asterisk/res/res_agi.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief AGI - the Asterisk Gateway Interface
*
* \author Mark Spencer <markster@digium.com>
*
*/
/*** MODULEINFO
<depend>res_speech</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <math.h>
#include <signal.h>
#include <sys/time.h>
#include <sys/wait.h>
#include <sys/stat.h>
#include <pthread.h>
#include "asterisk/paths.h" /* use many ast_config_AST_*_DIR */
#include "asterisk/network.h"
#include "asterisk/file.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/module.h"
#include "asterisk/astdb.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/image.h"
#include "asterisk/say.h"
#include "asterisk/app.h"
#include "asterisk/dsp.h"
#include "asterisk/musiconhold.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/strings.h"
#include "asterisk/manager.h"
#include "asterisk/ast_version.h"
#include "asterisk/speech.h"
#include "asterisk/manager.h"
#include "asterisk/term.h"
#include "asterisk/xmldoc.h"
#include "asterisk/srv.h"
#include "asterisk/test.h"
#include "asterisk/netsock2.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/stasis_message_router.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"
#define AST_API_MODULE
#include "asterisk/agi.h"
/*** DOCUMENTATION
<agi name="answer" language="en_US">
<synopsis>
Answer channel
</synopsis>
<syntax />
<description>
<para>Answers channel if not already in answer state. Returns <literal>-1</literal> on
channel failure, or <literal>0</literal> if successful.</para>
</description>
<see-also>
<ref type="agi">hangup</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="asyncagi break" language="en_US">
<synopsis>
Interrupts Async AGI
</synopsis>
<syntax />
<description>
<para>Interrupts expected flow of Async AGI commands and returns control to previous source
(typically, the PBX dialplan).</para>
</description>
<see-also>
<ref type="agi">hangup</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="channel status" language="en_US">
<synopsis>
Returns status of the connected channel.
</synopsis>
<syntax>
<parameter name="channelname" />
</syntax>
<description>
<para>Returns the status of the specified <replaceable>channelname</replaceable>.
If no channel name is given then returns the status of the current channel.</para>
<para>Return values:</para>
<enumlist>
<enum name="0">
<para>Channel is down and available.</para>
</enum>
<enum name="1">
<para>Channel is down, but reserved.</para>
</enum>
<enum name="2">
<para>Channel is off hook.</para>
</enum>
<enum name="3">
<para>Digits (or equivalent) have been dialed.</para>
</enum>
<enum name="4">
<para>Line is ringing.</para>
</enum>
<enum name="5">
<para>Remote end is ringing.</para>
</enum>
<enum name="6">
<para>Line is up.</para>
</enum>
<enum name="7">
<para>Line is busy.</para>
</enum>
</enumlist>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="control stream file" language="en_US">
<synopsis>
Sends audio file on channel and allows the listener to control the stream.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>The file extension must not be included in the filename.</para>
</parameter>
<parameter name="escape_digits" required="true" />
<parameter name="skipms" />
<parameter name="ffchar">
<para>Defaults to <literal>#</literal></para>
</parameter>
<parameter name="rewchr">
<para>Defaults to <literal>*</literal></para>
</parameter>
<parameter name="pausechr" />
<parameter name="offsetms">
<para>Offset, in milliseconds, to start the audio playback</para>
</parameter>
</syntax>
<description>
<para>Send the given file, allowing playback to be controlled by the given
digits, if any. Use double quotes for the digits if you wish none to be
permitted. If offsetms is provided then the audio will seek to offsetms
before play starts. Returns <literal>0</literal> if playback completes without a digit
being pressed, or the ASCII numerical value of the digit if one was pressed,
or <literal>-1</literal> on error or if the channel was disconnected. Returns the
position where playback was terminated as endpos.</para>
<para>It sets the following channel variables upon completion:</para>
<variablelist>
<variable name="CPLAYBACKSTATUS">
<para>Contains the status of the attempt as a text string</para>
<value name="SUCCESS" />
<value name="USERSTOPPED" />
<value name="REMOTESTOPPED" />
<value name="ERROR" />
</variable>
<variable name="CPLAYBACKOFFSET">
<para>Contains the offset in ms into the file where playback
was at when it stopped. <literal>-1</literal> is end of file.</para>
</variable>
<variable name="CPLAYBACKSTOPKEY">
<para>If the playback is stopped by the user this variable contains
the key that was pressed.</para>
</variable>
</variablelist>
</description>
<see-also>
<ref type="agi">get option</ref>
<ref type="agi">control stream file</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database del" language="en_US">
<synopsis>
Removes database key/value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>Deletes an entry in the Asterisk database for a given
<replaceable>family</replaceable> and <replaceable>key</replaceable>.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal>
otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database put</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database deltree" language="en_US">
<synopsis>
Removes database keytree/value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="keytree" />
</syntax>
<description>
<para>Deletes a <replaceable>family</replaceable> or specific <replaceable>keytree</replaceable>
within a <replaceable>family</replaceable> in the Asterisk database.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal> otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database put</ref>
<ref type="agi">database del</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database get" language="en_US">
<synopsis>
Gets database value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
</syntax>
<description>
<para>Retrieves an entry in the Asterisk database for a given <replaceable>family</replaceable>
and <replaceable>key</replaceable>.</para>
<para>Returns <literal>0</literal> if <replaceable>key</replaceable> is not set.
Returns <literal>1</literal> if <replaceable>key</replaceable> is set and returns the variable
in parenthesis.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">database put</ref>
<ref type="agi">database del</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="database put" language="en_US">
<synopsis>
Adds/updates database value
</synopsis>
<syntax>
<parameter name="family" required="true" />
<parameter name="key" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Adds or updates an entry in the Asterisk database for a given
<replaceable>family</replaceable>, <replaceable>key</replaceable>, and
<replaceable>value</replaceable>.</para>
<para>Returns <literal>1</literal> if successful, <literal>0</literal> otherwise.</para>
</description>
<see-also>
<ref type="agi">database get</ref>
<ref type="agi">database del</ref>
<ref type="agi">database deltree</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="exec" language="en_US">
<synopsis>
Executes a given Application
</synopsis>
<syntax>
<parameter name="application" required="true" />
<parameter name="options" required="true" />
</syntax>
<description>
<para>Executes <replaceable>application</replaceable> with given
<replaceable>options</replaceable>.</para>
<para>Returns whatever the <replaceable>application</replaceable> returns, or
<literal>-2</literal> on failure to find <replaceable>application</replaceable>.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get data" language="en_US">
<synopsis>
Prompts for DTMF on a channel
</synopsis>
<syntax>
<parameter name="file" required="true" />
<parameter name="timeout" />
<parameter name="maxdigits" />
</syntax>
<description>
<para>Stream the given <replaceable>file</replaceable>, and receive DTMF data.</para>
<para>Returns the digits received from the channel at the other end.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get full variable" language="en_US">
<synopsis>
Evaluates a channel expression
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
<parameter name="channel name" />
</syntax>
<description>
<para>Returns <literal>0</literal> if <replaceable>variablename</replaceable> is not set
or channel does not exist. Returns <literal>1</literal> if <replaceable>variablename</replaceable>
is set and returns the variable in parenthesis. Understands complex variable names and builtin
variables, unlike GET VARIABLE.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">get variable</ref>
<ref type="agi">set variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get option" language="en_US">
<synopsis>
Stream file, prompt for DTMF, with timeout.
</synopsis>
<syntax>
<parameter name="filename" required="true" />
<parameter name="escape_digits" required="true" />
<parameter name="timeout" />
</syntax>
<description>
<para>Behaves similar to STREAM FILE but used with a timeout option.</para>
</description>
<see-also>
<ref type="agi">stream file</ref>
<ref type="agi">control stream file</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="get variable" language="en_US">
<synopsis>
Gets a channel variable.
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
</syntax>
<description>
<para>Returns <literal>0</literal> if <replaceable>variablename</replaceable> is not set.
Returns <literal>1</literal> if <replaceable>variablename</replaceable> is set and returns
the variable in parentheses.</para>
<para>Example return code: 200 result=1 (testvariable)</para>
</description>
<see-also>
<ref type="agi">get full variable</ref>
<ref type="agi">set variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="hangup" language="en_US">
<synopsis>
Hangup a channel.
</synopsis>
<syntax>
<parameter name="channelname" />
</syntax>
<description>
<para>Hangs up the specified channel. If no channel name is given, hangs
up the current channel</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="noop" language="en_US">
<synopsis>
Does nothing.
</synopsis>
<syntax />
<description>
<para>Does nothing.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="receive char" language="en_US">
<synopsis>
Receives one character from channels supporting it.
</synopsis>
<syntax>
<parameter name="timeout" required="true">
<para>The maximum time to wait for input in milliseconds, or <literal>0</literal>
for infinite. Most channels</para>
</parameter>
</syntax>
<description>
<para>Receives a character of text on a channel. Most channels do not support
the reception of text. Returns the decimal value of the character
if one is received, or <literal>0</literal> if the channel does not support
text reception. Returns <literal>-1</literal> only on error/hangup.</para>
</description>
<see-also>
<ref type="agi">receive text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="receive text" language="en_US">
<synopsis>
Receives text from channels supporting it.
</synopsis>
<syntax>
<parameter name="timeout" required="true">
<para>The timeout to be the maximum time to wait for input in
milliseconds, or <literal>0</literal> for infinite.</para>
</parameter>
</syntax>
<description>
<para>Receives a string of text on a channel. Most channels
do not support the reception of text. Returns <literal>-1</literal> for failure
or <literal>1</literal> for success, and the string in parenthesis.</para>
</description>
<see-also>
<ref type="agi">receive char</ref>
<ref type="agi">send text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="record file" language="en_US">
<synopsis>
Records to a given file.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>The destination filename of the recorded audio.</para>
</parameter>
<parameter name="format" required="true">
<para>The audio format in which to save the resulting file.</para>
</parameter>
<parameter name="escape_digits" required="true">
<para>The DTMF digits that will terminate the recording process.</para>
</parameter>
<parameter name="timeout" required="true">
<para>The maximum recording time in milliseconds. Set to -1 for no
limit.</para>
</parameter>
<parameter name="offset_samples">
<para>Causes the recording to first seek to the specified offset before
recording begins.</para>
</parameter>
<parameter name="beep">
<para>Causes Asterisk to play a beep as recording begins. This argument
can take any value.</para>
</parameter>
<parameter name="s=silence">
<para>The number of seconds of silence that are permitted before the
recording is terminated, regardless of the
<replaceable>escape_digits</replaceable> or <replaceable>timeout</replaceable>
arguments. If specified, this parameter must be preceded by
<literal>s=</literal>.</para>
</parameter>
</syntax>
<description>
<para>Record to a file until a given dtmf digit in the sequence is received.
Returns <literal>-1</literal> on hangup or error. The format will specify what kind of file
will be recorded. The <replaceable>timeout</replaceable> is the maximum record time in
milliseconds, or <literal>-1</literal> for no <replaceable>timeout</replaceable>.
<replaceable>offset samples</replaceable> is optional, and, if provided, will seek
to the offset without exceeding the end of the
file. <replaceable>beep</replaceable> can take any value, and causes Asterisk
to play a beep to the channel that is about to be recorded. <replaceable>silence</replaceable> is
the number of seconds of silence allowed before the function returns despite the
lack of dtmf digits or reaching <replaceable>timeout</replaceable>. <replaceable>silence</replaceable>
value must be preceded by <literal>s=</literal> and is also optional.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say alpha" language="en_US">
<synopsis>
Says a given character string.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given character string, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if one
was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say digits" language="en_US">
<synopsis>
Says a given digit string.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given digit string, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if one
was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say number" language="en_US">
<synopsis>
Says a given number.
</synopsis>
<syntax>
<parameter name="number" required="true" />
<parameter name="escape_digits" required="true" />
<parameter name="gender" />
</syntax>
<description>
<para>Say a given number, returning early if any of the given DTMF digits
are received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of
the digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say phonetic" language="en_US">
<synopsis>
Says a given character string with phonetics.
</synopsis>
<syntax>
<parameter name="string" required="true" />
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given character string with phonetics, returning early if any of the
given DTMF digits are received on the channel. Returns <literal>0</literal> if
playback completes without a digit pressed, the ASCII numerical value of the digit
if one was pressed, or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say date" language="en_US">
<synopsis>
Says a given date.
</synopsis>
<syntax>
<parameter name="date" required="true">
<para>Is number of seconds elapsed since 00:00:00 on January 1, 1970.
Coordinated Universal Time (UTC).</para>
</parameter>
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given date, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of the
digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say time</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say time" language="en_US">
<synopsis>
Says a given time.
</synopsis>
<syntax>
<parameter name="time" required="true">
<para>Is number of seconds elapsed since 00:00:00 on January 1, 1970.
Coordinated Universal Time (UTC).</para>
</parameter>
<parameter name="escape_digits" required="true" />
</syntax>
<description>
<para>Say a given time, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback completes
without a digit being pressed, or the ASCII numerical value of the digit if
one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say datetime</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="say datetime" language="en_US">
<synopsis>
Says a given time as specified by the format given.
</synopsis>
<syntax>
<parameter name="time" required="true">
<para>Is number of seconds elapsed since 00:00:00
on January 1, 1970, Coordinated Universal Time (UTC)</para>
</parameter>
<parameter name="escape_digits" required="true" />
<parameter name="format">
<para>Is the format the time should be said in. See
<filename>voicemail.conf</filename> (defaults to <literal>ABdY
'digits/at' IMp</literal>).</para>
</parameter>
<parameter name="timezone">
<para>Acceptable values can be found in <filename>/usr/share/zoneinfo</filename>
Defaults to machine default.</para>
</parameter>
</syntax>
<description>
<para>Say a given time, returning early if any of the given DTMF digits are
received on the channel. Returns <literal>0</literal> if playback
completes without a digit being pressed, or the ASCII numerical value of the
digit if one was pressed or <literal>-1</literal> on error/hangup.</para>
</description>
<see-also>
<ref type="agi">say alpha</ref>
<ref type="agi">say digits</ref>
<ref type="agi">say number</ref>
<ref type="agi">say phonetic</ref>
<ref type="agi">say date</ref>
<ref type="agi">say time</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="send image" language="en_US">
<synopsis>
Sends images to channels supporting it.
</synopsis>
<syntax>
<parameter name="image" required="true" />
</syntax>
<description>
<para>Sends the given image on a channel. Most channels do not support the
transmission of images. Returns <literal>0</literal> if image is sent, or if
the channel does not support image transmission. Returns <literal>-1</literal>
only on error/hangup. Image names should not include extensions.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="send text" language="en_US">
<synopsis>
Sends text to channels supporting it.
</synopsis>
<syntax>
<parameter name="text to send" required="true">
<para>Text consisting of greater than one word should be placed
in quotes since the command only accepts a single argument.</para>
</parameter>
</syntax>
<description>
<para>Sends the given text on a channel. Most channels do not support the
transmission of text. Returns <literal>0</literal> if text is sent, or if the
channel does not support text transmission. Returns <literal>-1</literal> only
on error/hangup.</para>
</description>
<see-also>
<ref type="agi">receive text</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set autohangup" language="en_US">
<synopsis>
Autohangup channel in some time.
</synopsis>
<syntax>
<parameter name="time" required="true" />
</syntax>
<description>
<para>Cause the channel to automatically hangup at <replaceable>time</replaceable>
seconds in the future. Of course it can be hungup before then as well. Setting to
<literal>0</literal> will cause the autohangup feature to be disabled on this channel.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set callerid" language="en_US">
<synopsis>
Sets callerid for the current channel.
</synopsis>
<syntax>
<parameter name="number" required="true" />
</syntax>
<description>
<para>Changes the callerid of the current channel.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set context" language="en_US">
<synopsis>
Sets channel context.
</synopsis>
<syntax>
<parameter name="desired context" required="true" />
</syntax>
<description>
<para>Sets the context for continuation upon exiting the application.</para>
</description>
<see-also>
<ref type="agi">set extension</ref>
<ref type="agi">set priority</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set extension" language="en_US">
<synopsis>
Changes channel extension.
</synopsis>
<syntax>
<parameter name="new extension" required="true" />
</syntax>
<description>
<para>Changes the extension for continuation upon exiting the application.</para>
</description>
<see-also>
<ref type="agi">set context</ref>
<ref type="agi">set priority</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set music" language="en_US">
<synopsis>
Enable/Disable Music on hold generator
</synopsis>
<syntax>
<parameter required="true">
<enumlist>
<enum>
<parameter name="on" literal="true" required="true" />
</enum>
<enum>
<parameter name="off" literal="true" required="true" />
</enum>
</enumlist>
</parameter>
<parameter name="class" required="true" />
</syntax>
<description>
<para>Enables/Disables the music on hold generator. If <replaceable>class</replaceable>
is not specified, then the <literal>default</literal> music on hold class will be
used. This generator will be stopped automatically when playing a file.</para>
<para>Always returns <literal>0</literal>.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set priority" language="en_US">
<synopsis>
Set channel dialplan priority.
</synopsis>
<syntax>
<parameter name="priority" required="true" />
</syntax>
<description>
<para>Changes the priority for continuation upon exiting the application.
The priority must be a valid priority or label.</para>
</description>
<see-also>
<ref type="agi">set context</ref>
<ref type="agi">set extension</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="set variable" language="en_US">
<synopsis>
Sets a channel variable.
</synopsis>
<syntax>
<parameter name="variablename" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Sets a variable to the current channel.</para>
</description>
<see-also>
<ref type="agi">get variable</ref>
<ref type="agi">get full variable</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="stream file" language="en_US">
<synopsis>
Sends audio file on channel.
</synopsis>
<syntax>
<parameter name="filename" required="true">
<para>File name to play. The file extension must not be
included in the <replaceable>filename</replaceable>.</para>
</parameter>
<parameter name="escape_digits" required="true">
<para>Use double quotes for the digits if you wish none to be
permitted.</para>
</parameter>
<parameter name="sample offset">
<para>If sample offset is provided then the audio will seek to sample
offset before play starts.</para>
</parameter>
</syntax>
<description>
<para>Send the given file, allowing playback to be interrupted by the given
digits, if any. Returns <literal>0</literal> if playback completes without a digit
being pressed, or the ASCII numerical value of the digit if one was pressed,
or <literal>-1</literal> on error or if the channel was disconnected. If
musiconhold is playing before calling stream file it will be automatically
stopped and will not be restarted after completion.</para>
<para>It sets the following channel variables upon completion:</para>
<variablelist>
<variable name="PLAYBACKSTATUS">
<para>The status of the playback attempt as a text string.</para>
<value name="SUCCESS"/>
<value name="FAILED"/>
</variable>
</variablelist>
</description>
<see-also>
<ref type="agi">control stream file</ref>
<ref type="agi">get option</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="tdd mode" language="en_US">
<synopsis>
Toggles TDD mode (for the deaf).
</synopsis>
<syntax>
<parameter name="boolean" required="true">
<enumlist>
<enum name="on" />
<enum name="off" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Enable/Disable TDD transmission/reception on a channel. Returns <literal>1</literal> if
successful, or <literal>0</literal> if channel is not TDD-capable.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="verbose" language="en_US">
<synopsis>
Logs a message to the asterisk verbose log.
</synopsis>
<syntax>
<parameter name="message" required="true" />
<parameter name="level" required="true" />
</syntax>
<description>
<para>Sends <replaceable>message</replaceable> to the console via verbose
message system. <replaceable>level</replaceable> is the verbose level (1-4).
Always returns <literal>1</literal></para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="wait for digit" language="en_US">
<synopsis>
Waits for a digit to be pressed.
</synopsis>
<syntax>
<parameter name="timeout" required="true" />
</syntax>
<description>
<para>Waits up to <replaceable>timeout</replaceable> milliseconds for channel to
receive a DTMF digit. Returns <literal>-1</literal> on channel failure, <literal>0</literal>
if no digit is received in the timeout, or the numerical value of the ascii of the digit if
one is received. Use <literal>-1</literal> for the <replaceable>timeout</replaceable> value if
you desire the call to block indefinitely.</para>
</description>
<see-also>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech create" language="en_US">
<synopsis>
Creates a speech object.
</synopsis>
<syntax>
<parameter name="engine" required="true" />
</syntax>
<description>
<para>Create a speech object to be used by the other Speech AGI commands.</para>
</description>
<see-also>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech set" language="en_US">
<synopsis>
Sets a speech engine setting.
</synopsis>
<syntax>
<parameter name="name" required="true" />
<parameter name="value" required="true" />
</syntax>
<description>
<para>Set an engine-specific setting.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech destroy" language="en_US">
<synopsis>
Destroys a speech object.
</synopsis>
<syntax>
</syntax>
<description>
<para>Destroy the speech object created by <literal>SPEECH CREATE</literal>.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech load grammar" language="en_US">
<synopsis>
Loads a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
<parameter name="path to grammar" required="true" />
</syntax>
<description>
<para>Loads the specified grammar as the specified name.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech unload grammar" language="en_US">
<synopsis>
Unloads a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Unloads the specified grammar.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech activate grammar" language="en_US">
<synopsis>
Activates a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Activates the specified grammar on the speech object.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech deactivate grammar" language="en_US">
<synopsis>
Deactivates a grammar.
</synopsis>
<syntax>
<parameter name="grammar name" required="true" />
</syntax>
<description>
<para>Deactivates the specified grammar on the speech object.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech recognize</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<agi name="speech recognize" language="en_US">
<synopsis>
Recognizes speech.
</synopsis>
<syntax>
<parameter name="prompt" required="true" />
<parameter name="timeout" required="true" />
<parameter name="offset" />
</syntax>
<description>
<para>Plays back given <replaceable>prompt</replaceable> while listening for
speech and dtmf.</para>
</description>
<see-also>
<ref type="agi">speech create</ref>
<ref type="agi">speech set</ref>
<ref type="agi">speech destroy</ref>
<ref type="agi">speech load grammar</ref>
<ref type="agi">speech unload grammar</ref>
<ref type="agi">speech activate grammar</ref>
<ref type="agi">speech deactivate grammar</ref>
<ref type="application">AGI</ref>
</see-also>
</agi>
<application name="AGI" language="en_US">
<synopsis>
Executes an AGI compliant application.
</synopsis>
<syntax>
<parameter name="command" required="true">
<para>How AGI should be invoked on the channel.</para>
</parameter>
<parameter name="args">
<para>Arguments to pass to the AGI script or server.</para>
<argument name="arg1" required="true" />
<argument name="arg2" multiple="yes" />
</parameter>
</syntax>
<description>
<para>Executes an Asterisk Gateway Interface compliant
program on a channel. AGI allows Asterisk to launch external programs written
in any language to control a telephony channel, play audio, read DTMF digits,
etc. by communicating with the AGI protocol.</para>
<para>The following variants of AGI exist, and are chosen based on the value
passed to <replaceable>command</replaceable>:</para>
<enumlist>
<enum name="AGI">
<para>The classic variant of AGI, this will launch the script
specified by <replaceable>command</replaceable> as a new process.
Communication with the script occurs on <literal>stdin</literal> and
<literal>stdout</literal>. If the full path to the script is not
provided, the <directory>astagidir</directory> specified in
<filename>asterisk.conf</filename> will be used.
</para>
</enum>
<enum name="FastAGI">
<para>Connect Asterisk to a FastAGI server using a TCP connection.
The URI to the FastAGI server should be given in the form
<literal>[scheme]://host.domain[:port][/script/name]</literal>,
where <replaceable>scheme</replaceable> is either <literal>agi</literal>
or <literal>hagi</literal>.</para>
<para>In the case of <literal>hagi</literal>, an SRV lookup will be
performed to try to connect to a list of FastAGI servers. The hostname in
the URI must be prefixed with <literal>_agi._tcp</literal>. prior to the DNS resolution. For
example, if you specify the URI <literal>hagi://agi.example.com/foo.agi</literal>
the DNS query would be for <literal>_agi._tcp.agi.example.com</literal>. You
will need to make sure this resolves correctly.</para>
</enum>
<enum name="AsyncAGI">
<para>Use AMI to control the channel in AGI. AGI commands can be invoked
using the <literal>AMI</literal> action, with a variety of AGI specific
events passed back over the AMI connection. AsyncAGI should be invoked
by passing <literal>agi:async</literal> to the <replaceable>command</replaceable>
parameter.</para>
</enum>
</enumlist>
<note>
<para>As of <literal>1.6.0</literal>, this channel will
not stop dialplan execution on hangup inside of this application. Dialplan
execution will continue normally, even upon hangup until the AGI application
signals a desire to stop (either by exiting or, in the case of a net script, by
closing the connection).</para>
<para>A locally executed AGI script will receive <literal>SIGHUP</literal> on
hangup from the channel except when using <literal>DeadAGI</literal>
(or when the channel is already hungup). A fast AGI server will
correspondingly receive a <literal>HANGUP</literal> inline with the command dialog.
Both of these signals may be disabled by setting the <variable>AGISIGHUP</variable>
channel variable to <literal>no</literal> before executing the AGI application.
Alternatively, if you would like the AGI application to exit immediately
after a channel hangup is detected, set the <variable>AGIEXITONHANGUP</variable>
variable to <literal>yes</literal>.</para>
</note>
<example title="AGI invocation examples">
; Start the AGI script /tmp/my-cool-script.sh, passing it the contents
; of the channel variable FOO
same => n,AGI(/tmp/my-cool-script.sh,${FOO})
; Start the AGI script my-cool-script.sh located in the astagidir
; directory, specified in asterisk.conf
same => n,AGI(my-cool-script.sh)
; Connect to the FastAGI server located at 127.0.0.1 and start the script
; awesome-script
same => n,AGI(agi://127.0.0.1/awesome-script)
; Start AsyncAGI
same => n,AGI(agi:async)
</example>
<para>This application sets the following channel variable upon completion:</para>
<variablelist>
<variable name="AGISTATUS">
<para>The status of the attempt to the run the AGI script
text string, one of:</para>
<value name="SUCCESS" />
<value name="FAILURE" />
<value name="NOTFOUND" />
<value name="HANGUP" />
</variable>
</variablelist>
</description>
<see-also>
<ref type="manager">AGI</ref>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="application">EAGI</ref>
<ref type="application">DeadAGI</ref>
<ref type="filename">asterisk.conf</ref>
</see-also>
</application>
<application name="EAGI" language="en_US">
<synopsis>
Executes an EAGI compliant application.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='command'])" />
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='args'])" />
</syntax>
<description>
<para>Using 'EAGI' provides enhanced AGI, with incoming audio available out of band
on file descriptor 3. In all other respects, it behaves in the same fashion as
AGI. See the documentation for the <literal>AGI</literal> dialplan application for
more information on invoking AGI on a channel.</para>
<para>This application sets the following channel variable upon completion:</para>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/description/variablelist)" />
</description>
<see-also>
<ref type="application">AGI</ref>
<ref type="application">DeadAGI</ref>
</see-also>
</application>
<application name="DeadAGI" language="en_US">
<synopsis>
Executes AGI on a hungup channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='command'])" />
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/syntax/parameter[@name='args'])" />
</syntax>
<description>
<warning>
<para>This application is deprecated and may be removed in a future version
of Asterisk. Use the replacement application <literal>AGI</literal> instead
of <literal>DeadAGI</literal>.
</para>
</warning>
<para>Execute AGI on a 'dead' or hungup channel. See the documentation for the
<literal>AGI</literal> dialplan application for more information on invoking
AGI on a channel.</para>
<para>This application sets the following channel variable upon completion:</para>
<xi:include xpointer="xpointer(/docs/application[@name='AGI']/description/variablelist)" />
</description>
<see-also>
<ref type="application">AGI</ref>
<ref type="application">EAGI</ref>
</see-also>
</application>
<manager name="AGI" language="en_US">
<synopsis>
Add an AGI command to execute by Async AGI.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel that is currently in Async AGI.</para>
</parameter>
<parameter name="Command" required="true">
<para>Application to execute.</para>
</parameter>
<parameter name="CommandID">
<para>This will be sent back in CommandID header of AsyncAGI exec
event notification.</para>
</parameter>
</syntax>
<description>
<para>Add an AGI command to the execute queue of the channel in Async AGI.</para>
</description>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="AsyncAGIStart">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a channel starts AsyncAGI command processing.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Env">
<para>URL encoded string read from the AsyncAGI server.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AsyncAGIEnd">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a channel stops AsyncAGI command processing.</synopsis>
<syntax>
<channel_snapshot/>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIExec</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AsyncAGIExec">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when AsyncAGI completes an AGI command.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="CommandID" required="false">
<para>Optional command ID sent by the AsyncAGI server to identify the command.</para>
</parameter>
<parameter name="Result">
<para>URL encoded result string from the executed AGI command.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AsyncAGIStart</ref>
<ref type="managerEvent">AsyncAGIEnd</ref>
<ref type="application">AGI</ref>
<ref type="manager">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AGIExecStart">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a received AGI command starts processing.</synopsis>
<syntax>
<channel_snapshot/>
<parameter name="Command">
<para>The AGI command as received from the external source.</para>
</parameter>
<parameter name="CommandId">
<para>Random identification number assigned to the execution of this command.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AGIExecEnd</ref>
<ref type="application">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="AGIExecEnd">
<managerEventInstance class="EVENT_FLAG_AGI">
<synopsis>Raised when a received AGI command completes processing.</synopsis>
<syntax>
<channel_snapshot/>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='AGIExecStart']/managerEventInstance/syntax/parameter)" />
<parameter name="ResultCode">
<para>The numeric result code from AGI</para>
</parameter>
<parameter name="Result">
<para>The text result reason from AGI</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">AGIExecStart</ref>
<ref type="application">AGI</ref>
</see-also>
</managerEventInstance>
</managerEvent>
***/
#define MAX_ARGS 128
#define MAX_CMD_LEN 80
#define AGI_NANDFS_RETRY 3
#define AGI_BUF_LEN 2048
#define SRV_PREFIX "_agi._tcp."
static char *app = "AGI";
static char *eapp = "EAGI";
static char *deadapp = "DeadAGI";
static int agidebug = 0;
#define TONE_BLOCK_SIZE 200
/* Max time to connect to an AGI remote host */
#define MAX_AGI_CONNECT 2000
#define AGI_PORT 4573
/*! Special return code for "asyncagi break" command. */
#define ASYNC_AGI_BREAK 3
enum agi_result {
AGI_RESULT_FAILURE = -1,
AGI_RESULT_SUCCESS,
AGI_RESULT_SUCCESS_FAST,
AGI_RESULT_SUCCESS_ASYNC,
AGI_RESULT_NOTFOUND,
AGI_RESULT_HANGUP,
};
static struct ast_manager_event_blob *agi_channel_to_ami(const char *type, struct stasis_message *message)
{
struct ast_channel_blob *obj = stasis_message_data(message);
RAII_VAR(struct ast_str *, channel_string, NULL, ast_free);
RAII_VAR(struct ast_str *, event_string, NULL, ast_free);
channel_string = ast_manager_build_channel_state_string(obj->snapshot);
event_string = ast_manager_str_from_json_object(obj->blob, NULL);
if (!channel_string || !event_string) {
return NULL;
}
return ast_manager_event_blob_create(EVENT_FLAG_AGI, type,
"%s"
"%s",
ast_str_buffer(channel_string),
ast_str_buffer(event_string));
}
static struct ast_manager_event_blob *agi_exec_start_to_ami(struct stasis_message *message)
{
return agi_channel_to_ami("AGIExecStart", message);
}
static struct ast_manager_event_blob *agi_exec_end_to_ami(struct stasis_message *message)
{
return agi_channel_to_ami("AGIExecEnd", message);
}
static struct ast_manager_event_blob *agi_async_start_to_ami(struct stasis_message *message)
{
return agi_channel_to_ami("AsyncAGIStart", message);
}
static struct ast_manager_event_blob *agi_async_exec_to_ami(struct stasis_message *message)
{
return agi_channel_to_ami("AsyncAGIExec", message);
}
static struct ast_manager_event_blob *agi_async_end_to_ami(struct stasis_message *message)
{
return agi_channel_to_ami("AsyncAGIEnd", message);
}
STASIS_MESSAGE_TYPE_DEFN_LOCAL(agi_exec_start_type,
.to_ami = agi_exec_start_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN_LOCAL(agi_exec_end_type,
.to_ami = agi_exec_end_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN_LOCAL(agi_async_start_type,
.to_ami = agi_async_start_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN_LOCAL(agi_async_exec_type,
.to_ami = agi_async_exec_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN_LOCAL(agi_async_end_type,
.to_ami = agi_async_end_to_ami,
);
static agi_command *find_command(const char * const cmds[], int exact);
AST_THREADSTORAGE(agi_buf);
#define AGI_BUF_INITSIZE 256
int AST_OPTIONAL_API_NAME(ast_agi_send)(int fd, struct ast_channel *chan, char *fmt, ...)
{
int res = 0;
va_list ap;
struct ast_str *buf;
if (!(buf = ast_str_thread_get(&agi_buf, AGI_BUF_INITSIZE)))
return -1;
va_start(ap, fmt);
res = ast_str_set_va(&buf, 0, fmt, ap);
va_end(ap);
if (res == -1) {
ast_log(LOG_ERROR, "Out of memory\n");
return -1;
}
if (agidebug) {
if (chan) {
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
ast_verbose("<%s>AGI Tx >> %s", ast_channel_name(chan), ast_str_buffer(buf));
} else {
ast_verbose("AGI Tx >> %s", ast_str_buffer(buf));
}
}
return ast_carefulwrite(fd, ast_str_buffer(buf), ast_str_strlen(buf), 100);
}
/* linked list of AGI commands ready to be executed by Async AGI */
struct agi_cmd {
char *cmd_buffer;
char *cmd_id;
AST_LIST_ENTRY(agi_cmd) entry;
};
static void free_agi_cmd(struct agi_cmd *cmd)
{
ast_free(cmd->cmd_buffer);
ast_free(cmd->cmd_id);
ast_free(cmd);
}
/* AGI datastore destructor */
static void agi_destroy_commands_cb(void *data)
{
struct agi_cmd *cmd;
AST_LIST_HEAD(, agi_cmd) *chan_cmds = data;
AST_LIST_LOCK(chan_cmds);
while ( (cmd = AST_LIST_REMOVE_HEAD(chan_cmds, entry)) ) {
free_agi_cmd(cmd);
}
AST_LIST_UNLOCK(chan_cmds);
AST_LIST_HEAD_DESTROY(chan_cmds);
ast_free(chan_cmds);
}
/* channel datastore to keep the queue of AGI commands in the channel */
static const struct ast_datastore_info agi_commands_datastore_info = {
.type = "AsyncAGI",
.destroy = agi_destroy_commands_cb
};
/*!
* \brief Retrieve the list head to the requested channel's AGI datastore
* \param chan Channel datastore is requested for
* \param cmd Pointer to the struct pointer which will reference the head of the agi command list.
*
* \retval 0 if the datastore was valid and the list head was retrieved appropriately (even if it's
* NULL and the list is empty)
* \retval -1 if the datastore could not be retrieved causing an error
*/
static int get_agi_cmd(struct ast_channel *chan, struct agi_cmd **cmd)
{
struct ast_datastore *store;
AST_LIST_HEAD(, agi_cmd) *agi_commands;
ast_channel_lock(chan);
store = ast_channel_datastore_find(chan, &agi_commands_datastore_info, NULL);
ast_channel_unlock(chan);
if (!store) {
ast_log(LOG_ERROR, "Huh? Async AGI datastore disappeared on 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
ast_channel_name(chan));
*cmd = NULL;
return -1;
}
agi_commands = store->data;
AST_LIST_LOCK(agi_commands);
*cmd = AST_LIST_REMOVE_HEAD(agi_commands, entry);
AST_LIST_UNLOCK(agi_commands);
return 0;
}
/* channel is locked when calling this one either from the CLI or manager thread */
static int add_agi_cmd(struct ast_channel *chan, const char *cmd_buff, const char *cmd_id)
{
struct ast_datastore *store;
struct agi_cmd *cmd;
AST_LIST_HEAD(, agi_cmd) *agi_commands;
store = ast_channel_datastore_find(chan, &agi_commands_datastore_info, NULL);
if (!store) {
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
ast_log(LOG_WARNING, "Channel %s is not setup for Async AGI.\n", ast_channel_name(chan));
return -1;
}
agi_commands = store->data;
cmd = ast_calloc(1, sizeof(*cmd));
if (!cmd) {
return -1;
}
cmd->cmd_buffer = ast_strdup(cmd_buff);
if (!cmd->cmd_buffer) {
ast_free(cmd);
return -1;
}
cmd->cmd_id = ast_strdup(cmd_id);
if (!cmd->cmd_id) {
ast_free(cmd->cmd_buffer);
ast_free(cmd);
return -1;
}
AST_LIST_LOCK(agi_commands);
AST_LIST_INSERT_TAIL(agi_commands, cmd, entry);
AST_LIST_UNLOCK(agi_commands);
return 0;
}
static int add_to_agi(struct ast_channel *chan)
{
struct ast_datastore *datastore;
AST_LIST_HEAD(, agi_cmd) *agi_cmds_list;
/* check if already on AGI */
ast_channel_lock(chan);
datastore = ast_channel_datastore_find(chan, &agi_commands_datastore_info, NULL);
ast_channel_unlock(chan);
if (datastore) {
/* we already have an AGI datastore, let's just
return success */
return 0;
}
/* the channel has never been on Async AGI,
let's allocate it's datastore */
datastore = ast_datastore_alloc(&agi_commands_datastore_info, "AGI");
if (!datastore) {
return -1;
}
agi_cmds_list = ast_calloc(1, sizeof(*agi_cmds_list));
if (!agi_cmds_list) {
ast_log(LOG_ERROR, "Unable to allocate Async AGI commands list.\n");
ast_datastore_free(datastore);
return -1;
}
datastore->data = agi_cmds_list;
AST_LIST_HEAD_INIT(agi_cmds_list);
ast_channel_lock(chan);
ast_channel_datastore_add(chan, datastore);
ast_channel_unlock(chan);
return 0;
}
/*!
* \brief CLI command to add applications to execute in Async AGI
* \param e
* \param cmd
* \param a
*
* \retval CLI_SUCCESS on success
* \retval NULL when init or tab completion is used
*/
static char *handle_cli_agi_add_cmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_channel *chan;
switch (cmd) {
case CLI_INIT:
e->command = "agi exec";
e->usage = "Usage: agi exec <channel name> <app and arguments> [id]\n"
" Add AGI command to the execute queue of the specified channel in Async AGI\n";
return NULL;
case CLI_GENERATE:
if (a->pos == 2)
return ast_complete_channels(a->line, a->word, a->pos, a->n, 2);
return NULL;
}
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 (a->argc < 4) {
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(a->argv[2]))) {
ast_cli(a->fd, "Channel %s does not exist.\n", a->argv[2]);
return CLI_FAILURE;
}
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_channel_lock(chan);
if (add_agi_cmd(chan, a->argv[3], (a->argc > 4 ? a->argv[4] : ""))) {
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
ast_cli(a->fd, "Failed to add AGI command to queue of channel %s\n", ast_channel_name(chan));
ast_channel_unlock(chan);
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);
return CLI_FAILURE;
}
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
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
ast_debug(1, "Added AGI command to channel %s queue\n", ast_channel_name(chan));
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_channel_unlock(chan);
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);
return CLI_SUCCESS;
}
/*!
* \brief Add a new command to execute by the Async AGI application
* \param s
* \param m
*
* It will append the application to the specified channel's queue
* if the channel is not inside Async AGI application it will return an error
* \retval 0 on success or incorrect use
* \retval 1 on failure to add the command ( most likely because the channel
* is not in Async AGI loop )
*/
static int action_add_agi_cmd(struct mansession *s, const struct message *m)
{
const char *channel = astman_get_header(m, "Channel");
const char *cmdbuff = astman_get_header(m, "Command");
const char *cmdid = astman_get_header(m, "CommandID");
struct ast_channel *chan;
char buf[256];
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 (ast_strlen_zero(channel) || ast_strlen_zero(cmdbuff)) {
astman_send_error(s, m, "Both, Channel and Command are *required*");
return 0;
}
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(channel))) {
snprintf(buf, sizeof(buf), "Channel %s does not exist.", channel);
astman_send_error(s, m, buf);
return 0;
}
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_channel_lock(chan);
if (add_agi_cmd(chan, cmdbuff, cmdid)) {
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
snprintf(buf, sizeof(buf), "Failed to add AGI command to channel %s queue", ast_channel_name(chan));
astman_send_error(s, m, buf);
ast_channel_unlock(chan);
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);
return 0;
}
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_channel_unlock(chan);
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);
astman_send_ack(s, m, "Added AGI command to queue");
return 0;
}
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
static enum agi_result agi_handle_command(struct ast_channel *chan, AGI *agi, char *buf, int dead);
static void setup_env(struct ast_channel *chan, char *request, int fd, int enhanced, int argc, char *argv[]);
/*!
* \internal
* \brief Read and handle a channel frame for Async AGI.
*
* \param chan Channel to read a frame from.
*
* \retval AGI_RESULT_SUCCESS on success.
* \retval AGI_RESULT_HANGUP on hangup.
* \retval AGI_RESULT_FAILURE on error.
*/
static enum agi_result async_agi_read_frame(struct ast_channel *chan)
{
struct ast_frame *f;
f = ast_read(chan);
if (!f) {
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
ast_debug(3, "No frame read on channel %s, going out ...\n", ast_channel_name(chan));
return AGI_RESULT_HANGUP;
}
if (f->frametype == AST_FRAME_CONTROL) {
/*
* Is there any other frame we should care about besides
* AST_CONTROL_HANGUP?
*/
switch (f->subclass.integer) {
case AST_CONTROL_HANGUP:
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
ast_debug(3, "Got HANGUP frame on channel %s, going out ...\n", ast_channel_name(chan));
ast_frfree(f);
return AGI_RESULT_HANGUP;
default:
break;
}
}
ast_frfree(f);
return AGI_RESULT_SUCCESS;
}
static enum agi_result launch_asyncagi(struct ast_channel *chan, int argc, char *argv[], int *efd)
{
/* This buffer sizes might cause truncation if the AGI command writes more data
than AGI_BUF_SIZE as result. But let's be serious, is there an AGI command
that writes a response larger than 1024 bytes?, I don't think so, most of
them are just result=blah stuff. However probably if GET VARIABLE is called
and the variable has large amount of data, that could be a problem. We could
make this buffers dynamic, but let's leave that as a second step.
AMI_BUF_SIZE is twice AGI_BUF_SIZE just for the sake of choosing a safe
number. Some characters of AGI buf will be url encoded to be sent to manager
clients. An URL encoded character will take 3 bytes, but again, to cause
truncation more than about 70% of the AGI buffer should be URL encoded for
that to happen. Not likely at all.
On the other hand. I wonder if read() could eventually return less data than
the amount already available in the pipe? If so, how to deal with that?
So far, my tests on Linux have not had any problems.
*/
#define AGI_BUF_SIZE 1024
#define AMI_BUF_SIZE 2048
enum agi_result cmd_status;
struct agi_cmd *cmd;
int res;
int fds[2];
int hungup;
int timeout = 100;
char agi_buffer[AGI_BUF_SIZE + 1];
char ami_buffer[AMI_BUF_SIZE];
enum agi_result returnstatus = AGI_RESULT_SUCCESS;
AGI async_agi;
RAII_VAR(struct ast_json *, startblob, NULL, ast_json_unref);
if (efd) {
ast_log(LOG_WARNING, "Async AGI does not support Enhanced AGI yet\n");
return AGI_RESULT_FAILURE;
}
/* add AsyncAGI datastore to the channel */
if (add_to_agi(chan)) {
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
ast_log(LOG_ERROR, "Failed to start Async AGI on channel %s\n", ast_channel_name(chan));
return AGI_RESULT_FAILURE;
}
/* this pipe allows us to create a "fake" AGI struct to use
the AGI commands */
res = pipe(fds);
if (res) {
ast_log(LOG_ERROR, "Failed to create Async AGI pipe\n");
/*
* Intentionally do not remove the datastore added with
* add_to_agi() the from channel. It will be removed when the
* channel is hung up anyway.
*/
return AGI_RESULT_FAILURE;
}
/* handlers will get the pipe write fd and we read the AGI responses
from the pipe read fd */
async_agi.fd = fds[1];
async_agi.ctrl = fds[1];
async_agi.audio = -1; /* no audio support */
async_agi.fast = 0;
async_agi.speech = NULL;
/* notify possible manager users of a new channel ready to
receive commands */
setup_env(chan, "async", fds[1], 0, argc, argv);
/* read the environment */
res = read(fds[0], agi_buffer, AGI_BUF_SIZE);
if (res <= 0) {
ast_log(LOG_ERROR, "Failed to read from Async AGI pipe on channel %s: %s\n",
ast_channel_name(chan), res < 0 ? strerror(errno) : "EOF");
returnstatus = AGI_RESULT_FAILURE;
goto async_agi_abort;
}
agi_buffer[res] = '\0';
/* encode it and send it thru the manager so whoever is going to take
care of AGI commands on this channel can decide which AGI commands
to execute based on the setup info */
ast_uri_encode(agi_buffer, ami_buffer, AMI_BUF_SIZE, ast_uri_http);
startblob = ast_json_pack("{s: s}", "Env", ami_buffer);
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
ast_channel_publish_cached_blob(chan, agi_async_start_type(), startblob);
hungup = ast_check_hangup_locked(chan);
for (;;) {
/*
* Process as many commands as we can. Commands are added via
* the manager or the cli threads.
*/
while (!hungup) {
RAII_VAR(struct ast_json *, execblob, NULL, ast_json_unref);
res = get_agi_cmd(chan, &cmd);
if (res) {
returnstatus = AGI_RESULT_FAILURE;
goto async_agi_done;
} else if (!cmd) {
break;
}
/* OK, we have a command, let's call the command handler. */
cmd_status = agi_handle_command(chan, &async_agi, cmd->cmd_buffer, 0);
/*
* The command handler must have written to our fake AGI struct
* fd (the pipe), let's read the response.
*/
res = read(fds[0], agi_buffer, AGI_BUF_SIZE);
if (res <= 0) {
ast_log(LOG_ERROR, "Failed to read from Async AGI pipe on channel %s: %s\n",
ast_channel_name(chan), res < 0 ? strerror(errno) : "EOF");
free_agi_cmd(cmd);
returnstatus = AGI_RESULT_FAILURE;
goto async_agi_done;
}
/*
* We have a response, let's send the response thru the manager.
* Include the CommandID if it was specified when the command
* was added.
*/
agi_buffer[res] = '\0';
ast_uri_encode(agi_buffer, ami_buffer, AMI_BUF_SIZE, ast_uri_http);
execblob = ast_json_pack("{s: s}", "Result", ami_buffer);
if (execblob && !ast_strlen_zero(cmd->cmd_id)) {
ast_json_object_set(execblob, "CommandId", ast_json_string_create(cmd->cmd_id));
}
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
ast_channel_publish_cached_blob(chan, agi_async_exec_type(), execblob);
free_agi_cmd(cmd);
/*
* Check the command status to determine if we should continue
* executing more commands.
*/
hungup = ast_check_hangup(chan);
switch (cmd_status) {
case AGI_RESULT_FAILURE:
if (!hungup) {
/* The failure was not because of a hangup. */
returnstatus = AGI_RESULT_FAILURE;
goto async_agi_done;
}
break;
case AGI_RESULT_SUCCESS_ASYNC:
/* Only the "asyncagi break" command does this. */
returnstatus = AGI_RESULT_SUCCESS_ASYNC;
goto async_agi_done;
default:
break;
}
}
if (!hungup) {
/* Wait a bit for a frame to read or to poll for a new command. */
res = ast_waitfor(chan, timeout);
if (res < 0) {
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
ast_debug(1, "ast_waitfor returned <= 0 on chan %s\n", ast_channel_name(chan));
returnstatus = AGI_RESULT_FAILURE;
break;
}
} else {
/*
* Read the channel control queue until it is dry so we can
* quit.
*/
res = 1;
}
if (0 < res) {
do {
cmd_status = async_agi_read_frame(chan);
if (cmd_status != AGI_RESULT_SUCCESS) {
returnstatus = cmd_status;
goto async_agi_done;
}
hungup = ast_check_hangup(chan);
} while (hungup);
} else {
hungup = ast_check_hangup(chan);
}
}
async_agi_done:
if (async_agi.speech) {
ast_speech_destroy(async_agi.speech);
}
/* notify manager users this channel cannot be controlled anymore by Async AGI */
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
ast_channel_publish_cached_blob(chan, agi_async_end_type(), NULL);
async_agi_abort:
/* close the pipe */
close(fds[0]);
close(fds[1]);
/*
* Intentionally do not remove the datastore added with
* add_to_agi() the from channel. There might be commands still
* in the queue or in-flight to us and AsyncAGI may get called
* again. The datastore destructor will be called on channel
* destruction anyway.
*/
if (returnstatus == AGI_RESULT_SUCCESS) {
returnstatus = AGI_RESULT_SUCCESS_ASYNC;
}
return returnstatus;
#undef AGI_BUF_SIZE
#undef AMI_BUF_SIZE
}
Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22 19:40:02 +00:00
/*!
* \internal
* \brief Handle the connection that was started by launch_netscript.
*
* \param agiurl Url that we are trying to connect to.
* \param addr Address that host was resolved to.
* \param netsockfd File descriptor of socket.
*
* \retval 0 when connection is succesful.
* \retval 1 when there is an error.
*/
static int handle_connection(const char *agiurl, const struct ast_sockaddr addr, const int netsockfd)
{
struct pollfd pfds[1];
int res, conresult;
socklen_t reslen;
reslen = sizeof(conresult);
pfds[0].fd = netsockfd;
pfds[0].events = POLLOUT;
while ((res = ast_poll(pfds, 1, MAX_AGI_CONNECT)) != 1) {
if (errno != EINTR) {
if (!res) {
ast_log(LOG_WARNING, "FastAGI connection to '%s' timed out after MAX_AGI_CONNECT (%d) milliseconds.\n",
agiurl, MAX_AGI_CONNECT);
} else {
ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", agiurl, strerror(errno));
}
return 1;
}
}
if (getsockopt(pfds[0].fd, SOL_SOCKET, SO_ERROR, &conresult, &reslen) < 0) {
ast_log(LOG_WARNING, "Connection to %s failed with error: %s\n",
ast_sockaddr_stringify(&addr), strerror(errno));
return 1;
}
if (conresult) {
ast_log(LOG_WARNING, "Connecting to '%s' failed for url '%s': %s\n",
ast_sockaddr_stringify(&addr), agiurl, strerror(conresult));
return 1;
}
return 0;
}
/* launch_netscript: The fastagi handler.
FastAGI defaults to port 4573 */
static enum agi_result launch_netscript(char *agiurl, char *argv[], int *fds)
{
int s = 0;
char *host, *script;
int num_addrs = 0, i = 0;
struct ast_sockaddr *addrs;
/* agiurl is "agi://host.domain[:port][/script/name]" */
host = ast_strdupa(agiurl + 6); /* Remove agi:// */
/* Strip off any script name */
if ((script = strchr(host, '/'))) {
*script++ = '\0';
} else {
script = "";
}
if (!(num_addrs = ast_sockaddr_resolve(&addrs, host, 0, AST_AF_UNSPEC))) {
ast_log(LOG_WARNING, "Unable to locate host '%s'\n", host);
return AGI_RESULT_FAILURE;
}
for (i = 0; i < num_addrs; i++) {
if (!ast_sockaddr_port(&addrs[i])) {
ast_sockaddr_set_port(&addrs[i], AGI_PORT);
}
if ((s = socket(addrs[i].ss.ss_family, SOCK_STREAM, IPPROTO_TCP)) < 0) {
ast_log(LOG_WARNING, "Unable to create socket: %s\n", strerror(errno));
continue;
}
if (ast_fd_set_flags(s, O_NONBLOCK)) {
close(s);
continue;
}
Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22 19:40:02 +00:00
if (ast_connect(s, &addrs[i]) && errno == EINPROGRESS) {
if (handle_connection(agiurl, addrs[i], s)) {
close(s);
continue;
}
} else {
ast_log(LOG_WARNING, "Connection to %s failed with unexpected error: %s\n",
Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22 19:40:02 +00:00
ast_sockaddr_stringify(&addrs[i]), strerror(errno));
}
break;
}
ast_free(addrs);
if (i == num_addrs) {
ast_log(LOG_WARNING, "Couldn't connect to any host. FastAGI failed.\n");
return AGI_RESULT_FAILURE;
}
if (ast_agi_send(s, NULL, "agi_network: yes\n") < 0) {
if (errno != EINTR) {
ast_log(LOG_WARNING, "Connect to '%s' failed: %s\n", agiurl, strerror(errno));
close(s);
return AGI_RESULT_FAILURE;
}
}
/* If we have a script parameter, relay it to the fastagi server */
/* Script parameters take the form of: AGI(agi://my.example.com/?extension=${EXTEN}) */
Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22 19:40:02 +00:00
if (!ast_strlen_zero(script)) {
ast_agi_send(s, NULL, "agi_network_script: %s\n", script);
Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ ........ Merged revisions 381893 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@381894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-02-22 19:40:02 +00:00
}
ast_debug(4, "Wow, connected!\n");
fds[0] = s;
fds[1] = s;
return AGI_RESULT_SUCCESS_FAST;
}
/*!
* \internal
* \brief The HA fastagi handler.
* \param agiurl The request URL as passed to Agi() in the dial plan
* \param argv The parameters after the URL passed to Agi() in the dial plan
* \param fds Input/output file descriptors
*
* Uses SRV lookups to try to connect to a list of FastAGI servers. The hostname in
* the URI is prefixed with _agi._tcp. prior to the DNS resolution. For
* example, if you specify the URI \a hagi://agi.example.com/foo.agi the DNS
* query would be for \a _agi._tcp.agi.example.com and you'll need to make sure
* this resolves.
*
* This function parses the URI, resolves the SRV service name, forms new URIs
* with the results of the DNS lookup, and then calls launch_netscript on the
* new URIs until one succeeds.
*
* \return the result of the AGI operation.
*/
static enum agi_result launch_ha_netscript(char *agiurl, char *argv[], int *fds)
{
char *host, *script;
enum agi_result result;
struct srv_context *context = NULL;
int srv_ret;
char service[256];
char resolved_uri[1024];
const char *srvhost;
unsigned short srvport;
/* format of agiurl is "hagi://host.domain[:port][/script/name]" */
if (strlen(agiurl) < 7) { /* Remove hagi:// */
ast_log(LOG_WARNING, "An error occurred parsing the AGI URI: %s", agiurl);
return AGI_RESULT_FAILURE;
}
host = ast_strdupa(agiurl + 7);
/* Strip off any script name */
if ((script = strchr(host, '/'))) {
*script++ = '\0';
} else {
script = "";
}
if (strchr(host, ':')) {
ast_log(LOG_WARNING, "Specifying a port number disables SRV lookups: %s\n", agiurl);
return launch_netscript(agiurl + 1, argv, fds); /* +1 to strip off leading h from hagi:// */
}
snprintf(service, sizeof(service), "%s%s", SRV_PREFIX, host);
while (!(srv_ret = ast_srv_lookup(&context, service, &srvhost, &srvport))) {
snprintf(resolved_uri, sizeof(resolved_uri), "agi://%s:%d/%s", srvhost, srvport, script);
result = launch_netscript(resolved_uri, argv, fds);
if (result == AGI_RESULT_FAILURE || result == AGI_RESULT_NOTFOUND) {
ast_log(LOG_WARNING, "AGI request failed for host '%s' (%s:%d)\n", host, srvhost, srvport);
} else {
/* The script launched so we must cleanup the context. */
ast_srv_cleanup(&context);
return result;
}
}
/*
* The DNS SRV lookup failed or we ran out of servers to check.
* ast_srv_lookup() has already cleaned up the context for us.
*/
if (srv_ret < 0) {
ast_log(LOG_WARNING, "SRV lookup failed for %s\n", agiurl);
}
return AGI_RESULT_FAILURE;
}
static enum agi_result launch_script(struct ast_channel *chan, char *script, int argc, char *argv[], int *fds, int *efd, int *opid)
{
char tmp[256];
int pid, toast[2], fromast[2], audio[2], res;
struct stat st;
if (!strncasecmp(script, "agi://", 6)) {
return (efd == NULL) ? launch_netscript(script, argv, fds) : AGI_RESULT_FAILURE;
}
if (!strncasecmp(script, "hagi://", 7)) {
return (efd == NULL) ? launch_ha_netscript(script, argv, fds) : AGI_RESULT_FAILURE;
}
if (!strncasecmp(script, "agi:async", sizeof("agi:async") - 1)) {
return launch_asyncagi(chan, argc, argv, efd);
}
if (script[0] != '/') {
snprintf(tmp, sizeof(tmp), "%s/%s", ast_config_AST_AGI_DIR, script);
script = tmp;
}
/* Before even trying let's see if the file actually exists */
if (stat(script, &st)) {
ast_log(LOG_WARNING, "Failed to execute '%s': File does not exist.\n", script);
return AGI_RESULT_NOTFOUND;
}
if (pipe(toast)) {
ast_log(LOG_WARNING, "Unable to create toast pipe: %s\n",strerror(errno));
return AGI_RESULT_FAILURE;
}
if (pipe(fromast)) {
ast_log(LOG_WARNING, "unable to create fromast pipe: %s\n", strerror(errno));
close(toast[0]);
close(toast[1]);
return AGI_RESULT_FAILURE;
}
if (efd) {
if (pipe(audio)) {
ast_log(LOG_WARNING, "unable to create audio pipe: %s\n", strerror(errno));
close(fromast[0]);
close(fromast[1]);
close(toast[0]);
close(toast[1]);
return AGI_RESULT_FAILURE;
}
res = ast_fd_set_flags(audio[1], O_NONBLOCK);
if (res < 0) {
ast_log(LOG_WARNING, "unable to set audio pipe parameters: %s\n", strerror(errno));
close(fromast[0]);
close(fromast[1]);
close(toast[0]);
close(toast[1]);
close(audio[0]);
close(audio[1]);
return AGI_RESULT_FAILURE;
}
}
if ((pid = ast_safe_fork(1)) < 0) {
ast_log(LOG_WARNING, "Failed to fork(): %s\n", strerror(errno));
return AGI_RESULT_FAILURE;
}
if (!pid) {
/* Pass paths to AGI via environmental variables */
setenv("AST_CONFIG_DIR", ast_config_AST_CONFIG_DIR, 1);
setenv("AST_CONFIG_FILE", ast_config_AST_CONFIG_FILE, 1);
setenv("AST_MODULE_DIR", ast_config_AST_MODULE_DIR, 1);
setenv("AST_SPOOL_DIR", ast_config_AST_SPOOL_DIR, 1);
setenv("AST_MONITOR_DIR", ast_config_AST_MONITOR_DIR, 1);
setenv("AST_VAR_DIR", ast_config_AST_VAR_DIR, 1);
setenv("AST_DATA_DIR", ast_config_AST_DATA_DIR, 1);
setenv("AST_LOG_DIR", ast_config_AST_LOG_DIR, 1);
setenv("AST_AGI_DIR", ast_config_AST_AGI_DIR, 1);
setenv("AST_KEY_DIR", ast_config_AST_KEY_DIR, 1);
setenv("AST_RUN_DIR", ast_config_AST_RUN_DIR, 1);
/* Don't run AGI scripts with realtime priority -- it causes audio stutter */
ast_set_priority(0);
/* Redirect stdin and out, provide enhanced audio channel if desired */
dup2(fromast[0], STDIN_FILENO);
dup2(toast[1], STDOUT_FILENO);
if (efd)
dup2(audio[0], STDERR_FILENO + 1);
else
close(STDERR_FILENO + 1);
/* Close everything but stdin/out/error */
ast_close_fds_above_n(STDERR_FILENO + 1);
/* Execute script */
/* XXX argv should be deprecated in favor of passing agi_argX paramaters */
execv(script, argv);
/* Can't use ast_log since FD's are closed */
ast_child_verbose(1, "Failed to execute '%s': %s", script, strerror(errno));
/* Special case to set status of AGI to failure */
fprintf(stdout, "failure\n");
fflush(stdout);
_exit(1);
}
ast_verb(3, "Launched AGI Script %s\n", script);
fds[0] = toast[0];
fds[1] = fromast[1];
if (efd)
*efd = audio[1];
/* close what we're not using in the parent */
close(toast[1]);
close(fromast[0]);
if (efd)
close(audio[0]);
*opid = pid;
return AGI_RESULT_SUCCESS;
}
static void setup_env(struct ast_channel *chan, char *request, int fd, int enhanced, int argc, char *argv[])
{
int count;
/* Print initial environment, with agi_request always being the first
thing */
ast_agi_send(fd, chan, "agi_request: %s\n", request);
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
ast_agi_send(fd, chan, "agi_channel: %s\n", ast_channel_name(chan));
ast_agi_send(fd, chan, "agi_language: %s\n", ast_channel_language(chan));
ast_agi_send(fd, chan, "agi_type: %s\n", ast_channel_tech(chan)->type);
ast_agi_send(fd, chan, "agi_uniqueid: %s\n", ast_channel_uniqueid(chan));
ast_agi_send(fd, chan, "agi_version: %s\n", ast_get_version());
/* ANI/DNIS */
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_agi_send(fd, chan, "agi_callerid: %s\n",
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, "unknown"));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_agi_send(fd, chan, "agi_calleridname: %s\n",
S_COR(ast_channel_caller(chan)->id.name.valid, ast_channel_caller(chan)->id.name.str, "unknown"));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_agi_send(fd, chan, "agi_callingpres: %d\n",
ast_party_id_presentation(&ast_channel_caller(chan)->id));
ast_agi_send(fd, chan, "agi_callingani2: %d\n", ast_channel_caller(chan)->ani2);
ast_agi_send(fd, chan, "agi_callington: %d\n", ast_channel_caller(chan)->id.number.plan);
ast_agi_send(fd, chan, "agi_callingtns: %d\n", ast_channel_dialed(chan)->transit_network_select);
ast_agi_send(fd, chan, "agi_dnid: %s\n", S_OR(ast_channel_dialed(chan)->number.str, "unknown"));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
ast_agi_send(fd, chan, "agi_rdnis: %s\n",
S_COR(ast_channel_redirecting(chan)->from.number.valid, ast_channel_redirecting(chan)->from.number.str, "unknown"));
/* Context information */
ast_agi_send(fd, chan, "agi_context: %s\n", ast_channel_context(chan));
ast_agi_send(fd, chan, "agi_extension: %s\n", ast_channel_exten(chan));
ast_agi_send(fd, chan, "agi_priority: %d\n", ast_channel_priority(chan));
ast_agi_send(fd, chan, "agi_enhanced: %s\n", enhanced ? "1.0" : "0.0");
/* User information */
ast_agi_send(fd, chan, "agi_accountcode: %s\n", ast_channel_accountcode(chan) ? ast_channel_accountcode(chan) : "");
ast_agi_send(fd, chan, "agi_threadid: %ld\n", (long)pthread_self());
/* Send any parameters to the fastagi server that have been passed via the agi application */
/* Agi application paramaters take the form of: AGI(/path/to/example/script|${EXTEN}) */
for(count = 1; count < argc; count++)
ast_agi_send(fd, chan, "agi_arg_%d: %s\n", count, argv[count]);
/* End with empty return */
ast_agi_send(fd, chan, "\n");
}
static int handle_answer(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res = 0;
/* Answer the channel */
if (ast_channel_state(chan) != AST_STATE_UP)
res = ast_answer(chan);
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_asyncagi_break(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
ast_agi_send(agi->fd, chan, "200 result=0\n");
return ASYNC_AGI_BREAK;
}
static int handle_waitfordigit(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, to;
if (argc != 4)
return RESULT_SHOWUSAGE;
if (sscanf(argv[3], "%30d", &to) != 1)
return RESULT_SHOWUSAGE;
res = ast_waitfordigit_full(chan, to, NULL, agi->audio, agi->ctrl);
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_sendtext(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 3)
return RESULT_SHOWUSAGE;
/* At the moment, the parser (perhaps broken) returns with
the last argument PLUS the newline at the end of the input
buffer. This probably needs to be fixed, but I wont do that
because other stuff may break as a result. The right way
would probably be to strip off the trailing newline before
parsing, then here, add a newline at the end of the string
before sending it to ast_sendtext --DUDE */
res = ast_sendtext(chan, argv[2]);
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_recvchar(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 3)
return RESULT_SHOWUSAGE;
res = ast_recvchar(chan,atoi(argv[2]));
if (res == 0) {
ast_agi_send(agi->fd, chan, "200 result=%d (timeout)\n", res);
return RESULT_SUCCESS;
}
if (res > 0) {
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return RESULT_SUCCESS;
}
ast_agi_send(agi->fd, chan, "200 result=%d (hangup)\n", res);
return RESULT_FAILURE;
}
static int handle_recvtext(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
char *buf;
if (argc != 3)
return RESULT_SHOWUSAGE;
buf = ast_recvtext(chan, atoi(argv[2]));
if (buf) {
ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", buf);
ast_free(buf);
} else {
ast_agi_send(agi->fd, chan, "200 result=-1\n");
}
return RESULT_SUCCESS;
}
static int handle_tddmode(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, x;
if (argc != 3)
return RESULT_SHOWUSAGE;
if (!strncasecmp(argv[2],"on",2)) {
x = 1;
} else {
x = 0;
}
if (!strncasecmp(argv[2],"mate",4)) {
x = 2;
}
if (!strncasecmp(argv[2],"tdd",3)) {
x = 1;
}
res = ast_channel_setoption(chan, AST_OPTION_TDD, &x, sizeof(char), 0);
if (res) {
/* Set channel option failed */
ast_agi_send(agi->fd, chan, "200 result=0\n");
} else {
ast_agi_send(agi->fd, chan, "200 result=1\n");
}
return RESULT_SUCCESS;
}
static int handle_sendimage(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 3) {
return RESULT_SHOWUSAGE;
}
res = ast_send_image(chan, argv[2]);
if (!ast_check_hangup(chan)) {
res = 0;
}
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_controlstreamfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res = 0, skipms = 3000;
const char *fwd = "#", *rev = "*", *suspend = NULL, *stop = NULL; /* Default values */
char stopkeybuf[2];
long offsetms = 0;
char offsetbuf[20];
if (argc < 5 || argc > 10) {
return RESULT_SHOWUSAGE;
}
if (!ast_strlen_zero(argv[4])) {
stop = argv[4];
}
if ((argc > 5) && (sscanf(argv[5], "%30d", &skipms) != 1)) {
return RESULT_SHOWUSAGE;
}
if (argc > 6 && !ast_strlen_zero(argv[6])) {
fwd = argv[6];
}
if (argc > 7 && !ast_strlen_zero(argv[7])) {
rev = argv[7];
}
if (argc > 8 && !ast_strlen_zero(argv[8])) {
suspend = argv[8];
}
if (argc > 9 && (sscanf(argv[9], "%30ld", &offsetms) != 1)) {
return RESULT_SHOWUSAGE;
}
res = ast_control_streamfile(chan, argv[3], fwd, rev, stop, suspend, NULL, skipms, &offsetms);
/* If we stopped on one of our stop keys, return 0 */
if (res > 0 && stop && strchr(stop, res)) {
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "USERSTOPPED");
snprintf(stopkeybuf, sizeof(stopkeybuf), "%c", res);
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTOPKEY", stopkeybuf);
} else if (res > 0 && res == AST_CONTROL_STREAM_STOP) {
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "REMOTESTOPPED");
res = 0;
} else {
if (res < 0) {
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "ERROR");
} else {
pbx_builtin_setvar_helper(chan, "CPLAYBACKSTATUS", "SUCCESS");
}
}
snprintf(offsetbuf, sizeof(offsetbuf), "%ld", offsetms);
pbx_builtin_setvar_helper(chan, "CPLAYBACKOFFSET", offsetbuf);
ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res, offsetms);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_streamfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
struct ast_filestream *fs, *vfs;
long sample_offset = 0, max_length;
const char *edigits = "";
if (argc < 4 || argc > 5) {
return RESULT_SHOWUSAGE;
}
if (argv[3]) {
edigits = argv[3];
}
if ((argc > 4) && (sscanf(argv[4], "%30ld", &sample_offset) != 1)) {
return RESULT_SHOWUSAGE;
}
if (!(fs = ast_openstream(chan, argv[2], ast_channel_language(chan)))) {
ast_agi_send(agi->fd, chan, "200 result=-1 endpos=%ld\n", sample_offset);
return RESULT_FAILURE;
}
if ((vfs = ast_openvstream(chan, argv[2], ast_channel_language(chan)))) {
ast_debug(1, "Ooh, found a video stream, too\n");
}
ast_verb(3, "<%s> Playing '%s.%s' (escape_digits=%s) (sample_offset %ld) (language '%s')\n",
ast_channel_name(chan), argv[2], ast_format_get_name(ast_channel_writeformat(chan)),
edigits, sample_offset, S_OR(ast_channel_language(chan), "default"));
ast_seekstream(fs, 0, SEEK_END);
max_length = ast_tellstream(fs);
ast_seekstream(fs, sample_offset, SEEK_SET);
res = ast_applystream(chan, fs);
if (vfs) {
ast_applystream(chan, vfs);
}
ast_playstream(fs);
if (vfs) {
ast_playstream(vfs);
}
res = ast_waitstream_full(chan, argv[3], agi->audio, agi->ctrl);
/* this is to check for if ast_waitstream closed the stream, we probably are at
* the end of the stream, return that amount, else check for the amount */
sample_offset = (ast_channel_stream(chan)) ? ast_tellstream(fs) : max_length;
ast_stopstream(chan);
if (res == 1) {
/* Stop this command, don't print a result line, as there is a new command */
return RESULT_SUCCESS;
}
ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res, sample_offset);
pbx_builtin_setvar_helper(chan, "PLAYBACKSTATUS", res ? "FAILED" : "SUCCESS");
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
/*! \brief get option - really similar to the handle_streamfile, but with a timeout */
static int handle_getoption(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
struct ast_filestream *fs, *vfs;
long sample_offset = 0, max_length;
int timeout = 0;
const char *edigits = "";
if ( argc < 4 || argc > 5 )
return RESULT_SHOWUSAGE;
if ( argv[3] )
edigits = argv[3];
if ( argc == 5 )
timeout = atoi(argv[4]);
else if (ast_channel_pbx(chan)->dtimeoutms) {
/* by default dtimeout is set to 5sec */
timeout = ast_channel_pbx(chan)->dtimeoutms; /* in msec */
}
if (!(fs = ast_openstream(chan, argv[2], ast_channel_language(chan)))) {
ast_agi_send(agi->fd, chan, "200 result=-1 endpos=%ld\n", sample_offset);
ast_log(LOG_WARNING, "Unable to open %s\n", argv[2]);
return RESULT_FAILURE;
}
if ((vfs = ast_openvstream(chan, argv[2], ast_channel_language(chan))))
ast_debug(1, "Ooh, found a video stream, too\n");
ast_verb(3, "Playing '%s' (escape_digits=%s) (timeout %d)\n", argv[2], edigits, timeout);
ast_seekstream(fs, 0, SEEK_END);
max_length = ast_tellstream(fs);
ast_seekstream(fs, sample_offset, SEEK_SET);
res = ast_applystream(chan, fs);
if (vfs)
ast_applystream(chan, vfs);
ast_playstream(fs);
if (vfs)
ast_playstream(vfs);
res = ast_waitstream_full(chan, argv[3], agi->audio, agi->ctrl);
/* this is to check for if ast_waitstream closed the stream, we probably are at
* the end of the stream, return that amount, else check for the amount */
sample_offset = (ast_channel_stream(chan))?ast_tellstream(fs):max_length;
ast_stopstream(chan);
if (res == 1) {
/* Stop this command, don't print a result line, as there is a new command */
return RESULT_SUCCESS;
}
/* If the user didnt press a key, wait for digitTimeout*/
if (res == 0 ) {
res = ast_waitfordigit_full(chan, timeout, NULL, agi->audio, agi->ctrl);
/* Make sure the new result is in the escape digits of the GET OPTION */
if ( !strchr(edigits,res) )
res=0;
}
ast_agi_send(agi->fd, chan, "200 result=%d endpos=%ld\n", res, sample_offset);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
/*! \brief Say number in various language syntaxes */
/* While waiting, we're sending a NULL. */
static int handle_saynumber(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, num;
if (argc < 4 || argc > 5)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30d", &num) != 1)
return RESULT_SHOWUSAGE;
res = ast_say_number_full(chan, num, argv[3], ast_channel_language(chan), argc > 4 ? argv[4] : NULL, agi->audio, agi->ctrl);
if (res == 1)
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_saydigits(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, num;
if (argc != 4)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30d", &num) != 1)
return RESULT_SHOWUSAGE;
res = ast_say_digit_str_full(chan, argv[2], argv[3], ast_channel_language(chan), agi->audio, agi->ctrl);
if (res == 1) /* New command */
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_sayalpha(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
int sensitivity = AST_SAY_CASE_NONE;
if (argc < 4 || argc > 5) {
return RESULT_SHOWUSAGE;
}
if (argc > 4) {
switch (argv[4][0]) {
case 'a':
case 'A':
sensitivity = AST_SAY_CASE_ALL;
break;
case 'l':
case 'L':
sensitivity = AST_SAY_CASE_LOWER;
break;
case 'n':
case 'N':
sensitivity = AST_SAY_CASE_NONE;
break;
case 'u':
case 'U':
sensitivity = AST_SAY_CASE_UPPER;
break;
case '\0':
break;
default:
return RESULT_SHOWUSAGE;
}
}
res = ast_say_character_str_full(chan, argv[2], argv[3], ast_channel_language(chan), sensitivity, agi->audio, agi->ctrl);
if (res == 1) /* New command */
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_saydate(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, num;
if (argc != 4)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30d", &num) != 1)
return RESULT_SHOWUSAGE;
res = ast_say_date(chan, num, argv[3], ast_channel_language(chan));
if (res == 1)
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_saytime(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, num;
if (argc != 4)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30d", &num) != 1)
return RESULT_SHOWUSAGE;
res = ast_say_time(chan, num, argv[3], ast_channel_language(chan));
if (res == 1)
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_saydatetime(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res = 0;
time_t unixtime;
const char *format, *zone = NULL;
if (argc < 4)
return RESULT_SHOWUSAGE;
if (argc > 4) {
format = argv[4];
} else {
/* XXX this doesn't belong here, but in the 'say' module */
if (!strcasecmp(ast_channel_language(chan), "de")) {
format = "A dBY HMS";
} else {
format = "ABdY 'digits/at' IMp";
}
}
if (argc > 5 && !ast_strlen_zero(argv[5]))
zone = argv[5];
if (ast_get_time_t(argv[2], &unixtime, 0, NULL))
return RESULT_SHOWUSAGE;
res = ast_say_date_with_format(chan, unixtime, argv[3], ast_channel_language(chan), format, zone);
if (res == 1)
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_sayphonetic(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 4)
return RESULT_SHOWUSAGE;
res = ast_say_phonetic_str_full(chan, argv[2], argv[3], ast_channel_language(chan), agi->audio, agi->ctrl);
if (res == 1) /* New command */
return RESULT_SUCCESS;
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return (res >= 0) ? RESULT_SUCCESS : RESULT_FAILURE;
}
static int handle_getdata(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, max, timeout;
char data[1024];
if (argc < 3)
return RESULT_SHOWUSAGE;
if (argc >= 4)
timeout = atoi(argv[3]);
else
timeout = 0;
if (argc >= 5)
max = atoi(argv[4]);
else
max = 1024;
res = ast_app_getdata_full(chan, argv[2], data, max, timeout, agi->audio, agi->ctrl);
if (res == 2) /* New command */
return RESULT_SUCCESS;
else if (res == 1)
ast_agi_send(agi->fd, chan, "200 result=%s (timeout)\n", data);
else if (res < 0 )
ast_agi_send(agi->fd, chan, "200 result=-1\n");
else
ast_agi_send(agi->fd, chan, "200 result=%s\n", data);
return RESULT_SUCCESS;
}
static int handle_setcontext(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 3)
return RESULT_SHOWUSAGE;
ast_channel_context_set(chan, argv[2]);
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_setextension(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 3)
return RESULT_SHOWUSAGE;
ast_channel_exten_set(chan, argv[2]);
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_setpriority(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int pri;
if (argc != 3)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30d", &pri) != 1) {
pri = ast_findlabel_extension(chan, ast_channel_context(chan), ast_channel_exten(chan), argv[2],
S_COR(ast_channel_caller(chan)->id.number.valid, ast_channel_caller(chan)->id.number.str, NULL));
ast_callerid restructuring The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14 15:48:36 +00:00
if (pri < 1)
return RESULT_SHOWUSAGE;
}
ast_explicit_goto(chan, NULL, NULL, pri);
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
struct ast_filestream *fs;
struct ast_frame *f;
struct timeval start;
long sample_offset = 0;
int res = 0;
int ms;
struct ast_dsp *sildet=NULL; /* silence detector dsp */
int totalsilence = 0;
int dspsilence = 0;
int silence = 0; /* amount of silence to allow */
int gotsilence = 0; /* did we timeout for silence? */
char *silencestr = 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
RAII_VAR(struct ast_format *, rfmt, NULL, ao2_cleanup);
/* XXX EAGI FIXME XXX */
if (argc < 6)
return RESULT_SHOWUSAGE;
if (sscanf(argv[5], "%30d", &ms) != 1)
return RESULT_SHOWUSAGE;
if (argc > 6)
silencestr = strchr(argv[6],'s');
if ((argc > 7) && (!silencestr))
silencestr = strchr(argv[7],'s');
if ((argc > 8) && (!silencestr))
silencestr = strchr(argv[8],'s');
if (silencestr) {
if (strlen(silencestr) > 2) {
if ((silencestr[0] == 's') && (silencestr[1] == '=')) {
silencestr++;
silencestr++;
if (silencestr)
silence = atoi(silencestr);
if (silence > 0)
silence *= 1000;
}
}
}
if (silence > 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
rfmt = ao2_bump(ast_channel_readformat(chan));
res = ast_set_read_format(chan, ast_format_slin);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set to linear mode, giving up\n");
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
return RESULT_FAILURE;
}
sildet = ast_dsp_new();
if (!sildet) {
ast_log(LOG_WARNING, "Unable to create silence detector :(\n");
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
ast_agi_send(agi->fd, chan, "200 result=-1\n");
return RESULT_FAILURE;
}
ast_dsp_set_threshold(sildet, ast_dsp_get_threshold_from_settings(THRESHOLD_SILENCE));
}
/* backward compatibility, if no offset given, arg[6] would have been
* caught below and taken to be a beep, else if it is a digit then it is a
* offset */
if ((argc >6) && (sscanf(argv[6], "%30ld", &sample_offset) != 1) && (!strchr(argv[6], '=')))
res = ast_streamfile(chan, "beep", ast_channel_language(chan));
if ((argc > 7) && (!strchr(argv[7], '=')))
res = ast_streamfile(chan, "beep", ast_channel_language(chan));
if (!res)
res = ast_waitstream(chan, argv[4]);
if (res) {
ast_agi_send(agi->fd, chan, "200 result=%d (randomerror) endpos=%ld\n", res, sample_offset);
} else {
fs = ast_writefile(argv[2], argv[3], NULL, O_CREAT | O_WRONLY | (sample_offset ? O_APPEND : 0), 0, AST_FILE_MODE);
if (!fs) {
res = -1;
ast_agi_send(agi->fd, chan, "200 result=%d (writefile)\n", res);
if (sildet)
ast_dsp_free(sildet);
return RESULT_FAILURE;
}
/* Request a video update */
ast_indicate(chan, AST_CONTROL_VIDUPDATE);
ast_channel_stream_set(chan, fs);
ast_applystream(chan,fs);
/* really should have checks */
ast_seekstream(fs, sample_offset, SEEK_SET);
ast_truncstream(fs);
start = ast_tvnow();
while ((ms < 0) || ast_tvdiff_ms(ast_tvnow(), start) < ms) {
res = ast_waitfor(chan, ms - ast_tvdiff_ms(ast_tvnow(), start));
if (res < 0) {
ast_closestream(fs);
ast_agi_send(agi->fd, chan, "200 result=%d (waitfor) endpos=%ld\n", res,sample_offset);
if (sildet)
ast_dsp_free(sildet);
return RESULT_FAILURE;
}
f = ast_read(chan);
if (!f) {
ast_closestream(fs);
ast_agi_send(agi->fd, chan, "200 result=%d (hangup) endpos=%ld\n", -1, sample_offset);
if (sildet)
ast_dsp_free(sildet);
return RESULT_FAILURE;
}
switch(f->frametype) {
case AST_FRAME_DTMF:
if (strchr(argv[4], f->subclass.integer)) {
/* This is an interrupting chracter, so rewind to chop off any small
amount of DTMF that may have been recorded
*/
ast_stream_rewind(fs, 200);
ast_truncstream(fs);
sample_offset = ast_tellstream(fs);
ast_closestream(fs);
ast_agi_send(agi->fd, chan, "200 result=%d (dtmf) endpos=%ld\n", f->subclass.integer, sample_offset);
ast_frfree(f);
if (sildet)
ast_dsp_free(sildet);
return RESULT_SUCCESS;
}
break;
case AST_FRAME_VOICE:
ast_writestream(fs, f);
/* this is a safe place to check progress since we know that fs
* is valid after a write, and it will then have our current
* location */
sample_offset = ast_tellstream(fs);
if (silence > 0) {
dspsilence = 0;
ast_dsp_silence(sildet, f, &dspsilence);
if (dspsilence) {
totalsilence = dspsilence;
} else {
totalsilence = 0;
}
if (totalsilence > silence) {
/* Ended happily with silence */
gotsilence = 1;
break;
}
}
break;
case AST_FRAME_VIDEO:
ast_writestream(fs, f);
default:
/* Ignore all other frames */
break;
}
ast_frfree(f);
if (gotsilence)
break;
}
if (gotsilence) {
ast_stream_rewind(fs, silence-1000);
ast_truncstream(fs);
sample_offset = ast_tellstream(fs);
}
ast_closestream(fs);
ast_agi_send(agi->fd, chan, "200 result=%d (timeout) endpos=%ld\n", res, sample_offset);
}
if (silence > 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
res = ast_set_read_format(chan, rfmt);
if (res)
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
ast_log(LOG_WARNING, "Unable to restore read format on '%s'\n", ast_channel_name(chan));
ast_dsp_free(sildet);
}
return RESULT_SUCCESS;
}
static int handle_autohangup(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
double timeout;
struct timeval whentohangup = { 0, 0 };
if (argc != 3)
return RESULT_SHOWUSAGE;
if (sscanf(argv[2], "%30lf", &timeout) != 1)
return RESULT_SHOWUSAGE;
if (timeout < 0)
timeout = 0;
if (timeout) {
whentohangup.tv_sec = timeout;
whentohangup.tv_usec = (timeout - whentohangup.tv_sec) * 1000000.0;
}
ast_channel_lock(chan);
ast_channel_setwhentohangup_tv(chan, whentohangup);
ast_channel_unlock(chan);
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_hangup(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
struct ast_channel *c;
if (argc == 1) {
/* no argument: hangup the current channel */
ast_set_hangupsource(chan, "dialplan/agi", 0);
ast_softhangup(chan,AST_SOFTHANGUP_EXPLICIT);
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
} else if (argc == 2) {
/* one argument: look for info on the specified channel */
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 ((c = ast_channel_get_by_name(argv[1]))) {
/* we have a matching channel */
ast_set_hangupsource(c, "dialplan/agi", 0);
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_softhangup(c, AST_SOFTHANGUP_EXPLICIT);
c = ast_channel_unref(c);
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
/* if we get this far no channel name matched the argument given */
ast_agi_send(agi->fd, chan, "200 result=-1\n");
return RESULT_SUCCESS;
} else {
return RESULT_SHOWUSAGE;
}
}
static int handle_exec(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res, workaround;
struct ast_app *app_to_exec;
if (argc < 2)
return RESULT_SHOWUSAGE;
ast_verb(3, "AGI Script Executing Application: (%s) Options: (%s)\n", argv[1], argc >= 3 ? argv[2] : "");
if ((app_to_exec = pbx_findapp(argv[1]))) {
ast_channel_lock(chan);
if (!(workaround = ast_test_flag(ast_channel_flags(chan), AST_FLAG_DISABLE_WORKAROUNDS))) {
ast_set_flag(ast_channel_flags(chan), AST_FLAG_DISABLE_WORKAROUNDS);
}
ast_channel_unlock(chan);
Remove many deprecated modules Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-04 13:26:37 +00:00
res = pbx_exec(chan, app_to_exec, argc == 2 ? "" : argv[2]);
if (!workaround) {
ast_channel_clear_flag(chan, AST_FLAG_DISABLE_WORKAROUNDS);
}
} else {
ast_log(LOG_WARNING, "Could not find application (%s)\n", argv[1]);
res = -2;
}
ast_agi_send(agi->fd, chan, "200 result=%d\n", res);
/* Even though this is wrong, users are depending upon this result. */
return res;
}
static int handle_setcallerid(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
char tmp[256]="";
char *l = NULL, *n = NULL;
if (argv[2]) {
ast_copy_string(tmp, argv[2], sizeof(tmp));
ast_callerid_parse(tmp, &n, &l);
if (l)
ast_shrink_phone_number(l);
else
l = "";
if (!n)
n = "";
ast_set_callerid(chan, l, n, NULL);
}
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_channelstatus(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc == 2) {
/* no argument: supply info on the current channel */
ast_agi_send(agi->fd, chan, "200 result=%u\n", ast_channel_state(chan));
return RESULT_SUCCESS;
} else if (argc == 3) {
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
/* one argument: look for info on the specified channel */
if ((msg = stasis_cache_get(ast_channel_cache_by_name(), ast_channel_snapshot_type(), argv[2]))) {
struct ast_channel_snapshot *snapshot = stasis_message_data(msg);
ast_agi_send(agi->fd, chan, "200 result=%u\n", snapshot->state);
return RESULT_SUCCESS;
}
/* if we get this far no channel name matched the argument given */
ast_agi_send(agi->fd, chan, "200 result=-1\n");
return RESULT_SUCCESS;
} else {
return RESULT_SHOWUSAGE;
}
}
static int handle_setvariable(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 4) {
return RESULT_SHOWUSAGE;
}
if (argv[3])
pbx_builtin_setvar_helper(chan, argv[2], argv[3]);
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_getvariable(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
char *ret;
char tempstr[1024] = "";
if (argc != 3)
return RESULT_SHOWUSAGE;
/* check if we want to execute an ast_custom_function */
if (!ast_strlen_zero(argv[2]) && (argv[2][strlen(argv[2]) - 1] == ')')) {
ret = ast_func_read(chan, argv[2], tempstr, sizeof(tempstr)) ? NULL : tempstr;
} else {
pbx_retrieve_variable(chan, argv[2], &ret, tempstr, sizeof(tempstr), NULL);
}
if (ret)
ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", ret);
else
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_getvariablefull(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
struct ast_channel *chan2 = NULL;
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 (argc != 4 && argc != 5) {
return RESULT_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 (argc == 5) {
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
chan2 = ast_channel_get_by_name(argv[4]);
} else {
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
chan2 = ast_channel_ref(chan);
}
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 (chan2) {
struct ast_str *str = ast_str_create(16);
if (!str) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
ast_str_substitute_variables(&str, 0, chan2, argv[3]);
ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", ast_str_buffer(str));
ast_free(str);
} else {
ast_agi_send(agi->fd, chan, "200 result=0\n");
}
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 (chan2) {
chan2 = ast_channel_unref(chan2);
}
return RESULT_SUCCESS;
}
static int handle_verbose(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int level = 0;
if (argc < 2)
return RESULT_SHOWUSAGE;
if (argv[2])
sscanf(argv[2], "%30d", &level);
ast_verb(level, "%s: %s\n", ast_channel_data(chan), argv[1]);
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_dbget(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
struct ast_str *buf;
if (argc != 4)
return RESULT_SHOWUSAGE;
if (!(buf = ast_str_create(16))) {
ast_agi_send(agi->fd, chan, "200 result=-1\n");
return RESULT_SUCCESS;
}
do {
res = ast_db_get(argv[2], argv[3], ast_str_buffer(buf), ast_str_size(buf));
ast_str_update(buf);
if (ast_str_strlen(buf) < ast_str_size(buf) - 1) {
break;
}
if (ast_str_make_space(&buf, ast_str_size(buf) * 2)) {
break;
}
} while (1);
if (res)
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
ast_agi_send(agi->fd, chan, "200 result=1 (%s)\n", ast_str_buffer(buf));
ast_free(buf);
return RESULT_SUCCESS;
}
static int handle_dbput(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 5)
return RESULT_SHOWUSAGE;
res = ast_db_put(argv[2], argv[3], argv[4]);
ast_agi_send(agi->fd, chan, "200 result=%c\n", res ? '0' : '1');
return RESULT_SUCCESS;
}
static int handle_dbdel(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int res;
if (argc != 4)
return RESULT_SHOWUSAGE;
res = ast_db_del(argv[2], argv[3]);
ast_agi_send(agi->fd, chan, "200 result=%c\n", res ? '0' : '1');
return RESULT_SUCCESS;
}
static int handle_dbdeltree(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
int num_deleted;
if ((argc < 3) || (argc > 4)) {
return RESULT_SHOWUSAGE;
}
if (argc == 4) {
num_deleted = ast_db_deltree(argv[2], argv[3]);
} else {
num_deleted = ast_db_deltree(argv[2], NULL);
}
ast_agi_send(agi->fd, chan, "200 result=%c\n", num_deleted > 0 ? '0' : '1');
return RESULT_SUCCESS;
}
static char *handle_cli_agi_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "agi set debug [on|off]";
e->usage =
"Usage: agi set debug [on|off]\n"
" Enables/disables dumping of AGI transactions for\n"
" debugging purposes.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
return CLI_SHOWUSAGE;
if (strncasecmp(a->argv[3], "off", 3) == 0) {
agidebug = 0;
} else if (strncasecmp(a->argv[3], "on", 2) == 0) {
agidebug = 1;
} else {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "AGI Debugging %sabled\n", agidebug ? "En" : "Dis");
return CLI_SUCCESS;
}
static int handle_noop(struct ast_channel *chan, AGI *agi, int arg, const char * const argv[])
{
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_setmusic(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc < 3) {
return RESULT_SHOWUSAGE;
}
if (!strncasecmp(argv[2], "on", 2))
ast_moh_start(chan, argc > 3 ? argv[3] : NULL, NULL);
else if (!strncasecmp(argv[2], "off", 3))
ast_moh_stop(chan);
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
static int handle_speechcreate(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
struct ast_format_cap *cap;
/* If a structure already exists, return an error */
if (agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
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 (!(cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
return RESULT_FAILURE;
}
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
ast_format_cap_append(cap, ast_format_slin, 0);
if ((agi->speech = ast_speech_new(argv[2], cap))) {
ast_agi_send(agi->fd, chan, "200 result=1\n");
} else {
ast_agi_send(agi->fd, chan, "200 result=0\n");
}
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
ao2_ref(cap, -1);
return RESULT_SUCCESS;
}
static int handle_speechset(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
/* Check for minimum arguments */
if (argc != 4)
return RESULT_SHOWUSAGE;
/* Check to make sure speech structure exists */
if (!agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
ast_speech_change(agi->speech, argv[2], argv[3]);
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_speechdestroy(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (agi->speech) {
ast_speech_destroy(agi->speech);
agi->speech = NULL;
ast_agi_send(agi->fd, chan, "200 result=1\n");
} else {
ast_agi_send(agi->fd, chan, "200 result=0\n");
}
return RESULT_SUCCESS;
}
static int handle_speechloadgrammar(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 5)
return RESULT_SHOWUSAGE;
if (!agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
if (ast_speech_grammar_load(agi->speech, argv[3], argv[4]))
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_speechunloadgrammar(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 4)
return RESULT_SHOWUSAGE;
if (!agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
if (ast_speech_grammar_unload(agi->speech, argv[3]))
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_speechactivategrammar(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 4)
return RESULT_SHOWUSAGE;
if (!agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
if (ast_speech_grammar_activate(agi->speech, argv[3]))
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int handle_speechdeactivategrammar(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
if (argc != 4)
return RESULT_SHOWUSAGE;
if (!agi->speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
if (ast_speech_grammar_deactivate(agi->speech, argv[3]))
ast_agi_send(agi->fd, chan, "200 result=0\n");
else
ast_agi_send(agi->fd, chan, "200 result=1\n");
return RESULT_SUCCESS;
}
static int speech_streamfile(struct ast_channel *chan, const char *filename, const char *preflang, int offset)
{
struct ast_filestream *fs = NULL;
if (!(fs = ast_openstream(chan, filename, preflang)))
return -1;
if (offset)
ast_seekstream(fs, offset, SEEK_SET);
if (ast_applystream(chan, fs))
return -1;
if (ast_playstream(fs))
return -1;
return 0;
}
static int handle_speechrecognize(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[])
{
struct ast_speech *speech = agi->speech;
const char *prompt;
char dtmf = 0, tmp[4096] = "", *buf = tmp;
int timeout = 0, offset = 0, res = 0, i = 0;
long current_offset = 0;
const char *reason = NULL;
struct ast_frame *fr = NULL;
struct ast_speech_result *result = NULL;
size_t left = sizeof(tmp);
time_t start = 0, current;
if (argc < 4)
return RESULT_SHOWUSAGE;
if (!speech) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
prompt = argv[2];
timeout = atoi(argv[3]);
/* If offset is specified then convert from text to integer */
if (argc == 5)
offset = atoi(argv[4]);
/* We want frames coming in signed linear */
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 (ast_set_read_format(chan, ast_format_slin)) {
ast_agi_send(agi->fd, chan, "200 result=0\n");
return RESULT_SUCCESS;
}
/* Setup speech structure */
if (speech->state == AST_SPEECH_STATE_NOT_READY || speech->state == AST_SPEECH_STATE_DONE) {
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
ast_speech_start(speech);
}
/* Start playing prompt */
speech_streamfile(chan, prompt, ast_channel_language(chan), offset);
/* Go into loop reading in frames, passing to speech thingy, checking for hangup, all that jazz */
while (ast_strlen_zero(reason)) {
/* Run scheduled items */
ast_sched_runq(ast_channel_sched(chan));
/* See maximum time of waiting */
if ((res = ast_sched_wait(ast_channel_sched(chan))) < 0)
res = 1000;
/* Wait for frame */
if (ast_waitfor(chan, res) > 0) {
if (!(fr = ast_read(chan))) {
reason = "hangup";
break;
}
}
/* Perform timeout check */
if ((timeout > 0) && (start > 0)) {
time(&current);
if ((current - start) >= timeout) {
reason = "timeout";
if (fr)
ast_frfree(fr);
break;
}
}
/* Check the speech structure for any changes */
ast_mutex_lock(&speech->lock);
/* See if we need to quiet the audio stream playback */
if (ast_test_flag(speech, AST_SPEECH_QUIET) && ast_channel_stream(chan)) {
current_offset = ast_tellstream(ast_channel_stream(chan));
ast_stopstream(chan);
ast_clear_flag(speech, AST_SPEECH_QUIET);
}
/* Check each state */
switch (speech->state) {
case AST_SPEECH_STATE_READY:
/* If the stream is done, start timeout calculation */
if ((timeout > 0) && start == 0 && ((!ast_channel_stream(chan)) || (ast_channel_streamid(chan) == -1 && ast_channel_timingfunc(chan) == NULL))) {
ast_stopstream(chan);
time(&start);
}
/* Write audio frame data into speech engine if possible */
if (fr && fr->frametype == AST_FRAME_VOICE)
ast_speech_write(speech, fr->data.ptr, fr->datalen);
break;
case AST_SPEECH_STATE_WAIT:
/* Cue waiting sound if not already playing */
if ((!ast_channel_stream(chan)) || (ast_channel_streamid(chan) == -1 && ast_channel_timingfunc(chan) == NULL)) {
ast_stopstream(chan);
/* If a processing sound exists, or is not none - play it */
if (!ast_strlen_zero(speech->processing_sound) && strcasecmp(speech->processing_sound, "none"))
speech_streamfile(chan, speech->processing_sound, ast_channel_language(chan), 0);
}
break;
case AST_SPEECH_STATE_DONE:
/* Get the results */
speech->results = ast_speech_results_get(speech);
/* Change state to not ready */
ast_speech_change_state(speech, AST_SPEECH_STATE_NOT_READY);
reason = "speech";
break;
default:
break;
}
ast_mutex_unlock(&speech->lock);
/* Check frame for DTMF or hangup */
if (fr) {
if (fr->frametype == AST_FRAME_DTMF) {
reason = "dtmf";
dtmf = fr->subclass.integer;
} else if (fr->frametype == AST_FRAME_CONTROL && fr->subclass.integer == AST_CONTROL_HANGUP) {
reason = "hangup";
}
ast_frfree(fr);
}
}
if (!strcasecmp(reason, "speech")) {
/* Build string containing speech results */
for (result = speech->results; result; result = AST_LIST_NEXT(result, list)) {
/* Build result string */
ast_build_string(&buf, &left, "%sscore%d=%d text%d=\"%s\" grammar%d=%s", (i > 0 ? " " : ""), i, result->score, i, result->text, i, result->grammar);
/* Increment result count */
i++;
}
/* Print out */
ast_agi_send(agi->fd, chan, "200 result=1 (speech) endpos=%ld results=%d %s\n", current_offset, i, tmp);
} else if (!strcasecmp(reason, "dtmf")) {
ast_agi_send(agi->fd, chan, "200 result=1 (digit) digit=%c endpos=%ld\n", dtmf, current_offset);
} else if (!strcasecmp(reason, "hangup") || !strcasecmp(reason, "timeout")) {
ast_agi_send(agi->fd, chan, "200 result=1 (%s) endpos=%ld\n", reason, current_offset);
} else {
ast_agi_send(agi->fd, chan, "200 result=0 endpos=%ld\n", current_offset);
}
return RESULT_SUCCESS;
}
/*!
* \brief AGI commands list
*/
static struct agi_command commands[] = {
{ { "answer", NULL }, handle_answer, NULL, NULL, 0 },
{ { "asyncagi", "break", NULL }, handle_asyncagi_break, NULL, NULL, 1 },
{ { "channel", "status", NULL }, handle_channelstatus, NULL, NULL, 0 },
{ { "database", "del", NULL }, handle_dbdel, NULL, NULL, 1 },
{ { "database", "deltree", NULL }, handle_dbdeltree, NULL, NULL, 1 },
{ { "database", "get", NULL }, handle_dbget, NULL, NULL, 1 },
{ { "database", "put", NULL }, handle_dbput, NULL, NULL, 1 },
{ { "exec", NULL }, handle_exec, NULL, NULL, 1 },
{ { "get", "data", NULL }, handle_getdata, NULL, NULL, 0 },
{ { "get", "full", "variable", NULL }, handle_getvariablefull, NULL, NULL, 1 },
{ { "get", "option", NULL }, handle_getoption, NULL, NULL, 0 },
{ { "get", "variable", NULL }, handle_getvariable, NULL, NULL, 1 },
{ { "hangup", NULL }, handle_hangup, NULL, NULL, 0 },
{ { "noop", NULL }, handle_noop, NULL, NULL, 1 },
{ { "receive", "char", NULL }, handle_recvchar, NULL, NULL, 0 },
{ { "receive", "text", NULL }, handle_recvtext, NULL, NULL, 0 },
{ { "record", "file", NULL }, handle_recordfile, NULL, NULL, 0 },
{ { "say", "alpha", NULL }, handle_sayalpha, NULL, NULL, 0},
{ { "say", "digits", NULL }, handle_saydigits, NULL, NULL, 0 },
{ { "say", "number", NULL }, handle_saynumber, NULL, NULL, 0 },
{ { "say", "phonetic", NULL }, handle_sayphonetic, NULL, NULL, 0},
{ { "say", "date", NULL }, handle_saydate, NULL, NULL, 0},
{ { "say", "time", NULL }, handle_saytime, NULL, NULL, 0},
{ { "say", "datetime", NULL }, handle_saydatetime, NULL, NULL, 0},
{ { "send", "image", NULL }, handle_sendimage, NULL, NULL, 0},
{ { "send", "text", NULL }, handle_sendtext, NULL, NULL, 0},
{ { "set", "autohangup", NULL }, handle_autohangup, NULL, NULL, 0},
{ { "set", "callerid", NULL }, handle_setcallerid, NULL, NULL, 0},
{ { "set", "context", NULL }, handle_setcontext, NULL, NULL, 0},
{ { "set", "extension", NULL }, handle_setextension, NULL, NULL, 0},
{ { "set", "music", NULL }, handle_setmusic, NULL, NULL, 0 },
{ { "set", "priority", NULL }, handle_setpriority, NULL, NULL, 0 },
{ { "set", "variable", NULL }, handle_setvariable, NULL, NULL, 1 },
{ { "stream", "file", NULL }, handle_streamfile, NULL, NULL, 0 },
{ { "control", "stream", "file", NULL }, handle_controlstreamfile, NULL, NULL, 0 },
{ { "tdd", "mode", NULL }, handle_tddmode, NULL, NULL, 0 },
{ { "verbose", NULL }, handle_verbose, NULL, NULL, 1 },
{ { "wait", "for", "digit", NULL }, handle_waitfordigit, NULL, NULL, 0 },
{ { "speech", "create", NULL }, handle_speechcreate, NULL, NULL, 0 },
{ { "speech", "set", NULL }, handle_speechset, NULL, NULL, 0 },
{ { "speech", "destroy", NULL }, handle_speechdestroy, NULL, NULL, 1 },
{ { "speech", "load", "grammar", NULL }, handle_speechloadgrammar, NULL, NULL, 0 },
{ { "speech", "unload", "grammar", NULL }, handle_speechunloadgrammar, NULL, NULL, 1 },
{ { "speech", "activate", "grammar", NULL }, handle_speechactivategrammar, NULL, NULL, 0 },
{ { "speech", "deactivate", "grammar", NULL }, handle_speechdeactivategrammar, NULL, NULL, 0 },
{ { "speech", "recognize", NULL }, handle_speechrecognize, NULL, NULL, 0 },
};
static AST_RWLIST_HEAD_STATIC(agi_commands, agi_command);
static char *help_workhorse(int fd, const char * const match[])
{
char fullcmd[MAX_CMD_LEN], matchstr[MAX_CMD_LEN];
struct agi_command *e;
if (match)
ast_join(matchstr, sizeof(matchstr), match);
ast_cli(fd, "%5.5s %30.30s %s\n","Dead","Command","Description");
AST_RWLIST_RDLOCK(&agi_commands);
AST_RWLIST_TRAVERSE(&agi_commands, e, list) {
if (!e->cmda[0])
break;
/* Hide commands that start with '_' */
if ((e->cmda[0])[0] == '_')
continue;
ast_join(fullcmd, sizeof(fullcmd), e->cmda);
if (match && strncasecmp(matchstr, fullcmd, strlen(matchstr)))
continue;
ast_cli(fd, "%5.5s %30.30s %s\n", e->dead ? "Yes" : "No" , fullcmd, S_OR(e->summary, "Not available"));
}
AST_RWLIST_UNLOCK(&agi_commands);
return CLI_SUCCESS;
}
int AST_OPTIONAL_API_NAME(ast_agi_register)(struct ast_module *mod, agi_command *cmd)
{
char fullcmd[MAX_CMD_LEN];
ast_join(fullcmd, sizeof(fullcmd), cmd->cmda);
if (!find_command(cmd->cmda, 1)) {
*((enum ast_doc_src *) &cmd->docsrc) = AST_STATIC_DOC;
if (ast_strlen_zero(cmd->summary) && ast_strlen_zero(cmd->usage)) {
#ifdef AST_XML_DOCS
*((char **) &cmd->summary) = ast_xmldoc_build_synopsis("agi", fullcmd, NULL);
*((char **) &cmd->usage) = ast_xmldoc_build_description("agi", fullcmd, NULL);
*((char **) &cmd->syntax) = ast_xmldoc_build_syntax("agi", fullcmd, NULL);
*((char **) &cmd->seealso) = ast_xmldoc_build_seealso("agi", fullcmd, NULL);
*((enum ast_doc_src *) &cmd->docsrc) = AST_XML_DOC;
#endif
#ifndef HAVE_NULLSAFE_PRINTF
if (!cmd->summary) {
*((char **) &cmd->summary) = ast_strdup("");
}
if (!cmd->usage) {
*((char **) &cmd->usage) = ast_strdup("");
}
if (!cmd->syntax) {
*((char **) &cmd->syntax) = ast_strdup("");
}
if (!cmd->seealso) {
*((char **) &cmd->seealso) = ast_strdup("");
}
#endif
}
cmd->mod = mod;
AST_RWLIST_WRLOCK(&agi_commands);
AST_LIST_INSERT_TAIL(&agi_commands, cmd, list);
AST_RWLIST_UNLOCK(&agi_commands);
if (mod != ast_module_info->self)
ast_module_ref(ast_module_info->self);
ast_verb(2, "AGI Command '%s' registered\n",fullcmd);
return 1;
} else {
ast_log(LOG_WARNING, "Command already registered!\n");
return 0;
}
}
int AST_OPTIONAL_API_NAME(ast_agi_unregister)(struct ast_module *mod, agi_command *cmd)
{
struct agi_command *e;
int unregistered = 0;
char fullcmd[MAX_CMD_LEN];
ast_join(fullcmd, sizeof(fullcmd), cmd->cmda);
AST_RWLIST_WRLOCK(&agi_commands);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&agi_commands, e, list) {
if (cmd == e) {
AST_RWLIST_REMOVE_CURRENT(list);
if (mod != ast_module_info->self)
ast_module_unref(ast_module_info->self);
#ifdef AST_XML_DOCS
if (e->docsrc == AST_XML_DOC) {
ast_free((char *) e->summary);
ast_free((char *) e->usage);
ast_free((char *) e->syntax);
ast_free((char *) e->seealso);
*((char **) &e->summary) = NULL;
*((char **) &e->usage) = NULL;
*((char **) &e->syntax) = NULL;
*((char **) &e->seealso) = NULL;
}
#endif
unregistered=1;
break;
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&agi_commands);
if (unregistered) {
ast_verb(2, "AGI Command '%s' unregistered\n",fullcmd);
}
return unregistered;
}
int AST_OPTIONAL_API_NAME(ast_agi_register_multiple)(struct ast_module *mod, struct agi_command *cmd, unsigned int len)
{
unsigned int i, x = 0;
for (i = 0; i < len; i++) {
if (ast_agi_register(mod, cmd + i) == 1) {
x++;
continue;
}
/* registration failed, unregister everything
that had been registered up to that point
*/
for (; x > 0; x--) {
/* we are intentionally ignoring the
result of ast_agi_unregister() here,
but it should be safe to do so since
we just registered these commands and
the only possible way for unregistration
to fail is if the command is not
registered
*/
(void) ast_agi_unregister(mod, cmd + x - 1);
}
return -1;
}
return 0;
}
int AST_OPTIONAL_API_NAME(ast_agi_unregister_multiple)(struct ast_module *mod, struct agi_command *cmd, unsigned int len)
{
unsigned int i;
int res = 0;
for (i = 0; i < len; i++) {
/* remember whether any of the unregistration
attempts failed... there is no recourse if
any of them do
*/
res |= ast_agi_unregister(mod, cmd + i);
}
return res;
}
static agi_command *find_command(const char * const cmds[], int exact)
{
int y, match;
struct agi_command *e;
AST_RWLIST_RDLOCK(&agi_commands);
AST_RWLIST_TRAVERSE(&agi_commands, e, list) {
if (!e->cmda[0])
break;
/* start optimistic */
match = 1;
for (y = 0; match && cmds[y]; y++) {
/* If there are no more words in the command (and we're looking for
an exact match) or there is a difference between the two words,
then this is not a match */
if (!e->cmda[y] && !exact)
break;
/* don't segfault if the next part of a command doesn't exist */
if (!e->cmda[y]) {
AST_RWLIST_UNLOCK(&agi_commands);
return NULL;
}
if (strcasecmp(e->cmda[y], cmds[y]))
match = 0;
}
/* If more words are needed to complete the command then this is not
a candidate (unless we're looking for a really inexact answer */
if ((exact > -1) && e->cmda[y])
match = 0;
if (match) {
AST_RWLIST_UNLOCK(&agi_commands);
return e;
}
}
AST_RWLIST_UNLOCK(&agi_commands);
return NULL;
}
static int parse_args(char *s, int *max, const char *argv[])
{
int x = 0, quoted = 0, escaped = 0, whitespace = 1;
char *cur;
cur = s;
while(*s) {
switch(*s) {
case '"':
/* If it's escaped, put a literal quote */
if (escaped)
goto normal;
else
quoted = !quoted;
if (quoted && whitespace) {
/* If we're starting a quote, coming off white space start a new word, too */
argv[x++] = cur;
whitespace=0;
}
escaped = 0;
break;
case ' ':
case '\t':
if (!quoted && !escaped) {
/* If we're not quoted, mark this as whitespace, and
end the previous argument */
whitespace = 1;
*(cur++) = '\0';
} else
/* Otherwise, just treat it as anything else */
goto normal;
break;
case '\\':
/* If we're escaped, print a literal, otherwise enable escaping */
if (escaped) {
goto normal;
} else {
escaped=1;
}
break;
default:
normal:
if (whitespace) {
if (x >= MAX_ARGS -1) {
ast_log(LOG_WARNING, "Too many arguments, truncating\n");
break;
}
/* Coming off of whitespace, start the next argument */
argv[x++] = cur;
whitespace=0;
}
*(cur++) = *s;
escaped=0;
}
s++;
}
/* Null terminate */
*(cur++) = '\0';
argv[x] = NULL;
*max = x;
return 0;
}
static void publish_async_exec_end(struct ast_channel *chan, int command_id, const char *command, int result_code, const char *result)
{
RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
blob = ast_json_pack("{s: i, s: s, s: i, s: s}",
"CommandId", command_id,
"Command", command,
"ResultCode", result_code,
"Result", result);
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
ast_channel_publish_cached_blob(chan, agi_exec_end_type(), blob);
}
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
static enum agi_result agi_handle_command(struct ast_channel *chan, AGI *agi, char *buf, int dead)
{
const char *argv[MAX_ARGS] = {0};
int argc = MAX_ARGS;
int res;
agi_command *c;
char *ami_cmd = ast_strdupa(buf);
const char *ami_res;
int command_id = ast_random();
int resultcode = 0;
RAII_VAR(struct ast_json *, startblob, NULL, ast_json_unref);
startblob = ast_json_pack("{s: i, s: s}",
"CommandId", command_id,
"Command", ami_cmd);
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
ast_channel_publish_cached_blob(chan, agi_exec_start_type(), startblob);
parse_args(buf, &argc, argv);
c = find_command(argv, 0);
if (!c || !ast_module_running_ref(c->mod)) {
ami_res = "Invalid or unknown command";
resultcode = 510;
ast_agi_send(agi->fd, chan, "%d %s\n", resultcode, ami_res);
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
return AGI_RESULT_SUCCESS;
}
if (!dead || (dead && c->dead)) {
res = c->handler(chan, agi, argc, argv);
switch (res) {
case RESULT_SHOWUSAGE:
ami_res = "Usage";
resultcode = 520;
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
if (ast_strlen_zero(c->usage)) {
ast_agi_send(agi->fd, chan, "520 Invalid command syntax. Proper usage not available.\n");
} else {
ast_agi_send(agi->fd, chan, "520-Invalid command syntax. Proper usage follows:\n");
ast_agi_send(agi->fd, chan, "%s\n", c->usage);
ast_agi_send(agi->fd, chan, "520 End of proper usage.\n");
}
break;
case RESULT_FAILURE:
ami_res = "Failure";
resultcode = -1;
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
/* The RESULT_FAILURE code is usually because the channel hungup. */
return AGI_RESULT_FAILURE;
case ASYNC_AGI_BREAK:
ami_res = "Success";
resultcode = 200;
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
return AGI_RESULT_SUCCESS_ASYNC;
case RESULT_SUCCESS:
ami_res = "Success";
resultcode = 200;
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
break;
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
default:
ami_res = "Unknown Result";
resultcode = 200;
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
break;
}
} else {
ami_res = "Command Not Permitted on a dead channel or intercept routine";
resultcode = 511;
ast_agi_send(agi->fd, chan, "%d %s\n", resultcode, ami_res);
publish_async_exec_end(chan, command_id, ami_cmd, resultcode, ami_res);
}
ast_module_unref(c->mod);
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
return AGI_RESULT_SUCCESS;
}
static enum agi_result run_agi(struct ast_channel *chan, char *request, AGI *agi, int pid, int *status, int dead, int argc, char *argv[])
{
struct ast_channel *c;
int outfd;
int ms;
int needhup = 0;
enum agi_result returnstatus = AGI_RESULT_SUCCESS;
struct ast_frame *f;
char buf[AGI_BUF_LEN];
char *res = NULL;
FILE *readf;
/* how many times we'll retry if ast_waitfor_nandfs will return without either
channel or file descriptor in case select is interrupted by a system call (EINTR) */
int retry = AGI_NANDFS_RETRY;
int send_sighup;
const char *sighup_str;
const char *exit_on_hangup_str;
int exit_on_hangup;
/*! Running in an interception routine is like DeadAGI mode. No touchy the channel frames. */
int in_intercept = ast_channel_get_intercept_mode();
ast_channel_lock(chan);
sighup_str = pbx_builtin_getvar_helper(chan, "AGISIGHUP");
send_sighup = !ast_false(sighup_str);
exit_on_hangup_str = pbx_builtin_getvar_helper(chan, "AGIEXITONHANGUP");
exit_on_hangup = ast_true(exit_on_hangup_str);
ast_channel_unlock(chan);
if (!(readf = fdopen(agi->ctrl, "r"))) {
ast_log(LOG_WARNING, "Unable to fdopen file descriptor\n");
if (send_sighup && pid > -1)
kill(pid, SIGHUP);
close(agi->ctrl);
return AGI_RESULT_FAILURE;
}
setlinebuf(readf);
setup_env(chan, request, agi->fd, (agi->audio > -1), argc, argv);
for (;;) {
if (needhup) {
needhup = 0;
dead = 1;
if (send_sighup) {
if (pid > -1) {
kill(pid, SIGHUP);
} else if (agi->fast) {
ast_agi_send(agi->fd, chan, "HANGUP\n");
}
}
if (exit_on_hangup) {
break;
}
}
ms = -1;
if (dead || in_intercept) {
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
c = ast_waitfor_nandfds(&chan, 0, &agi->ctrl, 1, NULL, &outfd, &ms);
} else if (!ast_check_hangup(chan)) {
c = ast_waitfor_nandfds(&chan, 1, &agi->ctrl, 1, NULL, &outfd, &ms);
} else {
/*
* Read the channel control queue until it is dry so we can
* switch to dead mode.
*/
c = chan;
}
if (c) {
retry = AGI_NANDFS_RETRY;
/* Idle the channel until we get a command */
f = ast_read(c);
if (!f) {
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
ast_debug(1, "%s hungup\n", ast_channel_name(chan));
needhup = 1;
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
if (!returnstatus) {
returnstatus = AGI_RESULT_HANGUP;
}
} else {
/* If it's voice, write it to the audio pipe */
if ((agi->audio > -1) && (f->frametype == AST_FRAME_VOICE)) {
/* Write, ignoring errors */
if (write(agi->audio, f->data.ptr, f->datalen) < 0) {
}
}
ast_frfree(f);
}
} else if (outfd > -1) {
size_t len = sizeof(buf);
size_t buflen = 0;
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
enum agi_result cmd_status;
retry = AGI_NANDFS_RETRY;
buf[0] = '\0';
while (len > 1) {
res = fgets(buf + buflen, len, readf);
if (feof(readf))
break;
if (ferror(readf) && ((errno != EINTR) && (errno != EAGAIN)))
break;
if (res != NULL && !agi->fast)
break;
buflen = strlen(buf);
if (buflen && buf[buflen - 1] == '\n')
break;
len = sizeof(buf) - buflen;
if (agidebug)
ast_verbose("AGI Rx << temp buffer %s - errno %s\nNo \\n received, checking again.\n", buf, strerror(errno));
}
if (!buf[0]) {
/* Program terminated */
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
ast_verb(3, "<%s>AGI Script %s completed, returning %d\n", ast_channel_name(chan), request, returnstatus);
if (pid > 0)
waitpid(pid, status, 0);
/* No need to kill the pid anymore, since they closed us */
pid = -1;
break;
}
/* Special case for inability to execute child process */
if (*buf && strncasecmp(buf, "failure", 7) == 0) {
returnstatus = AGI_RESULT_FAILURE;
break;
}
/* get rid of trailing newline, if any */
buflen = strlen(buf);
if (buflen && buf[buflen - 1] == '\n') {
buf[buflen - 1] = '\0';
}
if (agidebug)
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
ast_verbose("<%s>AGI Rx << %s\n", ast_channel_name(chan), buf);
cmd_status = agi_handle_command(chan, agi, buf, dead || in_intercept);
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
switch (cmd_status) {
case AGI_RESULT_FAILURE:
if (dead || in_intercept || !ast_check_hangup(chan)) {
Merged revisions 313588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines Merged revisions 313579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines Merged revisions 313545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines Asterisk does not hangup a channel after endpoint hangs up. If the call that the dialplan started an AGI script for is hungup while the AGI script is in the middle of a command then the AGI script is not notified of the hangup. There are many AGI Exec commands that this can happen with. The reported applications have been: Background, Wait, Read, and Dial. Also the AGI Get Data command. * Don't wait on the Asterisk channel after it has hung up. The channel is likely to never need servicing again. * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value in run_agi(). It previously only could return AGI_RESULT_SUCCESS or AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged. (closes issue #17954) Reported by: mn3250 Patches: issue17954_v1.8.patch uploaded by rmudgett (license 664) issue17954_v1.6.2.patch uploaded by rmudgett (license 664) issue17954_v1.4.patch uploaded by rmudgett (license 664) Tested by: rmudgett JIRA SWP-2171 (closes issue #18492) Reported by: devmod Tested by: rmudgett JIRA SWP-2761 (closes issue #18935) Reported by: nvitaly Tested by: astmiv, rmudgett JIRA SWP-3216 (closes issue #17393) Reported by: siby Tested by: rmudgett JIRA SWP-2727 Review: https://reviewboard.asterisk.org/r/1165/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-04-13 16:37:06 +00:00
/* The failure was not because of a hangup. */
returnstatus = AGI_RESULT_FAILURE;
}
break;
default:
break;
}
} else {
if (--retry <= 0) {
ast_log(LOG_WARNING, "No channel, no fd?\n");
returnstatus = AGI_RESULT_FAILURE;
break;
}
}
}
if (agi->speech) {
ast_speech_destroy(agi->speech);
}
/* Notify process */
if (send_sighup) {
if (pid > -1) {
if (kill(pid, SIGHUP)) {
ast_log(LOG_WARNING, "unable to send SIGHUP to AGI process %d: %s\n", pid, strerror(errno));
} else { /* Give the process a chance to die */
usleep(1);
}
waitpid(pid, status, WNOHANG);
} else if (agi->fast) {
ast_agi_send(agi->fd, chan, "HANGUP\n");
}
}
fclose(readf);
return returnstatus;
}
static char *handle_cli_agi_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct agi_command *command;
char fullcmd[MAX_CMD_LEN];
int error = 0;
switch (cmd) {
case CLI_INIT:
e->command = "agi show commands [topic]";
e->usage =
"Usage: agi show commands [topic] <topic>\n"
" When called with a topic as an argument, displays usage\n"
" information on the given command. If called without a\n"
" topic, it provides a list of AGI commands.\n";
case CLI_GENERATE:
return NULL;
}
if (a->argc < e->args - 1 || (a->argc >= e->args && strcasecmp(a->argv[e->args - 1], "topic")))
return CLI_SHOWUSAGE;
if (a->argc > e->args - 1) {
command = find_command(a->argv + e->args, 1);
if (command) {
char *synopsis = NULL, *description = NULL, *syntax = NULL, *seealso = NULL;
char info[30 + MAX_CMD_LEN]; /* '-= Info about...' */
char infotitle[30 + MAX_CMD_LEN + AST_TERM_MAX_ESCAPE_CHARS]; /* '-= Info about...' with colors */
char syntitle[11 + AST_TERM_MAX_ESCAPE_CHARS]; /* [Syntax]\n with colors */
char desctitle[15 + AST_TERM_MAX_ESCAPE_CHARS]; /* [Description]\n with colors */
char deadtitle[13 + AST_TERM_MAX_ESCAPE_CHARS]; /* [Runs Dead]\n with colors */
char deadcontent[3 + AST_TERM_MAX_ESCAPE_CHARS]; /* 'Yes' or 'No' with colors */
char seealsotitle[12 + AST_TERM_MAX_ESCAPE_CHARS]; /* [See Also]\n with colors */
char stxtitle[10 + AST_TERM_MAX_ESCAPE_CHARS]; /* [Syntax]\n with colors */
size_t synlen, desclen, seealsolen, stxlen;
term_color(syntitle, "[Synopsis]\n", COLOR_MAGENTA, 0, sizeof(syntitle));
term_color(desctitle, "[Description]\n", COLOR_MAGENTA, 0, sizeof(desctitle));
term_color(deadtitle, "[Runs Dead]\n", COLOR_MAGENTA, 0, sizeof(deadtitle));
term_color(seealsotitle, "[See Also]\n", COLOR_MAGENTA, 0, sizeof(seealsotitle));
term_color(stxtitle, "[Syntax]\n", COLOR_MAGENTA, 0, sizeof(stxtitle));
term_color(deadcontent, command->dead ? "Yes" : "No", COLOR_CYAN, 0, sizeof(deadcontent));
ast_join(fullcmd, sizeof(fullcmd), a->argv + e->args);
snprintf(info, sizeof(info), "\n -= Info about agi '%s' =- ", fullcmd);
term_color(infotitle, info, COLOR_CYAN, 0, sizeof(infotitle));
#ifdef AST_XML_DOCS
if (command->docsrc == AST_XML_DOC) {
synopsis = ast_xmldoc_printable(S_OR(command->summary, "Not available"), 1);
description = ast_xmldoc_printable(S_OR(command->usage, "Not available"), 1);
seealso = ast_xmldoc_printable(S_OR(command->seealso, "Not available"), 1);
if (!seealso || !description || !synopsis) {
error = 1;
goto return_cleanup;
}
} else
#endif
{
synlen = strlen(S_OR(command->summary, "Not available")) + AST_TERM_MAX_ESCAPE_CHARS;
synopsis = ast_malloc(synlen);
desclen = strlen(S_OR(command->usage, "Not available")) + AST_TERM_MAX_ESCAPE_CHARS;
description = ast_malloc(desclen);
seealsolen = strlen(S_OR(command->seealso, "Not available")) + AST_TERM_MAX_ESCAPE_CHARS;
seealso = ast_malloc(seealsolen);
if (!synopsis || !description || !seealso) {
error = 1;
goto return_cleanup;
}
term_color(synopsis, S_OR(command->summary, "Not available"), COLOR_CYAN, 0, synlen);
term_color(description, S_OR(command->usage, "Not available"), COLOR_CYAN, 0, desclen);
term_color(seealso, S_OR(command->seealso, "Not available"), COLOR_CYAN, 0, seealsolen);
}
stxlen = strlen(S_OR(command->syntax, "Not available")) + AST_TERM_MAX_ESCAPE_CHARS;
syntax = ast_malloc(stxlen);
if (!syntax) {
error = 1;
goto return_cleanup;
}
term_color(syntax, S_OR(command->syntax, "Not available"), COLOR_CYAN, 0, stxlen);
ast_cli(a->fd, "%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n", infotitle, stxtitle, syntax,
desctitle, description, syntitle, synopsis, deadtitle, deadcontent,
seealsotitle, seealso);
return_cleanup:
ast_free(synopsis);
ast_free(description);
ast_free(syntax);
ast_free(seealso);
} else {
if (find_command(a->argv + e->args, -1)) {
return help_workhorse(a->fd, a->argv + e->args);
} else {
ast_join(fullcmd, sizeof(fullcmd), a->argv + e->args);
ast_cli(a->fd, "No such command '%s'.\n", fullcmd);
}
}
} else {
return help_workhorse(a->fd, NULL);
}
return (error ? CLI_FAILURE : CLI_SUCCESS);
}
/*! \brief Convert string to use HTML escaped characters
\note Maybe this should be a generic function?
*/
static void write_html_escaped(FILE *htmlfile, char *str)
{
char *cur = str;
while(*cur) {
switch (*cur) {
case '<':
fprintf(htmlfile, "%s", "&lt;");
break;
case '>':
fprintf(htmlfile, "%s", "&gt;");
break;
case '&':
fprintf(htmlfile, "%s", "&amp;");
break;
case '"':
fprintf(htmlfile, "%s", "&quot;");
break;
default:
fprintf(htmlfile, "%c", *cur);
break;
}
cur++;
}
return;
}
static int write_htmldump(const char *filename)
{
struct agi_command *command;
char fullcmd[MAX_CMD_LEN];
FILE *htmlfile;
if (!(htmlfile = fopen(filename, "wt")))
return -1;
fprintf(htmlfile, "<HTML>\n<HEAD>\n<TITLE>AGI Commands</TITLE>\n</HEAD>\n");
fprintf(htmlfile, "<BODY>\n<CENTER><B><H1>AGI Commands</H1></B></CENTER>\n\n");
fprintf(htmlfile, "<TABLE BORDER=\"0\" CELLSPACING=\"10\">\n");
AST_RWLIST_RDLOCK(&agi_commands);
AST_RWLIST_TRAVERSE(&agi_commands, command, list) {
char *tempstr, *stringp;
if (!command->cmda[0]) /* end ? */
break;
/* Hide commands that start with '_' */
if ((command->cmda[0])[0] == '_')
continue;
ast_join(fullcmd, sizeof(fullcmd), command->cmda);
fprintf(htmlfile, "<TR><TD><TABLE BORDER=\"1\" CELLPADDING=\"5\" WIDTH=\"100%%\">\n");
fprintf(htmlfile, "<TR><TH ALIGN=\"CENTER\"><B>%s - %s</B></TH></TR>\n", fullcmd, command->summary);
#ifdef AST_XML_DOCS
stringp = ast_xmldoc_printable(command->usage, 0);
#else
stringp = ast_strdup(command->usage);
#endif
tempstr = strsep(&stringp, "\n");
fprintf(htmlfile, "<TR><TD ALIGN=\"CENTER\">");
write_html_escaped(htmlfile, tempstr);
fprintf(htmlfile, "</TD></TR>\n");
fprintf(htmlfile, "<TR><TD ALIGN=\"CENTER\">\n");
while ((tempstr = strsep(&stringp, "\n")) != NULL) {
write_html_escaped(htmlfile, tempstr);
fprintf(htmlfile, "<BR>\n");
}
fprintf(htmlfile, "</TD></TR>\n");
fprintf(htmlfile, "</TABLE></TD></TR>\n\n");
ast_free(stringp);
}
AST_RWLIST_UNLOCK(&agi_commands);
fprintf(htmlfile, "</TABLE>\n</BODY>\n</HTML>\n");
fclose(htmlfile);
return 0;
}
static char *handle_cli_agi_dump_html(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "agi dump html";
e->usage =
"Usage: agi dump html <filename>\n"
" Dumps the AGI command list in HTML format to the given\n"
" file.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args + 1)
return CLI_SHOWUSAGE;
if (write_htmldump(a->argv[e->args]) < 0) {
ast_cli(a->fd, "Could not create file '%s'\n", a->argv[e->args]);
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "AGI HTML commands dumped to: %s\n", a->argv[e->args]);
return CLI_SUCCESS;
}
static int agi_exec_full(struct ast_channel *chan, const char *data, int enhanced, int dead)
{
enum agi_result res;
char *buf;
int fds[2], efd = -1, pid = -1;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(arg)[MAX_ARGS];
);
AGI agi;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "AGI requires an argument (script)\n");
return -1;
}
if (dead)
ast_debug(3, "Hungup channel detected, running agi in dead mode.\n");
memset(&agi, 0, sizeof(agi));
buf = ast_strdupa(data);
AST_STANDARD_APP_ARGS(args, buf);
args.argv[args.argc] = NULL;
#if 0
/* Answer if need be */
if (chan->_state != AST_STATE_UP) {
if (ast_answer(chan))
return -1;
}
#endif
res = launch_script(chan, args.argv[0], args.argc, args.argv, fds, enhanced ? &efd : NULL, &pid);
/* Async AGI do not require run_agi(), so just proceed if normal AGI
or Fast AGI are setup with success. */
if (res == AGI_RESULT_SUCCESS || res == AGI_RESULT_SUCCESS_FAST) {
int status = 0;
agi.fd = fds[1];
agi.ctrl = fds[0];
agi.audio = efd;
agi.fast = (res == AGI_RESULT_SUCCESS_FAST) ? 1 : 0;
res = run_agi(chan, args.argv[0], &agi, pid, &status, dead, args.argc, args.argv);
/* If the fork'd process returns non-zero, set AGISTATUS to FAILURE */
if ((res == AGI_RESULT_SUCCESS || res == AGI_RESULT_SUCCESS_FAST) && status)
res = AGI_RESULT_FAILURE;
if (fds[1] != fds[0])
close(fds[1]);
if (efd > -1)
close(efd);
}
ast_safe_fork_cleanup();
switch (res) {
case AGI_RESULT_SUCCESS:
case AGI_RESULT_SUCCESS_FAST:
case AGI_RESULT_SUCCESS_ASYNC:
pbx_builtin_setvar_helper(chan, "AGISTATUS", "SUCCESS");
break;
case AGI_RESULT_FAILURE:
pbx_builtin_setvar_helper(chan, "AGISTATUS", "FAILURE");
break;
case AGI_RESULT_NOTFOUND:
pbx_builtin_setvar_helper(chan, "AGISTATUS", "NOTFOUND");
break;
case AGI_RESULT_HANGUP:
pbx_builtin_setvar_helper(chan, "AGISTATUS", "HANGUP");
return -1;
}
return 0;
}
static int agi_exec(struct ast_channel *chan, const char *data)
{
if (!ast_check_hangup(chan))
return agi_exec_full(chan, data, 0, 0);
else
return agi_exec_full(chan, data, 0, 1);
}
static int eagi_exec(struct ast_channel *chan, const char *data)
{
int res;
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 *readformat;
struct ast_format *requested_format = NULL;
const char *requested_format_name;
if (ast_check_hangup(chan)) {
ast_log(LOG_ERROR, "EAGI cannot be run on a dead/hungup channel, please use AGI.\n");
return 0;
}
requested_format_name = pbx_builtin_getvar_helper(chan, "EAGI_AUDIO_FORMAT");
if (requested_format_name) {
requested_format = ast_format_cache_get(requested_format_name);
if (requested_format) {
ast_verb(3, "<%s> Setting EAGI audio pipe format to %s\n",
ast_channel_name(chan), ast_format_get_name(requested_format));
} else {
ast_log(LOG_ERROR, "Could not find requested format: %s\n", requested_format_name);
}
}
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
readformat = ao2_bump(ast_channel_readformat(chan));
if (ast_set_read_format(chan, requested_format ?: ast_format_slin)) {
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
ast_log(LOG_WARNING, "Unable to set channel '%s' to linear mode\n", ast_channel_name(chan));
ao2_cleanup(requested_format);
ao2_cleanup(readformat);
return -1;
}
res = agi_exec_full(chan, data, 1, 0);
if (!res) {
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 (ast_set_read_format(chan, readformat)) {
ast_log(LOG_WARNING, "Unable to restore channel '%s' to format %s\n", ast_channel_name(chan),
ast_format_get_name(readformat));
}
}
ao2_cleanup(requested_format);
ao2_cleanup(readformat);
return res;
}
static int deadagi_exec(struct ast_channel *chan, const char *data)
{
ast_log(LOG_WARNING, "DeadAGI has been deprecated, please use AGI in all cases!\n");
return agi_exec(chan, data);
}
static struct ast_cli_entry cli_agi[] = {
AST_CLI_DEFINE(handle_cli_agi_add_cmd, "Add AGI command to a channel in Async AGI"),
AST_CLI_DEFINE(handle_cli_agi_debug, "Enable/Disable AGI debugging"),
AST_CLI_DEFINE(handle_cli_agi_show, "List AGI commands or specific help"),
AST_CLI_DEFINE(handle_cli_agi_dump_html, "Dumps a list of AGI commands in HTML format")
};
#ifdef TEST_FRAMEWORK
AST_TEST_DEFINE(test_agi_null_docs)
{
int res = AST_TEST_PASS;
struct agi_command noop_command =
{ { "testnoop", NULL }, handle_noop, NULL, NULL, 0 };
switch (cmd) {
case TEST_INIT:
info->name = "null_agi_docs";
info->category = "/res/agi/";
info->summary = "AGI command with no documentation";
info->description = "Test whether an AGI command with no documentation will crash Asterisk";
return AST_TEST_NOT_RUN;
case TEST_EXECUTE:
break;
}
if (ast_agi_register(ast_module_info->self, &noop_command) == 0) {
ast_test_status_update(test, "Unable to register testnoop command, because res_agi is not loaded.\n");
return AST_TEST_NOT_RUN;
}
#ifndef HAVE_NULLSAFE_PRINTF
/* Test for condition without actually crashing Asterisk */
if (noop_command.usage == NULL) {
ast_test_status_update(test, "AGI testnoop usage was not updated properly.\n");
res = AST_TEST_FAIL;
}
if (noop_command.syntax == NULL) {
ast_test_status_update(test, "AGI testnoop syntax was not updated properly.\n");
res = AST_TEST_FAIL;
}
#endif
ast_agi_unregister(ast_module_info->self, &noop_command);
return res;
}
#endif
static int unload_module(void)
{
STASIS_MESSAGE_TYPE_CLEANUP(agi_exec_start_type);
STASIS_MESSAGE_TYPE_CLEANUP(agi_exec_end_type);
STASIS_MESSAGE_TYPE_CLEANUP(agi_async_start_type);
STASIS_MESSAGE_TYPE_CLEANUP(agi_async_exec_type);
STASIS_MESSAGE_TYPE_CLEANUP(agi_async_end_type);
ast_cli_unregister_multiple(cli_agi, ARRAY_LEN(cli_agi));
ast_agi_unregister_multiple(ast_module_info->self, commands, ARRAY_LEN(commands));
ast_unregister_application(eapp);
ast_unregister_application(deadapp);
ast_manager_unregister("AGI");
ast_unregister_application(app);
AST_TEST_UNREGISTER(test_agi_null_docs);
return 0;
}
static int load_module(void)
{
int err = 0;
err |= STASIS_MESSAGE_TYPE_INIT(agi_exec_start_type);
err |= STASIS_MESSAGE_TYPE_INIT(agi_exec_end_type);
err |= STASIS_MESSAGE_TYPE_INIT(agi_async_start_type);
err |= STASIS_MESSAGE_TYPE_INIT(agi_async_exec_type);
err |= STASIS_MESSAGE_TYPE_INIT(agi_async_end_type);
err |= ast_cli_register_multiple(cli_agi, ARRAY_LEN(cli_agi));
err |= ast_agi_register_multiple(ast_module_info->self, commands, ARRAY_LEN(commands));
err |= ast_register_application_xml(deadapp, deadagi_exec);
err |= ast_register_application_xml(eapp, eagi_exec);
err |= ast_manager_register_xml("AGI", EVENT_FLAG_AGI, action_add_agi_cmd);
err |= ast_register_application_xml(app, agi_exec);
AST_TEST_REGISTER(test_agi_null_docs);
if (err) {
unload_module();
return AST_MODULE_LOAD_DECLINE;
}
/* For Optional API. */
ast_module_shutdown_ref(AST_MODULE_SELF);
return AST_MODULE_LOAD_SUCCESS;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Asterisk Gateway Interface (AGI)",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
.requires = "res_speech",
);