitor->filename_write, fs_ext);
}
}
mixmonitor_free(mixmonitor);
ast_module_unref(ast_module_info->self);
return NULL;
}
static int setup_mixmonitor_ds(struct mixmonitor *mixmonitor, struct ast_channel *chan, char **datastore_id, const char *beep_id)
{
struct ast_datastore *datastore = NULL;
struct mixmonitor_ds *mixmonitor_ds;
if (!(mixmonitor_ds = ast_calloc(1, sizeof(*mixmonitor_ds)))) {
return -1;
}
if (ast_asprintf(datastore_id, "%p", mixmonitor_ds) == -1) {
ast_log(LOG_ERROR, "Failed to allocate memory for MixMonitor ID.\n");
ast_free(mixmonitor_ds);
return -1;
}
ast_mutex_init(&mixmonitor_ds->lock);
ast_cond_init(&mixmonitor_ds->destruction_condition, NULL);
if (!(datastore = ast_datastore_alloc(&mixmonitor_ds_info, *datastore_id))) {
ast_mutex_destroy(&mixmonitor_ds->lock);
ast_cond_destroy(&mixmonitor_ds->destruction_condition);
ast_free(mixmonitor_ds);
return -1;
}
if (ast_test_flag(mixmonitor, MUXFLAG_BEEP_START)) {
ast_autochan_channel_lock(mixmonitor->autochan);
ast_stream_and_wait(mixmonitor->autochan->chan, "beep", "");
ast_autochan_channel_unlock(mixmonitor->autochan);
}
mixmonitor_ds->samp_rate = 8000;
mixmonitor_ds->audiohook = &mixmonitor->audiohook;
mixmonitor_ds->filename = ast_strdup(mixmonitor->filename);
if (!ast_strlen_zero(beep_id)) {
mixmonitor_ds->beep_id = ast_strdup(beep_id);
}
datastore->data = mixmonitor_ds;
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
mixmonitor->mixmonitor_ds = mixmonitor_ds;
return 0;
}
static void mixmonitor_ds_remove_and_free(struct ast_channel *chan, const char *datastore_id)
{
struct ast_datastore *datastore;
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, datastore_id);
/*
* Currently the one place this function is called from guarantees a
* datastore is present, thus return checks can be avoided here.
*/
ast_channel_datastore_remove(chan, datastore);
ast_datastore_free(datastore);
ast_channel_unlock(chan);
}
static int launch_monitor_thread(struct ast_channel *chan, const char *filename,
unsigned int flags, int readvol, int writevol,
const char *post_process, const char *filename_write,
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
char *filename_read, const char *uid_channel_var,
const char *recipients, const char *beep_id)
{
pthread_t thread;
struct mixmonitor *mixmonitor;
char postprocess2[1024] = "";
char *datastore_id = NULL;
postprocess2[0] = 0;
/* If a post process system command is given attach it to the structure */
if (!ast_strlen_zero(post_process)) {
char *p1, *p2;
p1 = ast_strdupa(post_process);
for (p2 = p1; *p2; p2++) {
if (*p2 == '^' && *(p2+1) == '{') {
*p2 = '$';
}
}
ast_channel_lock(chan);
pbx_substitute_variables_helper(chan, p1, postprocess2, sizeof(postprocess2) - 1);
ast_channel_unlock(chan);
}
/* Pre-allocate mixmonitor structure and spy */
if (!(mixmonitor = ast_calloc(1, sizeof(*mixmonitor)))) {
return -1;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
/* Now that the struct has been calloced, go ahead and initialize the string fields. */
if (ast_string_field_init(mixmonitor, 512)) {
mixmonitor_free(mixmonitor);
return -1;
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
/* Setup the actual spy before creating our thread */
if (ast_audiohook_init(&mixmonitor->audiohook, AST_AUDIOHOOK_TYPE_SPY, mixmonitor_spy_type, 0)) {
mixmonitor_free(mixmonitor);
return -1;
}
/* Copy over flags and channel name */
mixmonitor->flags = flags;
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
if (!(mixmonitor->autochan = ast_autochan_setup(chan))) {
mixmonitor_free(mixmonitor);
return -1;
}
if (!ast_strlen_zero(filename)) {
mixmonitor->filename = ast_strdup(filename);
}
if (!ast_strlen_zero(filename_write)) {
mixmonitor->filename_write = ast_strdup(filename_write);
}
if (!ast_strlen_zero(filename_read)) {
mixmonitor->filename_read = ast_strdup(filename_read);
}
if (setup_mixmonitor_ds(mixmonitor, chan, &datastore_id, beep_id)) {
ast_autochan_destroy(mixmonitor->autochan);
mixmonitor_free(mixmonitor);
ast_free(datastore_id);
return -1;
Merged revisions 173559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r173559 | mmichelson | 2009-02-05 11:34:33 -0600 (Thu, 05 Feb 2009) | 25 lines Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up. app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed audio to a file. Since this thread runs independently of the channel, it is possible that the mixmonitor thread's channel pointer will point to freed memory when the channel either is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the cases slightly differently). The solution for this is to employ a datastore, which has the nice benefit of allowing us to hook into channel masquerades and hangups and update our pointer as necessary. If this looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more involved since it does a lot more operations on the channel that is being spied upon. app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em- ploy a condition-and-boolean combination to ensure that the channel thread finishes with our structure before the mixmonitor thread attempts to free it. No crashes! (closes issue #14374) Reported by: aragon Patches: 14374.patch uploaded by putnopvut (license 60) Tested by: aragon, putnopvut ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@173589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-05 18:34:06 +00:00
}
if (!ast_strlen_zero(uid_channel_var)) {
if (datastore_id) {
pbx_builtin_setvar_helper(chan, uid_channel_var, datastore_id);
}
}
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
mixmonitor->name = ast_strdup(ast_channel_name(chan));
if (!ast_strlen_zero(postprocess2)) {
mixmonitor->post_process = ast_strdup(postprocess2);
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (!ast_strlen_zero(recipients)) {
char callerid[256];
ast_channel_lock(chan);
/* We use the connected line of the invoking channel for caller ID,
* unless we've been told to use the Caller ID.
* The initial use for this relied on Connected Line to get the
* actual number for recording with Digium phones,
* but in generic use the Caller ID is likely what people want.
*/
if (ast_test_flag(mixmonitor, MUXFLAG_REAL_CALLERID)) {
struct ast_party_caller *caller;
caller = ast_channel_caller(chan);
ast_debug(3, "Caller ID = %d - %s : %d - %s\n", caller->id.name.valid,
caller->id.name.str, caller->id.number.valid,
caller->id.number.str);
ast_callerid_merge(callerid, sizeof(callerid),
S_COR(caller->id.name.valid, caller->id.name.str, NULL),
S_COR(caller->id.number.valid, caller->id.number.str, NULL),
"Unknown");
} else {
struct ast_party_connected_line *connected;
connected = ast_channel_connected(chan);
ast_debug(3, "Connected Line CID = %d - %s : %d - %s\n", connected->id.name.valid,
connected->id.name.str, connected->id.number.valid,
connected->id.number.str);
ast_callerid_merge(callerid, sizeof(callerid),
S_COR(connected->id.name.valid, connected->id.name.str, NULL),
S_COR(connected->id.number.valid, connected->id.number.str, NULL),
"Unknown");
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
ast_string_field_set(mixmonitor, call_context, ast_channel_context(chan));
ast_string_field_set(mixmonitor, call_extension, ast_channel_exten(chan));
ast_string_field_set(mixmonitor, call_callerchan, ast_channel_name(chan));
ast_string_field_set(mixmonitor, call_callerid, callerid);
mixmonitor->call_priority = ast_channel_priority(chan);
ast_channel_unlock(chan);
add_vm_recipients_from_string(mixmonitor, recipients);
}
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_TRIGGER_SYNC);
if (!ast_test_flag(mixmonitor, MUXFLAG_NO_RWSYNC)) {
ast_set_flag(&mixmonitor->audiohook, AST_AUDIOHOOK_SUBSTITUTE_SILENCE);
}
if (readvol)
mixmonitor->audiohook.options.read_volume = readvol;
if (writevol)
mixmonitor->audiohook.options.write_volume = writevol;
if (startmon(chan, &mixmonitor->audiohook)) {
ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-01-09 22:15:50 +00:00
mixmonitor_spy_type, ast_channel_name(chan));
mixmonitor_ds_remove_and_free(chan, datastore_id);
ast_free(datastore_id);
ast_autochan_destroy(mixmonitor->autochan);
ast_audiohook_destroy(&mixmonitor->audiohook);
mixmonitor_free(mixmonitor);
return -1;
}
ast_free(datastore_id);
/* reference be released at mixmonitor destruction */
mixmonitor->callid = ast_read_threadstorage_callid();
return ast_pthread_create_detached_background(&thread, NULL, mixmonitor_thread, mixmonitor);
}
/* a note on filename_parse: creates directory structure and assigns absolute path from relative paths for filenames */
/* requires immediate copying of string from return to retain data since otherwise it will immediately lose scope */
static char *filename_parse(char *filename, char *buffer, size_t len)
{
char *slash;
char *ext;
ast_assert(len > 0);
if (ast_strlen_zero(filename)) {
ast_log(LOG_WARNING, "No file name was provided for a file save option.\n");
buffer[0] = 0;
return buffer;
}
/* If we don't have an absolute path, make one */
if (*filename != '/') {
char *build = ast_alloca(strlen(ast_config_AST_MONITOR_DIR) + strlen(filename) + 3);
sprintf(build, "%s/%s", ast_config_AST_MONITOR_DIR, filename);
filename = build;
}
ast_copy_string(buffer, filename, len);
/* If the provided filename has a .wav49 extension, we need to convert it to .WAV to
match the behavior of build_filename in main/file.c. Otherwise MIXMONITOR_FILENAME
ends up referring to a file that does not/will not exist */
ext = strrchr(buffer, '.');
if (ext && !strcmp(ext, ".wav49")) {
/* Change to WAV - we know we have at least 6 writeable bytes where 'ext' points,
* so this is safe */
memcpy(ext, ".WAV", sizeof(".WAV"));
}
if ((slash = strrchr(filename, '/'))) {
*slash = '\0';
}
ast_mkdir(filename, 0777);
return buffer;
}
static int mixmonitor_exec(struct ast_channel *chan, const char *data)
{
int x, readvol = 0, writevol = 0;
char *filename_read = NULL;
char *filename_write = NULL;
char filename_buffer[1024] = "";
char *uid_channel_var = NULL;
char beep_id[64] = "";
struct ast_flags flags = { 0 };
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
char *recipients = NULL;
char *parse;
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(filename);
AST_APP_ARG(options);
AST_APP_ARG(post_process);
);
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename or ,t(filename) and/or r(filename)\n");
return -1;
}
parse = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, parse);
if (args.options) {
char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
ast_app_parse_options(mixmonitor_opts, &flags, opts, args.options);
if (ast_test_flag(&flags, MUXFLAG_DEPRECATED_RWSYNC)) {
ast_log(LOG_NOTICE, "The synchronization behavior enabled by the 'S' option is now the default"
" and does not need to be specified.\n");
}
if (ast_test_flag(&flags, MUXFLAG_READVOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_READVOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the heard volume ('v') option.\n");
} else if ((sscanf(opts[OPT_ARG_READVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Heard volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_READVOLUME]);
} else {
readvol = get_volfactor(x);
}
}
if (ast_test_flag(&flags, MUXFLAG_WRITEVOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_WRITEVOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the spoken volume ('V') option.\n");
} else if ((sscanf(opts[OPT_ARG_WRITEVOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Spoken volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_WRITEVOLUME]);
} else {
writevol = get_volfactor(x);
}
}
if (ast_test_flag(&flags, MUXFLAG_VOLUME)) {
if (ast_strlen_zero(opts[OPT_ARG_VOLUME])) {
ast_log(LOG_WARNING, "No volume level was provided for the combined volume ('W') option.\n");
} else if ((sscanf(opts[OPT_ARG_VOLUME], "%2d", &x) != 1) || (x < -4) || (x > 4)) {
ast_log(LOG_NOTICE, "Combined volume must be a number between -4 and 4, not '%s'\n", opts[OPT_ARG_VOLUME]);
} else {
readvol = writevol = get_volfactor(x);
}
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (ast_test_flag(&flags, MUXFLAG_VMRECIPIENTS)) {
if (ast_strlen_zero(opts[OPT_ARG_VMRECIPIENTS])) {
ast_log(LOG_WARNING, "No voicemail recipients were specified for the vm copy ('m') option.\n");
} else {
recipients = ast_strdupa(opts[OPT_ARG_VMRECIPIENTS]);
}
}
if (ast_test_flag(&flags, MUXFLAG_WRITE)) {
filename_write = ast_strdupa(filename_parse(opts[OPT_ARG_WRITENAME], filename_buffer, sizeof(filename_buffer)));
}
if (ast_test_flag(&flags, MUXFLAG_READ)) {
filename_read = ast_strdupa(filename_parse(opts[OPT_ARG_READNAME], filename_buffer, sizeof(filename_buffer)));
}
if (ast_test_flag(&flags, MUXFLAG_UID)) {
uid_channel_var = opts[OPT_ARG_UID];
}
if (ast_test_flag(&flags, MUXFLAG_BEEP)) {
const char *interval_str = S_OR(opts[OPT_ARG_BEEP_INTERVAL], "15");
unsigned int interval = 15;
if (sscanf(interval_str, "%30u", &interval) != 1) {
ast_log(LOG_WARNING, "Invalid interval '%s' for periodic beep. Using default of %u\n",
interval_str, interval);
}
if (ast_beep_start(chan, interval, beep_id, sizeof(beep_id))) {
ast_log(LOG_WARNING, "Unable to enable periodic beep, please ensure func_periodic_hook is loaded.\n");
return -1;
}
}
}
/* If there are no file writing arguments/options for the mix monitor, send a warning message and return -1 */
if (!ast_test_flag(&flags, MUXFLAG_WRITE) && !ast_test_flag(&flags, MUXFLAG_READ) && ast_strlen_zero(args.filename)) {
ast_log(LOG_WARNING, "MixMonitor requires an argument (filename)\n");
return -1;
}
/* If filename exists, try to create directories for it */
if (!(ast_strlen_zero(args.filename))) {
args.filename = ast_strdupa(filename_parse(args.filename, filename_buffer, sizeof(filename_buffer)));
}
pbx_builtin_setvar_helper(chan, "MIXMONITOR_FILENAME", args.filename);
/* If launch_monitor_thread works, the module reference must not be released until it is finished. */
ast_module_ref(ast_module_info->self);
if (launch_monitor_thread(chan,
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
args.filename,
flags.flags,
readvol,
writevol,
args.post_process,
filename_write,
filename_read,
uid_channel_var,
recipients,
beep_id)) {
ast_module_unref(ast_module_info->self);
}
message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
ast_channel_mixmonitor_start_type(), NULL);
if (message) {
stasis_publish(ast_channel_topic(chan), message);
}
return 0;
}
static int stop_mixmonitor_full(struct ast_channel *chan, const char *data)
{
struct ast_datastore *datastore = NULL;
char *parse = "";
struct mixmonitor_ds *mixmonitor_ds;
const char *beep_id = NULL;
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(mixmonid);
);
if (!ast_strlen_zero(data)) {
parse = ast_strdupa(data);
}
AST_STANDARD_APP_ARGS(args, parse);
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info,
S_OR(args.mixmonid, NULL));
if (!datastore) {
ast_channel_unlock(chan);
return -1;
}
mixmonitor_ds = datastore->data;
ast_mutex_lock(&mixmonitor_ds->lock);
/* closing the filestream here guarantees the file is available to the dialplan
* after calling StopMixMonitor */
mixmonitor_ds_close_fs(mixmonitor_ds);
/* The mixmonitor thread may be waiting on the audiohook trigger.
* In order to exit from the mixmonitor loop before waiting on channel
* destruction, poke the audiohook trigger. */
if (mixmonitor_ds->audiohook) {
if (mixmonitor_ds->audiohook->status != AST_AUDIOHOOK_STATUS_DONE) {
ast_audiohook_update_status(mixmonitor_ds->audiohook, AST_AUDIOHOOK_STATUS_SHUTDOWN);
}
ast_audiohook_lock(mixmonitor_ds->audiohook);
ast_cond_signal(&mixmonitor_ds->audiohook->trigger);
ast_audiohook_unlock(mixmonitor_ds->audiohook);
mixmonitor_ds->audiohook = NULL;
}
if (!ast_strlen_zero(mixmonitor_ds->beep_id)) {
beep_id = ast_strdupa(mixmonitor_ds->beep_id);
}
ast_mutex_unlock(&mixmonitor_ds->lock);
/* Remove the datastore so the monitor thread can exit */
if (!ast_channel_datastore_remove(chan, datastore)) {
ast_datastore_free(datastore);
}
ast_channel_unlock(chan);
if (!ast_strlen_zero(beep_id)) {
ast_beep_stop(chan, beep_id);
}
message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(chan),
ast_channel_mixmonitor_stop_type(),
NULL);
if (message) {
stasis_publish(ast_channel_topic(chan), message);
}
return 0;
}
static int stop_mixmonitor_exec(struct ast_channel *chan, const char *data)
{
stop_mixmonitor_full(chan, data);
return 0;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_cli_mixmonitor(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_channel *chan;
struct ast_datastore *datastore = NULL;
struct mixmonitor_ds *mixmonitor_ds = NULL;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "mixmonitor {start|stop|list}";
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
e->usage =
"Usage: mixmonitor start <chan_name> [args]\n"
" The optional arguments are passed to the MixMonitor application.\n"
" mixmonitor stop <chan_name> [args]\n"
" The optional arguments are passed to the StopMixMonitor application.\n"
" mixmonitor list <chan_name>\n";
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
}
if (a->argc < 3) {
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
}
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
if (!(chan = ast_channel_get_by_name_prefix(a->argv[2], strlen(a->argv[2])))) {
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "No channel matching '%s' found.\n", a->argv[2]);
/* Technically this is a failure, but we don't want 2 errors printing out */
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
if (!strcasecmp(a->argv[1], "start")) {
mixmonitor_exec(chan, (a->argc >= 4) ? a->argv[3] : "");
} else if (!strcasecmp(a->argv[1], "stop")){
stop_mixmonitor_exec(chan, (a->argc >= 4) ? a->argv[3] : "");
} else if (!strcasecmp(a->argv[1], "list")) {
ast_cli(a->fd, "MixMonitor ID\tFile\tReceive File\tTransmit File\n");
ast_cli(a->fd, "=========================================================================\n");
ast_channel_lock(chan);
AST_LIST_TRAVERSE(ast_channel_datastores(chan), datastore, entry) {
if (datastore->info == &mixmonitor_ds_info) {
char *filename = "";
char *filename_read = "";
char *filename_write = "";
mixmonitor_ds = datastore->data;
if (mixmonitor_ds->fs) {
filename = mixmonitor_ds->fs->filename;
}
if (mixmonitor_ds->fs_read) {
filename_read = mixmonitor_ds->fs_read->filename;
}
if (mixmonitor_ds->fs_write) {
filename_write = mixmonitor_ds->fs_write->filename;
}
ast_cli(a->fd, "%p\t%s\t%s\t%s\n", mixmonitor_ds, filename, filename_read, filename_write);
}
}
ast_channel_unlock(chan);
} else {
chan = ast_channel_unref(chan);
return CLI_SHOWUSAGE;
}
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
chan = ast_channel_unref(chan);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
/*! \brief Mute / unmute an individual MixMonitor by id */
static int mute_mixmonitor_instance(struct ast_channel *chan, const char *data,
enum ast_audiohook_flags flag, int clearmute)
{
struct ast_datastore *datastore = NULL;
char *parse = "";
struct mixmonitor_ds *mixmonitor_ds;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(mixmonid);
);
if (!ast_strlen_zero(data)) {
parse = ast_strdupa(data);
}
AST_STANDARD_APP_ARGS(args, parse);
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info,
S_OR(args.mixmonid, NULL));
if (!datastore) {
ast_channel_unlock(chan);
return -1;
}
mixmonitor_ds = datastore->data;
ast_mutex_lock(&mixmonitor_ds->lock);
if (mixmonitor_ds->audiohook) {
if (clearmute) {
ast_clear_flag(mixmonitor_ds->audiohook, flag);
} else {
ast_set_flag(mixmonitor_ds->audiohook, flag);
}
}
ast_mutex_unlock(&mixmonitor_ds->lock);
ast_channel_unlock(chan);
return 0;
}
/*! \brief Mute / unmute a MixMonitor channel */
static int manager_mute_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
const char *state = astman_get_header(m, "State");
const char *direction = astman_get_header(m,"Direction");
const char *mixmonitor_id = astman_get_header(m, "MixMonitorID");
int clearmute = 1, mutedcount = 0;
enum ast_audiohook_flags flag;
RAII_VAR(struct stasis_message *, stasis_message, NULL, ao2_cleanup);
RAII_VAR(struct ast_json *, stasis_message_blob, NULL, ast_json_unref);
if (ast_strlen_zero(direction)) {
astman_send_error(s, m, "No direction specified. Must be read, write or both");
return AMI_SUCCESS;
}
if (!strcasecmp(direction, "read")) {
flag = AST_AUDIOHOOK_MUTE_READ;
} else if (!strcasecmp(direction, "write")) {
flag = AST_AUDIOHOOK_MUTE_WRITE;
} else if (!strcasecmp(direction, "both")) {
flag = AST_AUDIOHOOK_MUTE_READ | AST_AUDIOHOOK_MUTE_WRITE;
} else {
astman_send_error(s, m, "Invalid direction specified. Must be read, write or both");
return AMI_SUCCESS;
}
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
if (ast_strlen_zero(state)) {
astman_send_error(s, m, "No state specified");
return AMI_SUCCESS;
}
clearmute = ast_false(state);
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
if (ast_strlen_zero(mixmonitor_id)) {
mutedcount = ast_audiohook_set_mute_all(c, mixmonitor_spy_type, flag, clearmute);
if (mutedcount < 0) {
ast_channel_unref(c);
astman_send_error(s, m, "Cannot set mute flag");
return AMI_SUCCESS;
}
} else {
if (mute_mixmonitor_instance(c, mixmonitor_id, flag, clearmute)) {
ast_channel_unref(c);
astman_send_error(s, m, "Cannot set mute flag");
return AMI_SUCCESS;
}
mutedcount = 1;
}
stasis_message_blob = ast_json_pack("{s: s, s: b, s: s, s: i}",
"direction", direction,
"state", ast_true(state),
"mixmonitorid", mixmonitor_id,
"count", mutedcount);
stasis_message = ast_channel_blob_create_from_cache(ast_channel_uniqueid(c),
ast_channel_mixmonitor_mute_type(), stasis_message_blob);
if (stasis_message) {
stasis_publish(ast_channel_topic(c), stasis_message);
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
astman_append(s, "\r\n");
ast_channel_unref(c);
return AMI_SUCCESS;
}
static int start_mixmonitor_callback(struct ast_channel *chan, const char *filename, const char *options)
{
char args[PATH_MAX];
if (ast_strlen_zero(options)) {
snprintf(args, sizeof(args), "%s", filename);
} else {
snprintf(args, sizeof(args), "%s,%s", filename, options);
}
return mixmonitor_exec(chan, args);
}
static int stop_mixmonitor_callback(struct ast_channel *chan, const char *mixmonitor_id)
{
return stop_mixmonitor_full(chan, mixmonitor_id);
}
static int manager_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
const char *file = astman_get_header(m, "File");
const char *options = astman_get_header(m, "Options");
const char *command = astman_get_header(m, "Command");
char *opts[OPT_ARG_ARRAY_SIZE] = { NULL, };
struct ast_flags flags = { 0 };
char *uid_channel_var = NULL;
const char *mixmonitor_id = NULL;
int res;
char args[PATH_MAX];
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
if (!ast_strlen_zero(options)) {
ast_app_parse_options(mixmonitor_opts, &flags, opts, ast_strdupa(options));
}
snprintf(args, sizeof(args), "%s,%s,%s", file, options, command);
res = mixmonitor_exec(c, args);
if (ast_test_flag(&flags, MUXFLAG_UID)) {
uid_channel_var = opts[OPT_ARG_UID];
ast_channel_lock(c);
mixmonitor_id = pbx_builtin_getvar_helper(c, uid_channel_var);
mixmonitor_id = ast_strdupa(S_OR(mixmonitor_id, ""));
ast_channel_unlock(c);
}
if (res) {
ast_channel_unref(c);
astman_send_error(s, m, "Could not start monitoring channel");
return AMI_SUCCESS;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
if (!ast_strlen_zero(mixmonitor_id)) {
astman_append(s, "MixMonitorID: %s\r\n", mixmonitor_id);
}
astman_append(s, "\r\n");
ast_channel_unref(c);
return AMI_SUCCESS;
}
static int manager_stop_mixmonitor(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
const char *id = astman_get_header(m, "ActionID");
const char *mixmonitor_id = astman_get_header(m, "MixMonitorID");
int res;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return AMI_SUCCESS;
}
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return AMI_SUCCESS;
}
res = stop_mixmonitor_full(c, mixmonitor_id);
if (res) {
ast_channel_unref(c);
astman_send_error(s, m, "Could not stop monitoring channel");
return AMI_SUCCESS;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
astman_append(s, "\r\n");
ast_channel_unref(c);
return AMI_SUCCESS;
}
static int func_mixmonitor_read(struct ast_channel *chan, const char *cmd, char *data,
char *buf, size_t len)
{
struct ast_datastore *datastore;
struct mixmonitor_ds *ds_data;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(id);
AST_APP_ARG(key);
);
AST_STANDARD_APP_ARGS(args, data);
if (ast_strlen_zero(args.id) || ast_strlen_zero(args.key)) {
ast_log(LOG_WARNING, "Not enough arguments provided to %s. "
"An ID and key must be provided\n", cmd);
return -1;
}
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, args.id);
ast_channel_unlock(chan);
if (!datastore) {
ast_log(LOG_WARNING, "Could not find MixMonitor with ID %s\n", args.id);
return -1;
}
ds_data = datastore->data;
if (!strcasecmp(args.key, "filename")) {
ast_copy_string(buf, ds_data->filename, len);
} else {
ast_log(LOG_WARNING, "Unrecognized %s option %s\n", cmd, args.key);
return -1;
}
return 0;
}
static struct ast_custom_function mixmonitor_function = {
.name = "MIXMONITOR",
.read = func_mixmonitor_read,
};
static struct ast_cli_entry cli_mixmonitor[] = {
AST_CLI_DEFINE(handle_cli_mixmonitor, "Execute a MixMonitor command")
};
static int set_mixmonitor_methods(void)
{
struct ast_mixmonitor_methods mixmonitor_methods = {
.start = start_mixmonitor_callback,
.stop = stop_mixmonitor_callback,
};
return ast_set_mixmonitor_methods(&mixmonitor_methods);
}
static int clear_mixmonitor_methods(void)
{
return ast_clear_mixmonitor_methods();
}
static int unload_module(void)
{
int res;
ast_cli_unregister_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_unregister_application(stop_app);
res |= ast_unregister_application(app);
res |= ast_manager_unregister("MixMonitorMute");
res |= ast_manager_unregister("MixMonitor");
res |= ast_manager_unregister("StopMixMonitor");
res |= ast_custom_function_unregister(&mixmonitor_function);
res |= clear_mixmonitor_methods();
return res;
}
static int load_module(void)
{
int res;
ast_cli_register_multiple(cli_mixmonitor, ARRAY_LEN(cli_mixmonitor));
res = ast_register_application_xml(app, mixmonitor_exec);
res |= ast_register_application_xml(stop_app, stop_mixmonitor_exec);
res |= ast_manager_register_xml("MixMonitorMute", EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, manager_mute_mixmonitor);
res |= ast_manager_register_xml("MixMonitor", EVENT_FLAG_SYSTEM, manager_mixmonitor);
res |= ast_manager_register_xml("StopMixMonitor", EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, manager_stop_mixmonitor);
res |= ast_custom_function_register(&mixmonitor_function);
res |= set_mixmonitor_methods();
return res;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "Mixed Audio Monitoring Application",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.optional_modules = "func_periodic_hook",
);