asterisk/main/manager.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 The Asterisk Management Interface - AMI
*
* \author Mark Spencer <markster@digium.com>
*
* OpenSSL http://www.openssl.org - for AMI/SSL
*
* At the moment this file contains a number of functions, namely:
*
* - data structures storing AMI state
* - AMI-related API functions, used by internal asterisk components
* - handlers for AMI-related CLI functions
* - handlers for AMI functions (available through the AMI socket)
* - the code for the main AMI listener thread and individual session threads
* - the http handlers invoked for AMI-over-HTTP by the threads in main/http.c
*
* \ref amiconf
*/
/*! \li \ref manager.c uses the configuration file \ref manager.conf and \ref users.conf
* \addtogroup configuration_file
*/
/*! \page manager.conf manager.conf
* \verbinclude manager.conf.sample
*/
/*! \page users.conf users.conf
* \verbinclude users.conf.sample
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/paths.h" /* use various ast_config_AST_* */
#include <ctype.h>
#include <sys/time.h>
#include <signal.h>
#include <sys/mman.h>
#include <sys/types.h>
#include <regex.h>
#include "asterisk/channel.h"
#include "asterisk/file.h"
#include "asterisk/manager.h"
#include "asterisk/module.h"
#include "asterisk/config.h"
#include "asterisk/callerid.h"
#include "asterisk/core_local.h"
#include "asterisk/lock.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/mwi.h"
#include "asterisk/pbx.h"
#include "asterisk/md5.h"
#include "asterisk/acl.h"
#include "asterisk/utils.h"
#include "asterisk/tcptls.h"
#include "asterisk/http.h"
#include "asterisk/ast_version.h"
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
#include "asterisk/threadstorage.h"
#include "asterisk/linkedlists.h"
#include "asterisk/term.h"
#include "asterisk/astobj2.h"
#include "asterisk/features.h"
#include "asterisk/security_events.h"
#include "asterisk/aoc.h"
#include "asterisk/strings.h"
#include "asterisk/stringfields.h"
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
#include "asterisk/presencestate.h"
#include "asterisk/stasis_message_router.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/stasis_bridges.h"
#include "asterisk/test.h"
#include "asterisk/json.h"
#include "asterisk/bridge.h"
#include "asterisk/features_config.h"
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
#include "asterisk/rtp_engine.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"
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
#include "asterisk/translate.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/message.h"
/*** DOCUMENTATION
<manager name="Ping" language="en_US">
<synopsis>
Keepalive command.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>A 'Ping' action will elicit a 'Pong' response. Used to keep the
manager connection open.</para>
</description>
</manager>
<manager name="Events" language="en_US">
<synopsis>
Control Event Flow.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="EventMask" required="true">
<enumlist>
<enum name="on">
<para>If all events should be sent.</para>
</enum>
<enum name="off">
<para>If no events should be sent.</para>
</enum>
<enum name="system,call,log,...">
<para>To select which flags events should have to be sent.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>Enable/Disable sending of events to this manager client.</para>
</description>
</manager>
<manager name="Logoff" language="en_US">
<synopsis>
Logoff Manager.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Logoff the current manager session.</para>
</description>
<see-also>
<ref type="manager">Login</ref>
</see-also>
</manager>
<manager name="Login" language="en_US">
<synopsis>
Login Manager.
</synopsis>
<syntax>
<parameter name="ActionID">
<para>ActionID for this transaction. Will be returned.</para>
</parameter>
<parameter name="Username" required="true">
<para>Username to login with as specified in manager.conf.</para>
</parameter>
<parameter name="Secret">
<para>Secret to login with as specified in manager.conf.</para>
</parameter>
</syntax>
<description>
<para>Login Manager.</para>
</description>
<see-also>
<ref type="manager">Logoff</ref>
</see-also>
</manager>
<manager name="Challenge" language="en_US">
<synopsis>
Generate Challenge for MD5 Auth.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="AuthType" required="true">
<para>Digest algorithm to use in the challenge. Valid values are:</para>
<enumlist>
<enum name="MD5" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Generate a challenge for MD5 authentication.</para>
</description>
</manager>
<manager name="Hangup" language="en_US">
<synopsis>
Hangup channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The exact channel name to be hungup, or to use a regular expression, set this parameter to: /regex/</para>
<para>Example exact channel: SIP/provider-0000012a</para>
<para>Example regular expression: /^SIP/provider-.*$/</para>
</parameter>
<parameter name="Cause">
<para>Numeric hangup cause.</para>
</parameter>
</syntax>
<description>
<para>Hangup a channel.</para>
</description>
</manager>
<manager name="Status" language="en_US">
<synopsis>
List channel status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="false">
<para>The name of the channel to query for status.</para>
</parameter>
<parameter name="Variables">
<para>Comma <literal>,</literal> separated list of variable to include.</para>
</parameter>
<parameter name="AllVariables">
<para>If set to "true", the Status event will include all channel variables for
the requested channel(s).</para>
<enumlist>
<enum name="true"/>
<enum name="false"/>
</enumlist>
</parameter>
</syntax>
<description>
<para>Will return the status information of each channel along with the
value for the specified channel variables.</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='Status'])" />
</list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='StatusComplete'])" />
</responses>
</manager>
<managerEvent language="en_US" name="Status">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a Status command.</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<channel_snapshot/>
<parameter name="Type">
<para>Type of channel</para>
</parameter>
<parameter name="DNID">
<para>Dialed number identifier</para>
</parameter>
<parameter name="EffectiveConnectedLineNum">
</parameter>
<parameter name="EffectiveConnectedLineName">
</parameter>
<parameter name="TimeToHangup">
<para>Absolute lifetime of the channel</para>
</parameter>
<parameter name="BridgeID">
<para>Identifier of the bridge the channel is in, may be empty if not in one</para>
</parameter>
<parameter name="Application">
<para>Application currently executing on the channel</para>
</parameter>
<parameter name="Data">
<para>Data given to the currently executing channel</para>
</parameter>
<parameter name="Nativeformats">
<para>Media formats the connected party is willing to send or receive</para>
</parameter>
<parameter name="Readformat">
<para>Media formats that frames from the channel are received in</para>
</parameter>
<parameter name="Readtrans">
<para>Translation path for media received in native formats</para>
</parameter>
<parameter name="Writeformat">
<para>Media formats that frames to the channel are accepted in</para>
</parameter>
<parameter name="Writetrans">
<para>Translation path for media sent to the connected party</para>
</parameter>
<parameter name="Callgroup">
<para>Configured call group on the channel</para>
</parameter>
<parameter name="Pickupgroup">
<para>Configured pickup group on the channel</para>
</parameter>
<parameter name="Seconds">
<para>Number of seconds the channel has been active</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">Status</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="StatusComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a Status command.</synopsis>
<syntax>
<parameter name="Items">
<para>Number of Status events returned</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">Status</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="Setvar" language="en_US">
<synopsis>
Sets a channel variable or function value.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel">
<para>Channel to set variable for.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Variable name, function or expression.</para>
</parameter>
<parameter name="Value" required="true">
<para>Variable or function value.</para>
</parameter>
</syntax>
<description>
<para>This command can be used to set the value of channel variables or dialplan
functions.</para>
<note>
<para>If a channel name is not provided then the variable is considered global.</para>
</note>
</description>
<see-also>
<ref type="manager">Getvar</ref>
</see-also>
</manager>
<manager name="Getvar" language="en_US">
<synopsis>
Gets a channel variable or function value.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel">
<para>Channel to read variable from.</para>
</parameter>
<parameter name="Variable" required="true">
<para>Variable name, function or expression.</para>
</parameter>
</syntax>
<description>
<para>Get the value of a channel variable or function return.</para>
<note>
<para>If a channel name is not provided then the variable is considered global.</para>
</note>
</description>
<see-also>
<ref type="manager">Setvar</ref>
</see-also>
</manager>
<manager name="GetConfig" language="en_US">
<synopsis>
Retrieve configuration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
<parameter name="Category">
<para>Category in configuration file.</para>
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Filter">
<para>A comma separated list of
<replaceable>name_regex</replaceable>=<replaceable>value_regex</replaceable>
expressions which will cause only categories whose variables match all expressions
to be considered. The special variable name <literal>TEMPLATES</literal>
can be used to control whether templates are included. Passing
<literal>include</literal> as the value will include templates
along with normal categories. Passing
<literal>restrict</literal> as the value will restrict the operation to
ONLY templates. Not specifying a <literal>TEMPLATES</literal> expression
results in the default behavior which is to not include templates.</para>
</parameter>
</syntax>
<description>
<para>This action will dump the contents of a configuration
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
file by category and contents or optionally by specified category only.
In the case where a category name is non-unique, a filter may be specified
to match only categories with matching variable values.</para>
</description>
<see-also>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="GetConfigJSON" language="en_US">
<synopsis>
Retrieve configuration (JSON format).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Category">
<para>Category in configuration file.</para>
</parameter>
<parameter name="Filter">
<xi:include xpointer="xpointer(/docs/manager[@name='GetConfig']/syntax/parameter[@name='Filter']/para[1])" />
</parameter>
</syntax>
<description>
<para>This action will dump the contents of a configuration file by category
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
and contents in JSON format or optionally by specified category only.
This only makes sense to be used using rawman over the HTTP interface.
In the case where a category name is non-unique, a filter may be specified
to match only categories with matching variable values.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="UpdateConfig" language="en_US">
<synopsis>
Update basic configuration.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="SrcFilename" required="true">
<para>Configuration filename to read (e.g. <filename>foo.conf</filename>).</para>
</parameter>
<parameter name="DstFilename" required="true">
<para>Configuration filename to write (e.g. <filename>foo.conf</filename>)</para>
</parameter>
<parameter name="Reload">
<para>Whether or not a reload should take place (or name of specific module).</para>
</parameter>
config: Add option to NOT preserve effective context when changing a template Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ ........ Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07 16:56:59 +00:00
<parameter name="PreserveEffectiveContext">
<para>Whether the effective category contents should be preserved on template change. Default is true (pre 13.2 behavior).</para>
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Action-000000">
<para>Action to take.</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<para>0's represent 6 digit number beginning with 000000.</para>
<enumlist>
<enum name="NewCat" />
<enum name="RenameCat" />
<enum name="DelCat" />
<enum name="EmptyCat" />
<enum name="Update" />
<enum name="Delete" />
<enum name="Append" />
<enum name="Insert" />
</enumlist>
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Cat-000000">
<para>Category to operate on.</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Var-000000">
<para>Variable to work on.</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Value-000000">
<para>Value to work on.</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Match-000000">
<para>Extra match required to match line.</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<parameter name="Line-000000">
<para>Line in category to operate on (used with delete and insert actions).</para>
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
<parameter name="Options-000000">
<para>A comma separated list of action-specific options.</para>
<enumlist>
<enum name="NewCat"><para>One or more of the following... </para>
<enumlist>
<enum name="allowdups"><para>Allow duplicate category names.</para></enum>
<enum name="template"><para>This category is a template.</para></enum>
<enum name="inherit=&quot;template[,...]&quot;"><para>Templates from which to inherit.</para></enum>
</enumlist>
</enum>
</enumlist>
<para> </para>
<para>The following actions share the same options...</para>
<enumlist>
<enum name="RenameCat"/>
<enum name="DelCat"/>
<enum name="EmptyCat"/>
<enum name="Update"/>
<enum name="Delete"/>
<enum name="Append"/>
<enum name="Insert"><para> </para>
<enumlist>
<enum name="catfilter=&quot;&lt;expression&gt;[,...]&quot;"><para> </para>
<xi:include xpointer="xpointer(/docs/manager[@name='GetConfig']/syntax/parameter[@name='Filter']/para[1])" />
<para><literal>catfilter</literal> is most useful when a file
contains multiple categories with the same name and you wish to
operate on specific ones instead of all of them.</para>
</enum>
</enumlist>
</enum>
</enumlist>
<xi:include xpointer="xpointer(/docs/manager[@name='UpdateConfig']/syntax/parameter[@name='Action-000000']/para[2])" />
</parameter>
</syntax>
<description>
<para>This action will modify, create, or delete configuration elements
in Asterisk configuration files.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">CreateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="CreateConfig" language="en_US">
<synopsis>
Creates an empty file in the configuration directory.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>The configuration filename to create (e.g. <filename>foo.conf</filename>).</para>
</parameter>
</syntax>
<description>
<para>This action will create an empty file in the configuration
directory. This action is intended to be used before an UpdateConfig
action.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">ListCategories</ref>
</see-also>
</manager>
<manager name="ListCategories" language="en_US">
<synopsis>
List categories in configuration file.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Filename" required="true">
<para>Configuration filename (e.g. <filename>foo.conf</filename>).</para>
</parameter>
</syntax>
<description>
<para>This action will dump the categories in a given file.</para>
</description>
<see-also>
<ref type="manager">GetConfig</ref>
<ref type="manager">GetConfigJSON</ref>
<ref type="manager">UpdateConfig</ref>
<ref type="manager">CreateConfig</ref>
</see-also>
</manager>
<manager name="Redirect" language="en_US">
<synopsis>
Redirect (transfer) a call.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel to redirect.</para>
</parameter>
<parameter name="ExtraChannel">
<para>Second call leg to transfer (optional).</para>
</parameter>
<parameter name="Exten" required="true">
<para>Extension to transfer to.</para>
</parameter>
<parameter name="ExtraExten">
<para>Extension to transfer extrachannel to (optional).</para>
</parameter>
<parameter name="Context" required="true">
<para>Context to transfer to.</para>
</parameter>
<parameter name="ExtraContext">
<para>Context to transfer extrachannel to (optional).</para>
</parameter>
<parameter name="Priority" required="true">
<para>Priority to transfer to.</para>
</parameter>
<parameter name="ExtraPriority">
<para>Priority to transfer extrachannel to (optional).</para>
</parameter>
</syntax>
<description>
<para>Redirect (transfer) a call.</para>
</description>
<see-also>
<ref type="manager">BlindTransfer</ref>
</see-also>
</manager>
<manager name="Atxfer" language="en_US">
<synopsis>
Attended transfer.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Transferer's channel.</para>
</parameter>
<parameter name="Exten" required="true">
<para>Extension to transfer to.</para>
</parameter>
<parameter name="Context">
<para>Context to transfer to.</para>
</parameter>
</syntax>
<description>
<para>Attended transfer.</para>
</description>
<see-also>
<ref type="managerEvent">AttendedTransfer</ref>
</see-also>
</manager>
<manager name="CancelAtxfer" language="en_US">
<since>
<version>13.18.0</version>
<version>14.7.0</version>
<version>15.1.0</version>
<version>16.0.0</version>
</since>
<synopsis>
Cancel an attended transfer.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The transferer channel.</para>
</parameter>
</syntax>
<description>
<para>Cancel an attended transfer. Note, this uses the configured cancel attended transfer
feature option (atxferabort) to cancel the transfer. If not available this action will fail.
</para>
</description>
<see-also>
<ref type="managerEvent">AttendedTransfer</ref>
</see-also>
</manager>
<manager name="Originate" language="en_US">
<synopsis>
Originate a call.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to call.</para>
</parameter>
<parameter name="Exten">
<para>Extension to use (requires <literal>Context</literal> and
<literal>Priority</literal>)</para>
</parameter>
<parameter name="Context">
<para>Context to use (requires <literal>Exten</literal> and
<literal>Priority</literal>)</para>
</parameter>
<parameter name="Priority">
<para>Priority to use (requires <literal>Exten</literal> and
<literal>Context</literal>)</para>
</parameter>
<parameter name="Application">
<para>Application to execute.</para>
</parameter>
<parameter name="Data">
<para>Data to use (requires <literal>Application</literal>).</para>
</parameter>
<parameter name="Timeout" default="30000">
<para>How long to wait for call to be answered (in ms.).</para>
</parameter>
<parameter name="CallerID">
<para>Caller ID to be set on the outgoing channel.</para>
</parameter>
<parameter name="Variable">
<para>Channel variable to set, multiple Variable: headers are allowed.</para>
</parameter>
<parameter name="Account">
<para>Account code.</para>
</parameter>
<parameter name="EarlyMedia">
<para>Set to <literal>true</literal> to force call bridge on early media..</para>
</parameter>
<parameter name="Async">
<para>Set to <literal>true</literal> for fast origination.</para>
</parameter>
<parameter name="Codecs">
<para>Comma-separated list of codecs to use for this call.</para>
</parameter>
<parameter name="ChannelId">
<para>Channel UniqueId to be set on the channel.</para>
</parameter>
<parameter name="OtherChannelId">
<para>Channel UniqueId to be set on the second local channel.</para>
</parameter>
</syntax>
<description>
<para>Generates an outgoing call to a
<replaceable>Extension</replaceable>/<replaceable>Context</replaceable>/<replaceable>Priority</replaceable>
or <replaceable>Application</replaceable>/<replaceable>Data</replaceable></para>
</description>
<see-also>
<ref type="managerEvent">OriginateResponse</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="OriginateResponse">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to an Originate command.</synopsis>
<syntax>
<parameter name="ActionID" required="false"/>
<parameter name="Response">
<enumlist>
<enum name="Failure"/>
<enum name="Success"/>
</enumlist>
</parameter>
<parameter name="Channel"/>
<parameter name="Context"/>
<parameter name="Exten"/>
<parameter name="Application"/>
<parameter name="Data"/>
<parameter name="Reason"/>
<parameter name="Uniqueid"/>
<parameter name="CallerIDNum"/>
<parameter name="CallerIDName"/>
</syntax>
<see-also>
<ref type="manager">Originate</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="Command" language="en_US">
<synopsis>
Execute Asterisk CLI Command.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Command" required="true">
<para>Asterisk CLI command to run.</para>
</parameter>
</syntax>
<description>
<para>Run a CLI command.</para>
</description>
</manager>
<manager name="ExtensionState" language="en_US">
<synopsis>
Check Extension Status.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Exten" required="true">
<para>Extension to check state on.</para>
</parameter>
<parameter name="Context" required="true">
<para>Context for extension.</para>
</parameter>
</syntax>
<description>
<para>Report the extension state for given extension. If the extension has a hint,
will use devicestate to check the status of the device connected to the extension.</para>
<para>Will return an <literal>Extension Status</literal> message. The response will include
the hint for the extension and the status.</para>
</description>
<see-also>
<ref type="managerEvent">ExtensionStatus</ref>
</see-also>
</manager>
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
<manager name="PresenceState" language="en_US">
<synopsis>
Check Presence State
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Provider" required="true">
<para>Presence Provider to check the state of</para>
</parameter>
</syntax>
<description>
<para>Report the presence state for the given presence provider.</para>
<para>Will return a <literal>Presence State</literal> message. The response will include the
presence state and, if set, a presence subtype and custom message.</para>
</description>
<see-also>
<ref type="managerEvent">PresenceStatus</ref>
</see-also>
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
</manager>
<manager name="AbsoluteTimeout" language="en_US">
<synopsis>
Set absolute timeout.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel name to hangup.</para>
</parameter>
<parameter name="Timeout" required="true">
<para>Maximum duration of the call (sec).</para>
</parameter>
</syntax>
<description>
<para>Hangup a channel after a certain time. Acknowledges set time with
<literal>Timeout Set</literal> message.</para>
</description>
</manager>
<manager name="MailboxStatus" language="en_US">
<synopsis>
Check mailbox.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Full mailbox ID <replaceable>mailbox</replaceable>@<replaceable>vm-context</replaceable>.</para>
</parameter>
</syntax>
<description>
<para>Checks a voicemail account for status.</para>
<para>Returns whether there are messages waiting.</para>
<para>Message: Mailbox Status.</para>
<para>Mailbox: <replaceable>mailboxid</replaceable>.</para>
<para>Waiting: <literal>0</literal> if messages waiting, <literal>1</literal>
if no messages waiting.</para>
</description>
<see-also>
<ref type="manager">MailboxCount</ref>
</see-also>
</manager>
<manager name="MailboxCount" language="en_US">
<synopsis>
Check Mailbox Message Count.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Mailbox" required="true">
<para>Full mailbox ID <replaceable>mailbox</replaceable>@<replaceable>vm-context</replaceable>.</para>
</parameter>
</syntax>
<description>
<para>Checks a voicemail account for new messages.</para>
<para>Returns number of urgent, new and old messages.</para>
<para>Message: Mailbox Message Count</para>
<para>Mailbox: <replaceable>mailboxid</replaceable></para>
<para>UrgentMessages: <replaceable>count</replaceable></para>
<para>NewMessages: <replaceable>count</replaceable></para>
<para>OldMessages: <replaceable>count</replaceable></para>
</description>
<see-also>
<ref type="manager">MailboxStatus</ref>
</see-also>
</manager>
<manager name="ListCommands" language="en_US">
<synopsis>
List available manager commands.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Returns the action name and synopsis for every action that
is available to the user.</para>
</description>
</manager>
<manager name="SendText" language="en_US">
<synopsis>
Sends a text message to channel. A content type can be optionally specified. If not set
it is set to an empty string allowing a custom handler to default it as it sees fit.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>Channel to send message to.</para>
</parameter>
<parameter name="Message" required="true">
<para>Message to send.</para>
</parameter>
<parameter name="Content-Type" required="false" default="">
<para>The type of content in the message</para>
</parameter>
</syntax>
<description>
<para>Sends A Text Message to a channel while in a call.</para>
</description>
<see-also>
<ref type="application">SendText</ref>
</see-also>
</manager>
<manager name="UserEvent" language="en_US">
<synopsis>
Send an arbitrary event.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="UserEvent" required="true">
<para>Event string to send.</para>
</parameter>
<parameter name="Header1">
<para>Content1.</para>
</parameter>
<parameter name="HeaderN">
<para>ContentN.</para>
</parameter>
</syntax>
<description>
<para>Send an event to manager sessions.</para>
</description>
<see-also>
<ref type="managerEvent">UserEvent</ref>
<ref type="application">UserEvent</ref>
</see-also>
</manager>
<manager name="WaitEvent" language="en_US">
<synopsis>
Wait for an event to occur.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Timeout" required="true">
<para>Maximum time (in seconds) to wait for events, <literal>-1</literal> means forever.</para>
</parameter>
</syntax>
<description>
<para>This action will elicit a <literal>Success</literal> response. Whenever
a manager event is queued. Once WaitEvent has been called on an HTTP manager
session, events will be generated and queued.</para>
</description>
</manager>
<manager name="CoreSettings" language="en_US">
<synopsis>
Show PBX core settings (version etc).
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Query for Core PBX settings.</para>
</description>
</manager>
<manager name="CoreStatus" language="en_US">
<synopsis>
Show PBX core status variables.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Query for Core PBX status.</para>
</description>
</manager>
<manager name="Reload" language="en_US">
<synopsis>
Send a reload event.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module">
<para>Name of the module to reload.</para>
</parameter>
</syntax>
<description>
<para>Send a reload event.</para>
</description>
<see-also>
<ref type="manager">ModuleLoad</ref>
</see-also>
</manager>
<managerEvent language="en_US" name="CoreShowChannel">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised in response to a CoreShowChannels command.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<channel_snapshot/>
<parameter name="BridgeId">
<para>Identifier of the bridge the channel is in, may be empty if not in one</para>
</parameter>
<parameter name="Application">
<para>Application currently executing on the channel</para>
</parameter>
<parameter name="ApplicationData">
<para>Data given to the currently executing application</para>
</parameter>
<parameter name="Duration">
<para>The amount of time the channel has existed</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">CoreShowChannels</ref>
<ref type="managerEvent">CoreShowChannelsComplete</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="CoreShowChannelsComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised at the end of the CoreShowChannel list produced by the CoreShowChannels command.</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="EventList">
<para>Conveys the status of the command reponse list</para>
</parameter>
<parameter name="ListItems">
<para>The total number of list items produced</para>
</parameter>
</syntax>
<see-also>
<ref type="manager">CoreShowChannels</ref>
<ref type="managerEvent">CoreShowChannel</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<manager name="CoreShowChannels" language="en_US">
<synopsis>
List currently active channels.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>List currently defined channels and some information about them.</para>
</description>
<responses>
<list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='CoreShowChannel'])" />
</list-elements>
<xi:include xpointer="xpointer(/docs/managerEvent[@name='CoreShowChannelsComplete'])" />
</responses>
</manager>
<managerEvent language="en_US" name="CoreShowChannelMapComplete">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised at the end of the CoreShowChannelMap list produced by the CoreShowChannelMap command.</synopsis>
<syntax>
<parameter name="EventList">
<para>Conveys the status of the command response list</para>
</parameter>
<parameter name="ListItems">
<para>The total number of list items produced</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<manager name="CoreShowChannelMap" language="en_US">
<synopsis>
List all channels connected to the specified channel.
</synopsis>
<syntax>
<parameter name="Channel">
<para>The channel to get the mapping for. Requires a channel name.</para>
</parameter>
</syntax>
<description>
<para>List all channels currently connected to the specified channel. This can be any channel, including
Local channels, and Local channels will be followed through to their other half.</para>
</description>
</manager>
<manager name="LoggerRotate" language="en_US">
<synopsis>
Reload and rotate the Asterisk logger.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
</syntax>
<description>
<para>Reload and rotate the logger. Analogous to the CLI command 'logger rotate'.</para>
</description>
</manager>
<manager name="ModuleLoad" language="en_US">
<synopsis>
Module management.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module">
<para>Asterisk module name (including .so extension) or subsystem identifier:</para>
<enumlist>
<enum name="cdr" />
<enum name="dnsmgr" />
<enum name="extconfig" />
<enum name="enum" />
<enum name="acl" />
<enum name="manager" />
<enum name="http" />
<enum name="logger" />
<enum name="features" />
<enum name="dsp" />
<enum name="udptl" />
<enum name="indications" />
<enum name="cel" />
<enum name="plc" />
</enumlist>
</parameter>
<parameter name="LoadType" required="true">
<para>The operation to be done on module. Subsystem identifiers may only
be reloaded.</para>
<enumlist>
<enum name="load" />
<enum name="unload" />
<enum name="reload" />
</enumlist>
<para>If no module is specified for a <literal>reload</literal> loadtype,
all modules are reloaded.</para>
</parameter>
</syntax>
<description>
<para>Loads, unloads or reloads an Asterisk module in a running system.</para>
</description>
<see-also>
<ref type="manager">Reload</ref>
<ref type="manager">ModuleCheck</ref>
</see-also>
</manager>
<manager name="ModuleCheck" language="en_US">
<synopsis>
Check if module is loaded.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Module" required="true">
<para>Asterisk module name (not including extension).</para>
</parameter>
</syntax>
<description>
<para>Checks if Asterisk module is loaded. Will return Success/Failure.
An empty Version header is also returned (which doesn't contain
the module revision number).</para>
</description>
<see-also>
<ref type="manager">ModuleLoad</ref>
</see-also>
</manager>
<manager name="AOCMessage" language="en_US">
<synopsis>
Generate an Advice of Charge message on a channel.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel">
<para>Channel name to generate the AOC message on.
This value is required unless ChannelPrefix is given.</para>
</parameter>
<parameter name="ChannelPrefix">
<para>Partial channel prefix. By using this option one can match the beginning part
of a channel name without having to put the entire name in. For example
if a channel name is SIP/snom-00000001 and this value is set to SIP/snom, then
that channel matches and the message will be sent. Note however that only
the first matched channel has the message sent on it. </para>
</parameter>
<parameter name="MsgType" required="true">
<para>Defines what type of AOC message to create, AOC-S, AOC-D or AOC-E</para>
<enumlist>
<enum name="S" />
<enum name="D" />
<enum name="E" />
</enumlist>
</parameter>
<parameter name="ChargeType">
<para>Defines what kind of charge this message represents for AOC-D and AOC-E.</para>
<enumlist>
<enum name="NA" />
<enum name="FREE" />
<enum name="Currency" />
<enum name="Unit" />
</enumlist>
</parameter>
<parameter name="UnitAmount(0)">
<para>This represents the amount of units charged. The ETSI AOC standard specifies that
this value along with the optional UnitType value are entries in a list. To accommodate this
these values take an index value starting at 0 which can be used to generate this list of
unit entries. For Example, If two unit entires were required this could be achieved by setting the
paramter UnitAmount(0)=1234 and UnitAmount(1)=5678. Note that UnitAmount at index 0 is
required when ChargeType=Unit, all other entries in the list are optional.
</para>
</parameter>
<parameter name="UnitType(0)">
<para>Defines the type of unit. ETSI AOC standard specifies this as an integer
value between 1 and 16, but this value is left open to accept any positive
integer. Like the UnitAmount parameter, this value represents a list entry
and has an index parameter that starts at 0.
</para>
</parameter>
<parameter name="CurrencyName">
<para>Specifies the currency's name. Note that this value is truncated after 10 characters.</para>
</parameter>
<parameter name="CurrencyAmount">
<para>Specifies the charge unit amount as a positive integer.
This value is required when ChargeType==Currency (AOC-D or AOC-E) or
RateType==Duration/Flat/Volume (AOC-S).</para>
</parameter>
<parameter name="CurrencyMultiplier">
<para>Specifies the currency multiplier.
This value is required when CurrencyAmount is given.</para>
<enumlist>
<enum name="OneThousandth" />
<enum name="OneHundredth" />
<enum name="OneTenth" />
<enum name="One" />
<enum name="Ten" />
<enum name="Hundred" />
<enum name="Thousand" />
</enumlist>
</parameter>
<parameter name="TotalType" default="Total">
<para>Defines what kind of AOC-D total is represented.</para>
<enumlist>
<enum name="Total" />
<enum name="SubTotal" />
</enumlist>
</parameter>
<parameter name="AOCBillingId">
<para>Represents a billing ID associated with an AOC-D or AOC-E message. Note
that only the first 3 items of the enum are valid AOC-D billing IDs</para>
<enumlist>
<enum name="Normal" />
<enum name="ReverseCharge" />
<enum name="CreditCard" />
<enum name="CallFwdUnconditional" />
<enum name="CallFwdBusy" />
<enum name="CallFwdNoReply" />
<enum name="CallDeflection" />
<enum name="CallTransfer" />
</enumlist>
</parameter>
<parameter name="ChargingAssociationId">
<para>Charging association identifier. This is optional for AOC-E and can be
set to any value between -32768 and 32767</para>
</parameter>
<parameter name="ChargingAssociationNumber">
<para>Represents the charging association party number. This value is optional
for AOC-E.</para>
</parameter>
<parameter name="ChargingAssociationPlan">
<para>Integer representing the charging plan associated with the ChargingAssociationNumber.
The value is bits 7 through 1 of the Q.931 octet containing the type-of-number and
numbering-plan-identification fields.</para>
</parameter>
<parameter name="ChargedItem">
<para>Defines what part of the call is charged in AOC-S. Usually this is set to
BasicCommunication, which refers to the time after the call is answered, but establishment
(CallAttempt) or successful establishment (CallSetup) of a call can also be used.
Other options are available, but these generally do not carry enough information to actually
calculate the price of a call.
It is possible to have multiple ChargedItem entries for a single call -- for example to
charge for both the establishment of the call and the actual call. In this case, each
ChargedItem is described by a ChargedItem: header and all other headers that follow it up to
the next ChargedItem: header.</para>
<enumlist>
<enum name="NA" />
<enum name="SpecialArrangement" />
<enum name="BasicCommunication" />
<enum name="CallAttempt" />
<enum name="CallSetup" />
<enum name="UserUserInfo" />
<enum name="SupplementaryService" />
</enumlist>
</parameter>
<parameter name="RateType">
<para>Defines how an AOC-S ChargedItem is charged.
The Duration option is only available when ChargedItem==BasicCommunication.</para>
<enumlist>
<enum name="NA" />
<enum name="Free" />
<enum name="FreeFromBeginning" />
<enum name="Duration" />
<enum name="Flat" />
<enum name="Volume" />
<enum name="SpecialCode" />
</enumlist>
</parameter>
<parameter name="Time">
<para>Specifies a positive integer which is the amount of time is paid for by one
CurrencyAmount.
This value is required when RateType==Duration.</para>
</parameter>
<parameter name="TimeScale">
<para>Specifies the time multiplier.
This value is required when Time is given.</para>
<enumlist>
<enum name="OneHundredthSecond" />
<enum name="OneTenthSecond" />
<enum name="Second" />
<enum name="TenSeconds" />
<enum name="Minute" />
<enum name="Hour" />
<enum name="Day" />
</enumlist>
</parameter>
<parameter name="Granularity">
<para>Specifies a positive integer which is the size of the charged time increments.
This value is optional when RateType==Duration and ChargingType==StepFunction.</para>
</parameter>
<parameter name="GranularityTimeScale">
<para>Specifies the granularity time multiplier.
This value is required when Granularity is given.</para>
<enumlist>
<enum name="OneHundredthSecond" />
<enum name="OneTenthSecond" />
<enum name="Second" />
<enum name="TenSeconds" />
<enum name="Minute" />
<enum name="Hour" />
<enum name="Day" />
</enumlist>
</parameter>
<parameter name="ChargingType">
<para>Specifies whether the charge increases continuously with time or in increments of
Time or, if provided, Granularity.
This value is required when RateType==Duration.</para>
<enumlist>
<enum name="ContinuousCharging" />
<enum name="StepFunction" />
</enumlist>
</parameter>
<parameter name="VolumeUnit">
<para>Specifies the quantity of which one unit is paid for by one CurrencyAmount.
This value is required when RateType==Volume.</para>
<enumlist>
<enum name="Octet" />
<enum name="Segment" />
<enum name="Message" />
</enumlist>
</parameter>
<parameter name="Code">
<para>Specifies the charging code, which can be set to a value between 1 and 10.
This value is required when ChargedItem==SpecialArrangement or RateType==SpecialCode.</para>
</parameter>
</syntax>
<description>
<para>Generates an AOC-S, AOC-D or AOC-E message on a channel.</para>
</description>
<see-also>
<ref type="managerEvent">AOC-S</ref>
<ref type="managerEvent">AOC-D</ref>
<ref type="managerEvent">AOC-E</ref>
</see-also>
</manager>
<function name="AMI_CLIENT" language="en_US">
<synopsis>
Checks attributes of manager accounts
</synopsis>
<syntax>
<parameter name="loginname" required="true">
<para>Login name, specified in manager.conf</para>
</parameter>
<parameter name="field" required="true">
<para>The manager account attribute to return</para>
<enumlist>
<enum name="sessions"><para>The number of sessions for this AMI account</para></enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>
Currently, the only supported parameter is "sessions" which will return the current number of
active sessions for this AMI account.
</para>
</description>
</function>
<manager name="Filter" language="en_US">
<synopsis>
Dynamically add filters for the current manager session.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Operation">
<enumlist>
<enum name="Add">
<para>Add a filter.</para>
</enum>
</enumlist>
</parameter>
<parameter name="Filter">
<para>Filters can be whitelist or blacklist</para>
<para>Example whitelist filter: "Event: Newchannel"</para>
<para>Example blacklist filter: "!Channel: DAHDI.*"</para>
<para>This filter option is used to whitelist or blacklist events per user to be
reported with regular expressions and are allowed if both the regex matches
and the user has read access as defined in manager.conf. Filters are assumed to be for whitelisting
unless preceeded by an exclamation point, which marks it as being black.
Evaluation of the filters is as follows:</para>
<para>- If no filters are configured all events are reported as normal.</para>
<para>- If there are white filters only: implied black all filter processed first, then white filters.</para>
<para>- If there are black filters only: implied white all filter processed first, then black filters.</para>
<para>- If there are both white and black filters: implied black all filter processed first, then white
filters, and lastly black filters.</para>
</parameter>
</syntax>
<description>
<para>The filters added are only used for the current session.
Once the connection is closed the filters are removed.</para>
<para>This comand requires the system permission because
this command can be used to create filters that may bypass
filters defined in manager.conf</para>
</description>
</manager>
<manager name="BlindTransfer" language="en_US">
<synopsis>
Blind transfer channel(s) to the given destination
</synopsis>
<syntax>
<parameter name="Channel" required="true">
</parameter>
<parameter name="Context">
</parameter>
<parameter name="Exten">
</parameter>
</syntax>
<description>
<para>Redirect all channels currently bridged to the specified channel to the specified destination.</para>
</description>
<see-also>
<ref type="manager">Redirect</ref>
<ref type="managerEvent">BlindTransfer</ref>
</see-also>
</manager>
<managerEvent name="ExtensionStatus" language="en_US">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a hint changes due to a device state change.</synopsis>
<syntax>
<parameter name="Exten">
<para>Name of the extension.</para>
</parameter>
<parameter name="Context">
<para>Context that owns the extension.</para>
</parameter>
<parameter name="Hint">
<para>Hint set for the extension</para>
</parameter>
<parameter name="Status">
<para>Numerical value of the extension status. Extension
status is determined by the combined device state of all items
contained in the hint.</para>
<enumlist>
<enum name="-2">
<para>The extension was removed from the dialplan.</para>
</enum>
<enum name="-1">
<para>The extension's hint was removed from the dialplan.</para>
</enum>
<enum name="0">
<para><literal>Idle</literal> - Related device(s) are in an idle
state.</para>
</enum>
<enum name="1">
<para><literal>InUse</literal> - Related device(s) are in active
calls but may take more calls.</para>
</enum>
<enum name="2">
<para><literal>Busy</literal> - Related device(s) are in active
calls and may not take any more calls.</para>
</enum>
<enum name="4">
<para><literal>Unavailable</literal> - Related device(s) are
not reachable.</para>
</enum>
<enum name="8">
<para><literal>Ringing</literal> - Related device(s) are
currently ringing.</para>
</enum>
<enum name="9">
<para><literal>InUse&amp;Ringing</literal> - Related device(s)
are currently ringing and in active calls.</para>
</enum>
<enum name="16">
<para><literal>Hold</literal> - Related device(s) are
currently on hold.</para>
</enum>
<enum name="17">
<para><literal>InUse&amp;Hold</literal> - Related device(s)
are currently on hold and in active calls.</para>
</enum>
</enumlist>
</parameter>
<parameter name="StatusText">
<para>Text representation of <literal>Status</literal>.</para>
<enumlist>
<enum name="Idle" />
<enum name="InUse" />
<enum name="Busy" />
<enum name="Unavailable" />
<enum name="Ringing" />
<enum name="InUse&amp;Ringing" />
<enum name="Hold" />
<enum name="InUse&amp;Hold" />
<enum name="Unknown">
<para>Status does not match any of the above values.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<see-also>
<ref type="manager">ExtensionState</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent name="PresenceStatus" language="en_US">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when a hint changes due to a presence state change.</synopsis>
<syntax>
<parameter name="Exten" />
<parameter name="Context" />
<parameter name="Hint" />
<parameter name="Status" />
<parameter name="Subtype" />
<parameter name="Message" />
</syntax>
<see-also>
<ref type="manager">PresenceState</ref>
</see-also>
</managerEventInstance>
</managerEvent>
***/
/*! \addtogroup Group_AMI AMI functions
*/
/*! @{
Doxygen group */
enum error_type {
UNKNOWN_ACTION = 1,
UNKNOWN_CATEGORY,
UNSPECIFIED_CATEGORY,
UNSPECIFIED_ARGUMENT,
FAILURE_ALLOCATION,
FAILURE_NEWCAT,
FAILURE_DELCAT,
FAILURE_EMPTYCAT,
FAILURE_UPDATE,
FAILURE_DELETE,
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
FAILURE_APPEND,
FAILURE_TEMPLATE
};
enum add_filter_result {
FILTER_SUCCESS,
FILTER_ALLOC_FAILED,
FILTER_COMPILE_FAIL,
};
/*!
* Linked list of events.
* Global events are appended to the list by append_event().
* The usecount is the number of stored pointers to the element,
* excluding the list pointers. So an element that is only in
* the list has a usecount of 0, not 1.
*
* Clients have a pointer to the last event processed, and for each
* of these clients we track the usecount of the elements.
* If we have a pointer to an entry in the list, it is safe to navigate
* it forward because elements will not be deleted, but only appended.
* The worst that can happen is seeing the pointer still NULL.
*
* When the usecount of an element drops to 0, and the element is the
* first in the list, we can remove it. Removal is done within the
* main thread, which is woken up for the purpose.
*
* For simplicity of implementation, we make sure the list is never empty.
*/
struct eventqent {
int usecount; /*!< # of clients who still need the event */
int category;
unsigned int seq; /*!< sequence number */
struct timeval tv; /*!< When event was allocated */
AST_RWLIST_ENTRY(eventqent) eq_next;
char eventdata[1]; /*!< really variable size, allocated by append_event() */
};
static AST_RWLIST_HEAD_STATIC(all_events, eventqent);
static int displayconnects = 1;
static int allowmultiplelogin = 1;
static int timestampevents;
static int httptimeout = 60;
static int broken_events_action = 0;
static int manager_enabled = 0;
static int subscribed = 0;
static int webmanager_enabled = 0;
static int manager_debug = 0; /*!< enable some debugging code in the manager */
static int authtimeout;
static int authlimit;
static char *manager_channelvars;
static char *manager_disabledevents;
#define DEFAULT_REALM "asterisk"
static char global_realm[MAXHOSTNAMELEN]; /*!< Default realm */
static int unauth_sessions = 0;
static struct stasis_subscription *acl_change_sub;
/*! \brief A \ref stasis_topic that all topics AMI cares about will be forwarded to */
static struct stasis_topic *manager_topic;
/*! \brief The \ref stasis_message_router for all \ref stasis messages */
static struct stasis_message_router *stasis_router;
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
/*! \brief The \ref stasis_subscription for forwarding the RTP topic to the AMI topic */
Multiple revisions 399887,400138,400178,400180-400181 ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30 18:55:27 +00:00
static struct stasis_forward *rtp_topic_forwarder;
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
/*! \brief The \ref stasis_subscription for forwarding the Security topic to the AMI topic */
static struct stasis_forward *security_topic_forwarder;
/*!
* \brief Set to true (non-zero) to globally allow all dangerous AMI actions to run
*/
static int live_dangerously;
#ifdef TEST_FRAMEWORK
/*! \brief The \ref stasis_subscription for forwarding the Test topic to the AMI topic */
static struct stasis_forward *test_suite_forwarder;
#endif
#define MGR_SHOW_TERMINAL_WIDTH 80
#define MAX_VARS 128
/*! \brief Fake event class used to end sessions at shutdown */
#define EVENT_FLAG_SHUTDOWN -1
/*! \brief
* Descriptor for a manager session, either on the AMI socket or over HTTP.
*
* \note
* AMI session have managerid == 0; the entry is created upon a connect,
* and destroyed with the socket.
* HTTP sessions have managerid != 0, the value is used as a search key
* to lookup sessions (using the mansession_id cookie, or nonce key from
* Digest Authentication http header).
*/
#define MAX_BLACKLIST_CMD_LEN 2
static const struct {
const char *words[AST_MAX_CMD_LEN];
} command_blacklist[] = {
{{ "module", "load", NULL }},
{{ "module", "unload", NULL }},
{{ "restart", "gracefully", NULL }},
};
Multiple revisions 399887,400138,400178,400180-400181 ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30 18:55:27 +00:00
static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
static void acl_change_stasis_subscribe(void)
{
if (!acl_change_sub) {
acl_change_sub = stasis_subscribe(ast_security_topic(),
acl_change_stasis_cb, NULL);
stasis_subscription_accept_message_type(acl_change_sub, ast_named_acl_change_type());
stasis_subscription_set_filter(acl_change_sub, STASIS_SUBSCRIPTION_FILTER_SELECTIVE);
}
}
static void acl_change_stasis_unsubscribe(void)
{
acl_change_sub = stasis_unsubscribe_and_join(acl_change_sub);
}
/* In order to understand what the heck is going on with the
* mansession_session and mansession structs, we need to have a bit of a history
* lesson.
*
* In the beginning, there was the mansession. The mansession contained data that was
* intrinsic to a manager session, such as the time that it started, the name of the logged-in
* user, etc. In addition to these parameters were the f and fd parameters. For typical manager
* sessions, these were used to represent the TCP socket over which the AMI session was taking
* place. It makes perfect sense for these fields to be a part of the session-specific data since
* the session actually defines this information.
*
* Then came the HTTP AMI sessions. With these, the f and fd fields need to be opened and closed
* for every single action that occurs. Thus the f and fd fields aren't really specific to the session
* but rather to the action that is being executed. Because a single session may execute many commands
* at once, some sort of safety needed to be added in order to be sure that we did not end up with fd
* leaks from one action overwriting the f and fd fields used by a previous action before the previous action
* has had a chance to properly close its handles.
*
* The initial idea to solve this was to use thread synchronization, but this prevented multiple actions
* from being run at the same time in a single session. Some manager actions may block for a long time, thus
* creating a large queue of actions to execute. In addition, this fix did not address the basic architectural
* issue that for HTTP manager sessions, the f and fd variables are not really a part of the session, but are
* part of the action instead.
*
* The new idea was to create a structure on the stack for each HTTP Manager action. This structure would
* contain the action-specific information, such as which file to write to. In order to maintain expectations
* of action handlers and not have to change the public API of the manager code, we would need to name this
* new stacked structure 'mansession' and contain within it the old mansession struct that we used to use.
* We renamed the old mansession struct 'mansession_session' to hopefully convey that what is in this structure
* is session-specific data. The structure that it is wrapped in, called a 'mansession' really contains action-specific
* data.
*/
struct mansession_session {
/*! \todo XXX need to document which fields it is protecting */
struct ast_sockaddr addr; /*!< address we are connecting from */
struct ast_iostream *stream; /*!< AMI stream */
int inuse; /*!< number of HTTP sessions using this entry */
int needdestroy; /*!< Whether an HTTP session should be destroyed */
pthread_t waiting_thread; /*!< Sleeping thread using this descriptor */
uint32_t managerid; /*!< Unique manager identifier, 0 for AMI sessions */
time_t sessionstart; /*!< Session start time */
struct timeval sessionstart_tv; /*!< Session start time */
time_t sessiontimeout; /*!< Session timeout if HTTP */
char username[80]; /*!< Logged in username */
char challenge[10]; /*!< Authentication challenge */
int authenticated; /*!< Authentication status */
int readperm; /*!< Authorization for reading */
int writeperm; /*!< Authorization for writing */
char inbuf[1025]; /*!< Buffer - we use the extra byte to add a '\\0' and simplify parsing */
int inlen; /*!< number of buffered bytes */
struct ao2_container *whitefilters; /*!< Manager event filters - white list */
struct ao2_container *blackfilters; /*!< Manager event filters - black list */
struct ast_variable *chanvars; /*!< Channel variables to set for originate */
int send_events; /*!< XXX what ? */
struct eventqent *last_ev; /*!< last event processed. */
int writetimeout; /*!< Timeout for ast_carefulwrite() */
time_t authstart;
int pending_event; /*!< Pending events indicator in case when waiting_thread is NULL */
time_t noncetime; /*!< Timer for nonce value expiration */
unsigned long oldnonce; /*!< Stale nonce value */
unsigned long nc; /*!< incremental nonce counter */
ast_mutex_t notify_lock; /*!< Lock for notifying this session of events */
AST_LIST_HEAD_NOLOCK(mansession_datastores, ast_datastore) datastores; /*!< Data stores on the session */
AST_LIST_ENTRY(mansession_session) list;
};
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
enum mansession_message_parsing {
MESSAGE_OKAY,
MESSAGE_LINE_TOO_LONG
};
/*! \brief In case you didn't read that giant block of text above the mansession_session struct, the
* \ref mansession is named this solely to keep the API the same in Asterisk. This structure really
* represents data that is different from Manager action to Manager action. The mansession_session pointer
* contained within points to session-specific data.
*/
struct mansession {
struct mansession_session *session;
struct ast_iostream *stream;
struct ast_tcptls_session_instance *tcptls_session;
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
enum mansession_message_parsing parsing;
unsigned int write_error:1;
struct manager_custom_hook *hook;
ast_mutex_t lock;
};
/*! Active manager connection sessions container. */
static AO2_GLOBAL_OBJ_STATIC(mgr_sessions);
/*! \brief user descriptor, as read from the config file.
*
* \note It is still missing some fields -- e.g. we can have multiple permit and deny
* lines which are not supported here, and readperm/writeperm/writetimeout
* are not stored.
*/
struct ast_manager_user {
char username[80];
char *secret; /*!< Secret for logging in */
int readperm; /*!< Authorization for reading */
int writeperm; /*!< Authorization for writing */
int writetimeout; /*!< Per user Timeout for ast_carefulwrite() */
int displayconnects; /*!< XXX unused */
int allowmultiplelogin; /*!< Per user option*/
int keep; /*!< mark entries created on a reload */
struct ao2_container *whitefilters; /*!< Manager event filters - white list */
struct ao2_container *blackfilters; /*!< Manager event filters - black list */
struct ast_acl_list *acl; /*!< ACL setting */
char *a1_hash; /*!< precalculated A1 for Digest auth */
struct ast_variable *chanvars; /*!< Channel variables to set for originate */
AST_RWLIST_ENTRY(ast_manager_user) list;
};
/*! \brief list of users found in the config file */
static AST_RWLIST_HEAD_STATIC(users, ast_manager_user);
/*! \brief list of actions registered */
static AST_RWLIST_HEAD_STATIC(actions, manager_action);
/*! \brief list of hooks registered */
static AST_RWLIST_HEAD_STATIC(manager_hooks, manager_custom_hook);
#ifdef AST_XML_DOCS
/*! \brief A container of event documentation nodes */
static AO2_GLOBAL_OBJ_STATIC(event_docs);
#endif
static int __attribute__((format(printf, 9, 0))) __manager_event_sessions(
struct ao2_container *sessions,
int category,
const char *event,
int chancount,
struct ast_channel **chans,
const char *file,
int line,
const char *func,
const char *fmt,
...);
static enum add_filter_result manager_add_filter(const char *filter_pattern, struct ao2_container *whitefilters, struct ao2_container *blackfilters);
static int match_filter(struct mansession *s, char *eventdata);
/*!
* @{ \brief Define AMI message types.
*/
STASIS_MESSAGE_TYPE_DEFN(ast_manager_get_generic_type);
/*! @} */
/*!
* \internal
* \brief Find a registered action object.
*
* \param name Name of AMI action to find.
*
* \return Reffed action found or NULL
*/
static struct manager_action *action_find(const char *name)
{
struct manager_action *act;
AST_RWLIST_RDLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, act, list) {
if (!strcasecmp(name, act->action)) {
ao2_t_ref(act, +1, "found action object");
break;
}
}
AST_RWLIST_UNLOCK(&actions);
return act;
}
struct stasis_topic *ast_manager_get_topic(void)
{
return manager_topic;
}
struct stasis_message_router *ast_manager_get_message_router(void)
{
return stasis_router;
}
static void manager_json_value_str_append(struct ast_json *value, const char *key,
struct ast_str **res)
{
switch (ast_json_typeof(value)) {
case AST_JSON_STRING:
ast_str_append(res, 0, "%s: %s\r\n", key, ast_json_string_get(value));
break;
case AST_JSON_INTEGER:
ast_str_append(res, 0, "%s: %jd\r\n", key, ast_json_integer_get(value));
break;
case AST_JSON_TRUE:
ast_str_append(res, 0, "%s: True\r\n", key);
break;
case AST_JSON_FALSE:
ast_str_append(res, 0, "%s: False\r\n", key);
break;
default:
ast_str_append(res, 0, "%s: \r\n", key);
break;
}
}
static void manager_json_to_ast_str(struct ast_json *obj, const char *key,
struct ast_str **res, key_exclusion_cb exclusion_cb);
static void manager_json_array_with_key(struct ast_json *obj, const char* key,
size_t index, struct ast_str **res,
key_exclusion_cb exclusion_cb)
{
struct ast_str *key_str = ast_str_alloca(64);
ast_str_set(&key_str, 0, "%s(%zu)", key, index);
manager_json_to_ast_str(obj, ast_str_buffer(key_str),
res, exclusion_cb);
}
static void manager_json_obj_with_key(struct ast_json *obj, const char* key,
const char *parent_key, struct ast_str **res,
key_exclusion_cb exclusion_cb)
{
if (parent_key) {
struct ast_str *key_str = ast_str_alloca(64);
ast_str_set(&key_str, 0, "%s/%s", parent_key, key);
manager_json_to_ast_str(obj, ast_str_buffer(key_str),
res, exclusion_cb);
return;
}
manager_json_to_ast_str(obj, key, res, exclusion_cb);
}
void manager_json_to_ast_str(struct ast_json *obj, const char *key,
struct ast_str **res, key_exclusion_cb exclusion_cb)
{
struct ast_json_iter *i;
/* If obj or res is not given, just return */
if (!obj || !res) {
return;
}
if (!*res && !(*res = ast_str_create(1024))) {
return;
}
if (exclusion_cb && key && exclusion_cb(key)) {
return;
}
if (ast_json_typeof(obj) != AST_JSON_OBJECT &&
ast_json_typeof(obj) != AST_JSON_ARRAY) {
manager_json_value_str_append(obj, key, res);
return;
}
if (ast_json_typeof(obj) == AST_JSON_ARRAY) {
size_t j;
for (j = 0; j < ast_json_array_size(obj); ++j) {
manager_json_array_with_key(ast_json_array_get(obj, j),
key, j, res, exclusion_cb);
}
return;
}
for (i = ast_json_object_iter(obj); i;
i = ast_json_object_iter_next(obj, i)) {
manager_json_obj_with_key(ast_json_object_iter_value(i),
ast_json_object_iter_key(i),
key, res, exclusion_cb);
}
}
struct ast_str *ast_manager_str_from_json_object(struct ast_json *blob, key_exclusion_cb exclusion_cb)
{
struct ast_str *res = ast_str_create(1024);
if (!ast_json_is_null(blob)) {
manager_json_to_ast_str(blob, NULL, &res, exclusion_cb);
}
return res;
}
#define manager_event_sessions(sessions, category, event, contents , ...) \
__manager_event_sessions(sessions, category, event, 0, NULL, __FILE__, __LINE__, __PRETTY_FUNCTION__, contents , ## __VA_ARGS__)
#define any_manager_listeners(sessions) \
((sessions && ao2_container_count(sessions)) || !AST_RWLIST_EMPTY(&manager_hooks))
static void manager_default_msg_cb(void *data, struct stasis_subscription *sub,
struct stasis_message *message)
{
struct ao2_container *sessions;
struct ast_manager_event_blob *ev;
if (!stasis_message_can_be_ami(message)) {
/* Not an AMI message; disregard */
return;
}
sessions = ao2_global_obj_ref(mgr_sessions);
if (!any_manager_listeners(sessions)) {
/* Nobody is listening */
ao2_cleanup(sessions);
return;
}
ev = stasis_message_to_ami(message);
if (!ev) {
/* Conversion failure */
ao2_cleanup(sessions);
return;
}
manager_event_sessions(sessions, ev->event_flags, ev->manager_event,
"%s", ev->extra_fields);
ao2_ref(ev, -1);
ao2_cleanup(sessions);
}
static void manager_generic_msg_cb(void *data, struct stasis_subscription *sub,
struct stasis_message *message)
{
struct ast_json_payload *payload;
int class_type;
const char *type;
struct ast_json *event;
struct ast_str *event_buffer;
struct ao2_container *sessions;
sessions = ao2_global_obj_ref(mgr_sessions);
if (!any_manager_listeners(sessions)) {
/* Nobody is listening */
ao2_cleanup(sessions);
return;
}
payload = stasis_message_data(message);
class_type = ast_json_integer_get(ast_json_object_get(payload->json, "class_type"));
type = ast_json_string_get(ast_json_object_get(payload->json, "type"));
event = ast_json_object_get(payload->json, "event");
event_buffer = ast_manager_str_from_json_object(event, NULL);
if (!event_buffer) {
ast_log(AST_LOG_WARNING, "Error while creating payload for event %s\n", type);
ao2_cleanup(sessions);
return;
}
manager_event_sessions(sessions, class_type, type,
"%s", ast_str_buffer(event_buffer));
ast_free(event_buffer);
ao2_cleanup(sessions);
}
void ast_manager_publish_event(const char *type, int class_type, struct ast_json *obj)
{
RAII_VAR(struct ast_json *, event_info, NULL, ast_json_unref);
RAII_VAR(struct ast_json_payload *, payload, NULL, ao2_cleanup);
RAII_VAR(struct stasis_message *, message, NULL, ao2_cleanup);
if (!obj || !ast_manager_get_generic_type()) {
return;
}
ast_json_ref(obj);
event_info = ast_json_pack("{s: s, s: i, s: o}",
"type", type,
"class_type", class_type,
"event", obj);
if (!event_info) {
return;
}
payload = ast_json_payload_create(event_info);
if (!payload) {
return;
}
message = stasis_message_create(ast_manager_get_generic_type(), payload);
if (!message) {
return;
}
stasis_publish(ast_manager_get_topic(), message);
}
/*! \brief Add a custom hook to be called when an event is fired */
void ast_manager_register_hook(struct manager_custom_hook *hook)
{
AST_RWLIST_WRLOCK(&manager_hooks);
AST_RWLIST_INSERT_TAIL(&manager_hooks, hook, list);
AST_RWLIST_UNLOCK(&manager_hooks);
}
/*! \brief Delete a custom hook to be called when an event is fired */
void ast_manager_unregister_hook(struct manager_custom_hook *hook)
{
AST_RWLIST_WRLOCK(&manager_hooks);
AST_RWLIST_REMOVE(&manager_hooks, hook, list);
AST_RWLIST_UNLOCK(&manager_hooks);
}
int ast_manager_check_enabled(void)
{
return manager_enabled;
}
int ast_webmanager_check_enabled(void)
{
return (webmanager_enabled && manager_enabled);
}
/*!
* Grab a reference to the last event, update usecount as needed.
* Can handle a NULL pointer.
*/
static struct eventqent *grab_last(void)
{
struct eventqent *ret;
AST_RWLIST_WRLOCK(&all_events);
ret = AST_RWLIST_LAST(&all_events);
/* the list is never empty now, but may become so when
* we optimize it in the future, so be prepared.
*/
if (ret) {
ast_atomic_fetchadd_int(&ret->usecount, 1);
}
AST_RWLIST_UNLOCK(&all_events);
return ret;
}
/*!
* Purge unused events. Remove elements from the head
* as long as their usecount is 0 and there is a next element.
*/
static void purge_events(void)
{
struct eventqent *ev;
struct timeval now = ast_tvnow();
AST_RWLIST_WRLOCK(&all_events);
while ( (ev = AST_RWLIST_FIRST(&all_events)) &&
ev->usecount == 0 && AST_RWLIST_NEXT(ev, eq_next)) {
AST_RWLIST_REMOVE_HEAD(&all_events, eq_next);
ast_free(ev);
}
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&all_events, ev, eq_next) {
/* Never release the last event */
if (!AST_RWLIST_NEXT(ev, eq_next)) {
break;
}
/* 2.5 times whatever the HTTP timeout is (maximum 2.5 hours) is the maximum time that we will definitely cache an event */
if (ev->usecount == 0 && ast_tvdiff_sec(now, ev->tv) > (httptimeout > 3600 ? 3600 : httptimeout) * 2.5) {
AST_RWLIST_REMOVE_CURRENT(eq_next);
ast_free(ev);
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&all_events);
}
/*!
* helper functions to convert back and forth between
* string and numeric representation of set of flags
*/
static const struct permalias {
int num;
const char *label;
} perms[] = {
{ EVENT_FLAG_SYSTEM, "system" },
{ EVENT_FLAG_CALL, "call" },
{ EVENT_FLAG_LOG, "log" },
{ EVENT_FLAG_VERBOSE, "verbose" },
{ EVENT_FLAG_COMMAND, "command" },
{ EVENT_FLAG_AGENT, "agent" },
{ EVENT_FLAG_USER, "user" },
{ EVENT_FLAG_CONFIG, "config" },
{ EVENT_FLAG_DTMF, "dtmf" },
{ EVENT_FLAG_REPORTING, "reporting" },
{ EVENT_FLAG_CDR, "cdr" },
{ EVENT_FLAG_DIALPLAN, "dialplan" },
{ EVENT_FLAG_ORIGINATE, "originate" },
{ EVENT_FLAG_AGI, "agi" },
Merge Call completion support into trunk. From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-09 15:31:32 +00:00
{ EVENT_FLAG_CC, "cc" },
{ EVENT_FLAG_AOC, "aoc" },
{ EVENT_FLAG_TEST, "test" },
{ EVENT_FLAG_SECURITY, "security" },
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
{ EVENT_FLAG_MESSAGE, "message" },
{ INT_MAX, "all" },
{ 0, "none" },
};
/*! Maximum string length of the AMI authority permission string buildable from perms[]. */
#define MAX_AUTH_PERM_STRING 150
/*! \brief Checks to see if a string which can be used to evaluate functions should be rejected */
static int function_capable_string_allowed_with_auths(const char *evaluating, int writepermlist)
{
if (!(writepermlist & EVENT_FLAG_SYSTEM)
&& (
strstr(evaluating, "SHELL") || /* NoOp(${SHELL(rm -rf /)}) */
strstr(evaluating, "EVAL") /* NoOp(${EVAL(${some_var_containing_SHELL})}) */
)) {
return 0;
}
return 1;
}
/*! \brief Convert authority code to a list of options for a user. This will only
* display those authority codes that have an explicit match on authority */
static const char *user_authority_to_str(int authority, struct ast_str **res)
{
int i;
char *sep = "";
ast_str_reset(*res);
for (i = 0; i < ARRAY_LEN(perms) - 1; i++) {
if ((authority & perms[i].num) == perms[i].num) {
ast_str_append(res, 0, "%s%s", sep, perms[i].label);
sep = ",";
}
}
if (ast_str_strlen(*res) == 0) {
/* replace empty string with something sensible */
ast_str_append(res, 0, "<none>");
}
return ast_str_buffer(*res);
}
/*! \brief Convert authority code to a list of options. Note that the EVENT_FLAG_ALL
* authority will always be returned. */
static const char *authority_to_str(int authority, struct ast_str **res)
{
int i;
char *sep = "";
ast_str_reset(*res);
if (authority != EVENT_FLAG_SHUTDOWN) {
for (i = 0; i < ARRAY_LEN(perms) - 1; i++) {
if (authority & perms[i].num) {
ast_str_append(res, 0, "%s%s", sep, perms[i].label);
sep = ",";
}
}
}
if (ast_str_strlen(*res) == 0) {
/* replace empty string with something sensible */
ast_str_append(res, 0, "<none>");
}
return ast_str_buffer(*res);
}
/*! Tells you if smallstr exists inside bigstr
which is delim by delim and uses no buf or stringsep
ast_instring("this|that|more","this",'|') == 1;
feel free to move this to app.c -anthm */
static int ast_instring(const char *bigstr, const char *smallstr, const char delim)
{
const char *val = bigstr, *next;
do {
if ((next = strchr(val, delim))) {
if (!strncmp(val, smallstr, (next - val))) {
return 1;
} else {
continue;
}
} else {
return !strcmp(smallstr, val);
}
} while (*(val = (next + 1)));
return 0;
}
static int get_perm(const char *instr)
{
int x = 0, ret = 0;
if (!instr) {
return 0;
}
for (x = 0; x < ARRAY_LEN(perms); x++) {
if (ast_instring(instr, perms[x].label, ',')) {
ret |= perms[x].num;
}
}
return ret;
}
/*!
* A number returns itself, false returns 0, true returns all flags,
* other strings return the flags that are set.
*/
static int strings_to_mask(const char *string)
{
const char *p;
if (ast_strlen_zero(string)) {
return -1;
}
for (p = string; *p; p++) {
if (*p < '0' || *p > '9') {
break;
}
}
if (!*p) { /* all digits */
return atoi(string);
}
if (ast_false(string)) {
return 0;
}
if (ast_true(string)) { /* all permissions */
int x, ret = 0;
for (x = 0; x < ARRAY_LEN(perms); x++) {
ret |= perms[x].num;
}
return ret;
}
return get_perm(string);
}
/*! \brief Unreference manager session object.
If no more references, then go ahead and delete it */
static struct mansession_session *unref_mansession(struct mansession_session *s)
{
int refcount = ao2_ref(s, -1);
if (manager_debug) {
ast_debug(1, "Mansession: %p refcount now %d\n", s, refcount - 1);
}
return NULL;
}
static void event_filter_destructor(void *obj)
{
regex_t *regex_filter = obj;
regfree(regex_filter);
}
static void session_destructor(void *obj)
{
struct mansession_session *session = obj;
struct eventqent *eqe = session->last_ev;
struct ast_datastore *datastore;
/* Get rid of each of the data stores on the session */
while ((datastore = AST_LIST_REMOVE_HEAD(&session->datastores, entry))) {
/* Free the data store */
ast_datastore_free(datastore);
}
if (eqe) {
ast_atomic_fetchadd_int(&eqe->usecount, -1);
}
if (session->chanvars) {
ast_variables_destroy(session->chanvars);
}
if (session->whitefilters) {
ao2_t_ref(session->whitefilters, -1, "decrement ref for white container, should be last one");
}
if (session->blackfilters) {
ao2_t_ref(session->blackfilters, -1, "decrement ref for black container, should be last one");
}
ast_mutex_destroy(&session->notify_lock);
}
/*! \brief Allocate manager session structure and add it to the list of sessions */
static struct mansession_session *build_mansession(const struct ast_sockaddr *addr)
{
struct ao2_container *sessions;
struct mansession_session *newsession;
newsession = ao2_alloc(sizeof(*newsession), session_destructor);
if (!newsession) {
return NULL;
}
newsession->whitefilters = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, NULL);
newsession->blackfilters = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, NULL);
if (!newsession->whitefilters || !newsession->blackfilters) {
ao2_ref(newsession, -1);
return NULL;
}
newsession->waiting_thread = AST_PTHREADT_NULL;
newsession->writetimeout = 100;
newsession->send_events = -1;
ast_sockaddr_copy(&newsession->addr, addr);
ast_mutex_init(&newsession->notify_lock);
sessions = ao2_global_obj_ref(mgr_sessions);
if (sessions) {
ao2_link(sessions, newsession);
ao2_ref(sessions, -1);
}
return newsession;
}
static int mansession_cmp_fn(void *obj, void *arg, int flags)
{
struct mansession_session *s = obj;
char *str = arg;
return !strcasecmp(s->username, str) ? CMP_MATCH : 0;
}
static void session_destroy(struct mansession_session *s)
{
struct ao2_container *sessions;
sessions = ao2_global_obj_ref(mgr_sessions);
if (sessions) {
ao2_unlink(sessions, s);
ao2_ref(sessions, -1);
}
unref_mansession(s);
}
static int check_manager_session_inuse(const char *name)
{
struct ao2_container *sessions;
struct mansession_session *session;
int inuse = 0;
sessions = ao2_global_obj_ref(mgr_sessions);
if (sessions) {
session = ao2_find(sessions, (char *) name, 0);
ao2_ref(sessions, -1);
if (session) {
unref_mansession(session);
inuse = 1;
}
}
return inuse;
}
/*!
* lookup an entry in the list of registered users.
* must be called with the list lock held.
*/
static struct ast_manager_user *get_manager_by_name_locked(const char *name)
{
struct ast_manager_user *user = NULL;
AST_RWLIST_TRAVERSE(&users, user, list) {
if (!strcasecmp(user->username, name)) {
break;
}
}
return user;
}
/*! \brief Get displayconnects config option.
* \param session manager session to get parameter from.
* \return displayconnects config option value.
*/
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
static int manager_displayconnects(struct mansession_session *session)
{
struct ast_manager_user *user = NULL;
int ret = 0;
AST_RWLIST_RDLOCK(&users);
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
if ((user = get_manager_by_name_locked(session->username))) {
ret = user->displayconnects;
}
AST_RWLIST_UNLOCK(&users);
return ret;
}
#ifdef AST_XML_DOCS
static void print_event_instance(struct ast_cli_args *a, struct ast_xml_doc_item *instance);
#endif
static char *handle_showmancmd(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct manager_action *cur;
struct ast_str *authority;
int num;
int l;
const char *auth_str;
#ifdef AST_XML_DOCS
char syntax_title[64], description_title[64], synopsis_title[64], seealso_title[64];
char arguments_title[64], privilege_title[64], final_response_title[64], list_responses_title[64];
#endif
switch (cmd) {
case CLI_INIT:
e->command = "manager show command";
e->usage =
"Usage: manager show command <actionname> [<actionname> [<actionname> [...]]]\n"
" Shows the detailed description for a specific Asterisk manager interface command.\n";
return NULL;
case CLI_GENERATE:
l = strlen(a->word);
AST_RWLIST_RDLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, cur, list) {
if (!strncasecmp(a->word, cur->action, l)) {
if (ast_cli_completion_add(ast_strdup(cur->action))) {
break;
}
}
}
AST_RWLIST_UNLOCK(&actions);
return NULL;
}
if (a->argc < 4) {
return CLI_SHOWUSAGE;
}
authority = ast_str_alloca(MAX_AUTH_PERM_STRING);
#ifdef AST_XML_DOCS
/* setup the titles */
term_color(synopsis_title, "[Synopsis]\n", COLOR_MAGENTA, 0, 40);
term_color(description_title, "[Description]\n", COLOR_MAGENTA, 0, 40);
term_color(syntax_title, "[Syntax]\n", COLOR_MAGENTA, 0, 40);
term_color(seealso_title, "[See Also]\n", COLOR_MAGENTA, 0, 40);
term_color(arguments_title, "[Arguments]\n", COLOR_MAGENTA, 0, 40);
term_color(privilege_title, "[Privilege]\n", COLOR_MAGENTA, 0, 40);
term_color(final_response_title, "[Final Response]\n", COLOR_MAGENTA, 0, 40);
term_color(list_responses_title, "[List Responses]\n", COLOR_MAGENTA, 0, 40);
#endif
AST_RWLIST_RDLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, cur, list) {
for (num = 3; num < a->argc; num++) {
if (!strcasecmp(cur->action, a->argv[num])) {
auth_str = authority_to_str(cur->authority, &authority);
#ifdef AST_XML_DOCS
if (cur->docsrc == AST_XML_DOC) {
char *syntax = ast_xmldoc_printable(S_OR(cur->syntax, "Not available"), 1);
char *synopsis = ast_xmldoc_printable(S_OR(cur->synopsis, "Not available"), 1);
char *description = ast_xmldoc_printable(S_OR(cur->description, "Not available"), 1);
char *arguments = ast_xmldoc_printable(S_OR(cur->arguments, "Not available"), 1);
char *seealso = ast_xmldoc_printable(S_OR(cur->seealso, "Not available"), 1);
char *privilege = ast_xmldoc_printable(S_OR(auth_str, "Not available"), 1);
char *responses = ast_xmldoc_printable("None", 1);
if (!syntax || !synopsis || !description || !arguments
|| !seealso || !privilege || !responses) {
ast_free(syntax);
ast_free(synopsis);
ast_free(description);
ast_free(arguments);
ast_free(seealso);
ast_free(privilege);
ast_free(responses);
ast_cli(a->fd, "Allocation failure.\n");
AST_RWLIST_UNLOCK(&actions);
return CLI_FAILURE;
}
ast_cli(a->fd, "%s%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n%s%s\n\n%s",
syntax_title, syntax,
synopsis_title, synopsis,
description_title, description,
arguments_title, arguments,
seealso_title, seealso,
privilege_title, privilege,
list_responses_title);
if (!cur->list_responses) {
ast_cli(a->fd, "%s\n\n", responses);
} else {
struct ast_xml_doc_item *temp;
for (temp = cur->list_responses; temp; temp = AST_LIST_NEXT(temp, next)) {
ast_cli(a->fd, "Event: %s\n", temp->name);
print_event_instance(a, temp);
}
}
ast_cli(a->fd, "%s", final_response_title);
if (!cur->final_response) {
ast_cli(a->fd, "%s\n\n", responses);
} else {
ast_cli(a->fd, "Event: %s\n", cur->final_response->name);
print_event_instance(a, cur->final_response);
}
ast_free(syntax);
ast_free(synopsis);
ast_free(description);
ast_free(arguments);
ast_free(seealso);
ast_free(privilege);
ast_free(responses);
} else
#endif
{
ast_cli(a->fd, "Action: %s\nSynopsis: %s\nPrivilege: %s\n%s\n",
cur->action, cur->synopsis,
auth_str,
S_OR(cur->description, ""));
}
}
}
}
AST_RWLIST_UNLOCK(&actions);
return CLI_SUCCESS;
}
static char *handle_mandebug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "manager set debug [on|off]";
e->usage = "Usage: manager set debug [on|off]\n Show, enable, disable debugging of the manager code.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc == 3) {
ast_cli(a->fd, "manager debug is %s\n", manager_debug? "on" : "off");
} else if (a->argc == 4) {
if (!strcasecmp(a->argv[3], "on")) {
manager_debug = 1;
} else if (!strcasecmp(a->argv[3], "off")) {
manager_debug = 0;
} else {
return CLI_SHOWUSAGE;
}
}
return CLI_SUCCESS;
}
static char *handle_showmanager(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_manager_user *user = NULL;
int l;
struct ast_str *rauthority = ast_str_alloca(MAX_AUTH_PERM_STRING);
struct ast_str *wauthority = ast_str_alloca(MAX_AUTH_PERM_STRING);
struct ast_variable *v;
switch (cmd) {
case CLI_INIT:
e->command = "manager show user";
e->usage =
" Usage: manager show user <user>\n"
" Display all information related to the manager user specified.\n";
return NULL;
case CLI_GENERATE:
l = strlen(a->word);
if (a->pos != 3) {
return NULL;
}
AST_RWLIST_RDLOCK(&users);
AST_RWLIST_TRAVERSE(&users, user, list) {
if (!strncasecmp(a->word, user->username, l)) {
if (ast_cli_completion_add(ast_strdup(user->username))) {
break;
}
}
}
AST_RWLIST_UNLOCK(&users);
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
AST_RWLIST_RDLOCK(&users);
if (!(user = get_manager_by_name_locked(a->argv[3]))) {
ast_cli(a->fd, "There is no manager called %s\n", a->argv[3]);
AST_RWLIST_UNLOCK(&users);
return CLI_SUCCESS;
}
ast_cli(a->fd, "\n");
ast_cli(a->fd,
" username: %s\n"
" secret: %s\n"
" ACL: %s\n"
" read perm: %s\n"
" write perm: %s\n"
" displayconnects: %s\n"
"allowmultiplelogin: %s\n",
S_OR(user->username, "(N/A)"),
(user->secret ? "<Set>" : "(N/A)"),
((user->acl && !ast_acl_list_is_empty(user->acl)) ? "yes" : "no"),
user_authority_to_str(user->readperm, &rauthority),
user_authority_to_str(user->writeperm, &wauthority),
(user->displayconnects ? "yes" : "no"),
(user->allowmultiplelogin ? "yes" : "no"));
ast_cli(a->fd, " Variables: \n");
for (v = user->chanvars ; v ; v = v->next) {
ast_cli(a->fd, " %s = %s\n", v->name, v->value);
}
if (!ast_acl_list_is_empty(user->acl)) {
ast_acl_output(a->fd, user->acl, NULL);
}
AST_RWLIST_UNLOCK(&users);
return CLI_SUCCESS;
}
static char *handle_showmanagers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ast_manager_user *user = NULL;
int count_amu = 0;
switch (cmd) {
case CLI_INIT:
e->command = "manager show users";
e->usage =
"Usage: manager show users\n"
" Prints a listing of all managers that are currently configured on that\n"
" system.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
AST_RWLIST_RDLOCK(&users);
/* If there are no users, print out something along those lines */
if (AST_RWLIST_EMPTY(&users)) {
ast_cli(a->fd, "There are no manager users.\n");
AST_RWLIST_UNLOCK(&users);
return CLI_SUCCESS;
}
ast_cli(a->fd, "\nusername\n--------\n");
AST_RWLIST_TRAVERSE(&users, user, list) {
ast_cli(a->fd, "%s\n", user->username);
count_amu++;
}
AST_RWLIST_UNLOCK(&users);
ast_cli(a->fd,"-------------------\n"
"%d manager users configured.\n", count_amu);
return CLI_SUCCESS;
}
/*! \brief CLI command manager list commands */
static char *handle_showmancmds(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct manager_action *cur;
int name_len = 1;
int space_remaining;
#define HSMC_FORMAT " %-*.*s %-.*s\n"
switch (cmd) {
case CLI_INIT:
e->command = "manager show commands";
e->usage =
"Usage: manager show commands\n"
" Prints a listing of all the available Asterisk manager interface commands.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
AST_RWLIST_RDLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, cur, list) {
int incoming_len = strlen(cur->action);
if (incoming_len > name_len) {
name_len = incoming_len;
}
}
space_remaining = MGR_SHOW_TERMINAL_WIDTH - name_len - 4;
if (space_remaining < 0) {
space_remaining = 0;
}
ast_cli(a->fd, HSMC_FORMAT, name_len, name_len, "Action", space_remaining, "Synopsis");
ast_cli(a->fd, HSMC_FORMAT, name_len, name_len, "------", space_remaining, "--------");
AST_RWLIST_TRAVERSE(&actions, cur, list) {
ast_cli(a->fd, HSMC_FORMAT, name_len, name_len, cur->action, space_remaining, cur->synopsis);
}
AST_RWLIST_UNLOCK(&actions);
return CLI_SUCCESS;
}
/*! \brief CLI command manager list connected */
static char *handle_showmanconn(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ao2_container *sessions;
struct mansession_session *session;
time_t now = time(NULL);
#define HSMCONN_FORMAT1 " %-15.15s %-55.55s %-10.10s %-10.10s %-8.8s %-8.8s %-10.10s %-10.10s\n"
#define HSMCONN_FORMAT2 " %-15.15s %-55.55s %-10d %-10d %-8d %-8d %-10.10d %-10.10d\n"
int count = 0;
struct ao2_iterator i;
switch (cmd) {
case CLI_INIT:
e->command = "manager show connected";
e->usage =
"Usage: manager show connected\n"
" Prints a listing of the users that are currently connected to the\n"
"Asterisk manager interface.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_cli(a->fd, HSMCONN_FORMAT1, "Username", "IP Address", "Start", "Elapsed", "FileDes", "HttpCnt", "ReadPerms", "WritePerms");
sessions = ao2_global_obj_ref(mgr_sessions);
if (sessions) {
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i))) {
ao2_lock(session);
ast_cli(a->fd, HSMCONN_FORMAT2, session->username,
ast_sockaddr_stringify_addr(&session->addr),
(int) (session->sessionstart),
(int) (now - session->sessionstart),
session->stream ? ast_iostream_get_fd(session->stream) : -1,
session->inuse,
session->readperm,
session->writeperm);
count++;
ao2_unlock(session);
unref_mansession(session);
}
ao2_iterator_destroy(&i);
}
ast_cli(a->fd, "%d users connected.\n", count);
return CLI_SUCCESS;
}
/*! \brief CLI command manager list eventq */
/* Should change to "manager show connected" */
static char *handle_showmaneventq(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct eventqent *s;
switch (cmd) {
case CLI_INIT:
e->command = "manager show eventq";
e->usage =
"Usage: manager show eventq\n"
" Prints a listing of all events pending in the Asterisk manger\n"
"event queue.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
AST_RWLIST_RDLOCK(&all_events);
AST_RWLIST_TRAVERSE(&all_events, s, eq_next) {
ast_cli(a->fd, "Usecount: %d\n", s->usecount);
ast_cli(a->fd, "Category: %d\n", s->category);
ast_cli(a->fd, "Event:\n%s", s->eventdata);
}
AST_RWLIST_UNLOCK(&all_events);
return CLI_SUCCESS;
}
static int reload_module(void);
/*! \brief CLI command manager reload */
static char *handle_manager_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "manager reload";
e->usage =
"Usage: manager reload\n"
" Reloads the manager configuration.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc > 2) {
return CLI_SHOWUSAGE;
}
reload_module();
return CLI_SUCCESS;
}
static struct eventqent *advance_event(struct eventqent *e)
{
struct eventqent *next;
AST_RWLIST_RDLOCK(&all_events);
if ((next = AST_RWLIST_NEXT(e, eq_next))) {
ast_atomic_fetchadd_int(&next->usecount, 1);
ast_atomic_fetchadd_int(&e->usecount, -1);
}
AST_RWLIST_UNLOCK(&all_events);
return next;
}
#define GET_HEADER_FIRST_MATCH 0
#define GET_HEADER_LAST_MATCH 1
#define GET_HEADER_SKIP_EMPTY 2
/*!
* \brief Return a matching header value.
*
* \details
* Generic function to return either the first or the last
* matching header from a list of variables, possibly skipping
* empty strings.
*
* \note At the moment there is only one use of this function in
* this file, so we make it static.
*
* \note Never returns NULL.
*/
static const char *__astman_get_header(const struct message *m, char *var, int mode)
{
int x, l = strlen(var);
const char *result = "";
if (!m) {
return result;
}
for (x = 0; x < m->hdrcount; x++) {
const char *h = m->headers[x];
if (!strncasecmp(var, h, l) && h[l] == ':') {
const char *value = h + l + 1;
value = ast_skip_blanks(value); /* ignore leading spaces in the value */
/* found a potential candidate */
if ((mode & GET_HEADER_SKIP_EMPTY) && ast_strlen_zero(value)) {
continue; /* not interesting */
}
if (mode & GET_HEADER_LAST_MATCH) {
result = value; /* record the last match so far */
} else {
return value;
}
}
}
return result;
}
/*!
* \brief Return the first matching variable from an array.
*
* \note This is the legacy function and is implemented in
* therms of __astman_get_header().
*
* \note Never returns NULL.
*/
const char *astman_get_header(const struct message *m, char *var)
{
return __astman_get_header(m, var, GET_HEADER_FIRST_MATCH);
}
/*!
* \brief Append additional headers into the message structure from params.
*
* \note You likely want to initialize m->hdrcount to 0 before calling this.
*/
static void astman_append_headers(struct message *m, const struct ast_variable *params)
{
const struct ast_variable *v;
for (v = params; v && m->hdrcount < ARRAY_LEN(m->headers); v = v->next) {
if (ast_asprintf((char**)&m->headers[m->hdrcount], "%s: %s", v->name, v->value) > -1) {
++m->hdrcount;
}
}
}
/*!
* \brief Free headers inside message structure, but not the message structure itself.
*/
static void astman_free_headers(struct message *m)
{
while (m->hdrcount) {
--m->hdrcount;
ast_free((void *) m->headers[m->hdrcount]);
m->headers[m->hdrcount] = NULL;
}
}
/*!
* \internal
* \brief Process one "Variable:" header value string.
*
* \param head Current list of AMI variables to get new values added.
* \param hdr_val Header value string to process.
*
* \return New variable list head.
*/
static struct ast_variable *man_do_variable_value(struct ast_variable *head, const char *hdr_val)
{
char *parse;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(vars)[64];
);
hdr_val = ast_skip_blanks(hdr_val); /* ignore leading spaces in the value */
parse = ast_strdupa(hdr_val);
/* Break the header value string into name=val pair items. */
AST_STANDARD_APP_ARGS(args, parse);
if (args.argc) {
int y;
/* Process each name=val pair item. */
for (y = 0; y < args.argc; y++) {
struct ast_variable *cur;
char *var;
char *val;
if (!args.vars[y]) {
continue;
}
var = val = args.vars[y];
strsep(&val, "=");
/* XXX We may wish to trim whitespace from the strings. */
if (!val || ast_strlen_zero(var)) {
continue;
}
/* Create new variable list node and prepend it to the list. */
cur = ast_variable_new(var, val, "");
if (cur) {
cur->next = head;
head = cur;
}
}
}
return head;
}
struct ast_variable *astman_get_variables(const struct message *m)
{
return astman_get_variables_order(m, ORDER_REVERSE);
}
struct ast_variable *astman_get_variables_order(const struct message *m,
enum variable_orders order)
{
int varlen;
int x;
struct ast_variable *head = NULL;
static const char var_hdr[] = "Variable:";
/* Process all "Variable:" headers. */
varlen = strlen(var_hdr);
for (x = 0; x < m->hdrcount; x++) {
if (strncasecmp(var_hdr, m->headers[x], varlen)) {
continue;
}
head = man_do_variable_value(head, m->headers[x] + varlen);
}
if (order == ORDER_NATURAL) {
head = ast_variables_reverse(head);
}
return head;
}
/*! \brief access for hooks to send action messages to ami */
int ast_hook_send_action(struct manager_custom_hook *hook, const char *msg)
{
const char *action;
int ret = 0;
struct manager_action *act_found;
struct mansession s = {.session = NULL, };
struct message m = { 0 };
char *dup_str;
char *src;
int x = 0;
int curlen;
if (hook == NULL) {
return -1;
}
/* Create our own copy of the AMI action msg string. */
src = dup_str = ast_strdup(msg);
if (!dup_str) {
return -1;
}
/* convert msg string to message struct */
curlen = strlen(src);
for (x = 0; x < curlen; x++) {
int cr; /* set if we have \r */
if (src[x] == '\r' && x+1 < curlen && src[x+1] == '\n')
cr = 2; /* Found. Update length to include \r\n */
else if (src[x] == '\n')
cr = 1; /* also accept \n only */
else
continue;
/* don't keep empty lines */
if (x && m.hdrcount < ARRAY_LEN(m.headers)) {
/* ... but trim \r\n and terminate the header string */
src[x] = '\0';
m.headers[m.hdrcount++] = src;
}
x += cr;
curlen -= x; /* remaining size */
src += x; /* update pointer */
x = -1; /* reset loop */
}
action = astman_get_header(&m, "Action");
do {
if (!strcasecmp(action, "login")) {
break;
}
act_found = action_find(action);
if (!act_found) {
break;
}
/*
* we have to simulate a session for this action request
* to be able to pass it down for processing
* This is necessary to meet the previous design of manager.c
*/
s.hook = hook;
ret = -1;
ao2_lock(act_found);
if (act_found->registered && act_found->func) {
struct ast_module *mod_ref = ast_module_running_ref(act_found->module);
ao2_unlock(act_found);
/* If the action is in a module it must be running. */
if (!act_found->module || mod_ref) {
ret = act_found->func(&s, &m);
ast_module_unref(mod_ref);
}
} else {
ao2_unlock(act_found);
}
ao2_t_ref(act_found, -1, "done with found action object");
} while (0);
ast_free(dup_str);
return ret;
}
/*!
* helper function to send a string to the socket.
* Return -1 on error (e.g. buffer full).
*/
static int send_string(struct mansession *s, char *string)
{
struct ast_iostream *stream;
int len, res;
/* It's a result from one of the hook's action invocation */
if (s->hook) {
/*
* to send responses, we're using the same function
* as for receiving events. We call the event "HookResponse"
*/
s->hook->helper(EVENT_FLAG_HOOKRESPONSE, "HookResponse", string);
return 0;
}
stream = s->stream ? s->stream : s->session->stream;
len = strlen(string);
ast_iostream_set_timeout_inactivity(stream, s->session->writetimeout);
res = ast_iostream_write(stream, string, len);
ast_iostream_set_timeout_disable(stream);
if (res < len) {
s->write_error = 1;
}
return res;
}
/*!
* \brief thread local buffer for astman_append
*
* \note This can not be defined within the astman_append() function
* because it declares a couple of functions that get used to
* initialize the thread local storage key.
*/
AST_THREADSTORAGE(astman_append_buf);
AST_THREADSTORAGE(userevent_buf);
/*! \brief initial allocated size for the astman_append_buf and astman_send_*_va */
#define ASTMAN_APPEND_BUF_INITSIZE 256
static void astman_flush(struct mansession *s, struct ast_str *buf)
{
if (s->hook || (s->tcptls_session && s->tcptls_session->stream)) {
send_string(s, ast_str_buffer(buf));
} else {
ast_verbose("No connection stream in astman_append, should not happen\n");
}
}
/*!
* utility functions for creating AMI replies
*/
void astman_append(struct mansession *s, const char *fmt, ...)
{
int res;
va_list ap;
struct ast_str *buf;
if (!(buf = ast_str_thread_get(&astman_append_buf, ASTMAN_APPEND_BUF_INITSIZE))) {
return;
}
va_start(ap, fmt);
res = ast_str_set_va(&buf, 0, fmt, ap);
va_end(ap);
if (res == AST_DYNSTR_BUILD_FAILED) {
return;
}
if (s->hook || (s->tcptls_session != NULL && s->tcptls_session->stream != NULL)) {
send_string(s, ast_str_buffer(buf));
} else {
ast_verbose("No connection stream in astman_append, should not happen\n");
}
}
/*! \note NOTE: XXX this comment is unclear and possibly wrong.
Callers of astman_send_error(), astman_send_response() or astman_send_ack() must EITHER
hold the session lock _or_ be running in an action callback (in which case s->session->busy will
be non-zero). In either of these cases, there is no need to lock-protect the session's
fd, since no other output will be sent (events will be queued), and no input will
be read until either the current action finishes or get_input() obtains the session
lock.
*/
/*! \todo XXX MSG_MOREDATA should go to a header file. */
#define MSG_MOREDATA ((char *)astman_send_response)
/*! \brief send a response with an optional message,
* and terminate it with an empty line.
* m is used only to grab the 'ActionID' field.
*
* Use the explicit constant MSG_MOREDATA to remove the empty line.
* XXX MSG_MOREDATA should go to a header file.
*/
static void astman_send_response_full(struct mansession *s, const struct message *m, char *resp, char *msg, char *listflag)
{
const char *id = astman_get_header(m, "ActionID");
struct ast_str *buf;
buf = ast_str_thread_get(&astman_append_buf, ASTMAN_APPEND_BUF_INITSIZE);
if (!buf) {
return;
}
ast_str_set(&buf, 0, "Response: %s\r\n", resp);
if (!ast_strlen_zero(id)) {
ast_str_append(&buf, 0, "ActionID: %s\r\n", id);
}
if (listflag) {
/* Start, complete, cancelled */
ast_str_append(&buf, 0, "EventList: %s\r\n", listflag);
}
if (msg != MSG_MOREDATA) {
if (msg) {
ast_str_append(&buf, 0, "Message: %s\r\n", msg);
}
ast_str_append(&buf, 0, "\r\n");
}
astman_flush(s, buf);
}
void astman_send_response(struct mansession *s, const struct message *m, char *resp, char *msg)
{
astman_send_response_full(s, m, resp, msg, NULL);
}
void astman_send_error(struct mansession *s, const struct message *m, char *error)
{
astman_send_response_full(s, m, "Error", error, NULL);
}
void astman_send_error_va(struct mansession *s, const struct message *m, const char *fmt, ...)
{
int res;
va_list ap;
struct ast_str *buf;
char *msg;
if (!(buf = ast_str_thread_get(&astman_append_buf, ASTMAN_APPEND_BUF_INITSIZE))) {
return;
}
va_start(ap, fmt);
res = ast_str_set_va(&buf, 0, fmt, ap);
va_end(ap);
if (res == AST_DYNSTR_BUILD_FAILED) {
return;
}
/* astman_append will use the same underlying buffer, so copy the message out
* before sending the response */
msg = ast_str_buffer(buf);
if (msg) {
msg = ast_strdupa(msg);
}
astman_send_response_full(s, m, "Error", msg, NULL);
}
void astman_send_ack(struct mansession *s, const struct message *m, char *msg)
{
astman_send_response_full(s, m, "Success", msg, NULL);
}
static void astman_start_ack(struct mansession *s, const struct message *m)
{
astman_send_response_full(s, m, "Success", MSG_MOREDATA, NULL);
}
void astman_send_listack(struct mansession *s, const struct message *m, char *msg, char *listflag)
{
astman_send_response_full(s, m, "Success", msg, listflag);
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
}
static struct ast_str *astman_send_list_complete_start_common(struct mansession *s, const struct message *m, const char *event_name, int count)
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
{
const char *id = astman_get_header(m, "ActionID");
struct ast_str *buf;
buf = ast_str_thread_get(&astman_append_buf, ASTMAN_APPEND_BUF_INITSIZE);
if (!buf) {
return NULL;
}
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
ast_str_set(&buf, 0, "Event: %s\r\n", event_name);
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
if (!ast_strlen_zero(id)) {
ast_str_append(&buf, 0, "ActionID: %s\r\n", id);
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
}
ast_str_append(&buf, 0,
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
"EventList: Complete\r\n"
"ListItems: %d\r\n",
count);
return buf;
}
static void astman_send_list_complete(struct mansession *s, const struct message *m, const char *event_name, int count)
{
struct ast_str *buf = astman_send_list_complete_start_common(s, m, event_name, count);
if (buf) {
ast_str_append(&buf, 0, "\r\n");
astman_flush(s, buf);
}
}
void astman_send_list_complete_start(struct mansession *s, const struct message *m, const char *event_name, int count)
{
struct ast_str *buf = astman_send_list_complete_start_common(s, m, event_name, count);
if (buf) {
astman_flush(s, buf);
}
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
}
void astman_send_list_complete_end(struct mansession *s)
{
astman_append(s, "\r\n");
}
/*! \brief Lock the 'mansession' structure. */
static void mansession_lock(struct mansession *s)
{
ast_mutex_lock(&s->lock);
}
/*! \brief Unlock the 'mansession' structure. */
static void mansession_unlock(struct mansession *s)
{
ast_mutex_unlock(&s->lock);
}
/*! \brief
Rather than braindead on,off this now can also accept a specific int mask value
or a ',' delim list of mask strings (the same as manager.conf) -anthm
*/
static int set_eventmask(struct mansession *s, const char *eventmask)
{
int maskint = strings_to_mask(eventmask);
ao2_lock(s->session);
if (maskint >= 0) {
s->session->send_events = maskint;
}
ao2_unlock(s->session);
return maskint;
}
static enum ast_transport mansession_get_transport(const struct mansession *s)
{
return s->tcptls_session->parent->tls_cfg ? AST_TRANSPORT_TLS :
AST_TRANSPORT_TCP;
}
static void report_invalid_user(const struct mansession *s, const char *username)
{
char session_id[32];
struct ast_security_event_inval_acct_id inval_acct_id = {
.common.event_type = AST_SECURITY_EVENT_INVAL_ACCT_ID,
.common.version = AST_SECURITY_EVENT_INVAL_ACCT_ID_VERSION,
.common.service = "AMI",
.common.account_id = username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", s);
ast_security_event_report(AST_SEC_EVT(&inval_acct_id));
}
static void report_failed_acl(const struct mansession *s, const char *username)
{
char session_id[32];
struct ast_security_event_failed_acl failed_acl_event = {
.common.event_type = AST_SECURITY_EVENT_FAILED_ACL,
.common.version = AST_SECURITY_EVENT_FAILED_ACL_VERSION,
.common.service = "AMI",
.common.account_id = username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
ast_security_event_report(AST_SEC_EVT(&failed_acl_event));
}
static void report_inval_password(const struct mansession *s, const char *username)
{
char session_id[32];
struct ast_security_event_inval_password inval_password = {
.common.event_type = AST_SECURITY_EVENT_INVAL_PASSWORD,
.common.version = AST_SECURITY_EVENT_INVAL_PASSWORD_VERSION,
.common.service = "AMI",
.common.account_id = username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
ast_security_event_report(AST_SEC_EVT(&inval_password));
}
static void report_auth_success(const struct mansession *s)
{
char session_id[32];
struct ast_security_event_successful_auth successful_auth = {
.common.event_type = AST_SECURITY_EVENT_SUCCESSFUL_AUTH,
.common.version = AST_SECURITY_EVENT_SUCCESSFUL_AUTH_VERSION,
.common.service = "AMI",
.common.account_id = s->session->username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
ast_security_event_report(AST_SEC_EVT(&successful_auth));
}
static void report_req_not_allowed(const struct mansession *s, const char *action)
{
char session_id[32];
char request_type[64];
struct ast_security_event_req_not_allowed req_not_allowed = {
.common.event_type = AST_SECURITY_EVENT_REQ_NOT_ALLOWED,
.common.version = AST_SECURITY_EVENT_REQ_NOT_ALLOWED_VERSION,
.common.service = "AMI",
.common.account_id = s->session->username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
.request_type = request_type,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
snprintf(request_type, sizeof(request_type), "Action: %s", action);
ast_security_event_report(AST_SEC_EVT(&req_not_allowed));
}
static void report_req_bad_format(const struct mansession *s, const char *action)
{
char session_id[32];
char request_type[64];
struct ast_security_event_req_bad_format req_bad_format = {
.common.event_type = AST_SECURITY_EVENT_REQ_BAD_FORMAT,
.common.version = AST_SECURITY_EVENT_REQ_BAD_FORMAT_VERSION,
.common.service = "AMI",
.common.account_id = s->session->username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
.request_type = request_type,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
snprintf(request_type, sizeof(request_type), "Action: %s", action);
ast_security_event_report(AST_SEC_EVT(&req_bad_format));
}
static void report_failed_challenge_response(const struct mansession *s,
const char *response, const char *expected_response)
{
char session_id[32];
struct ast_security_event_chal_resp_failed chal_resp_failed = {
.common.event_type = AST_SECURITY_EVENT_CHAL_RESP_FAILED,
.common.version = AST_SECURITY_EVENT_CHAL_RESP_FAILED_VERSION,
.common.service = "AMI",
.common.account_id = s->session->username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
.challenge = s->session->challenge,
.response = response,
.expected_response = expected_response,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
ast_security_event_report(AST_SEC_EVT(&chal_resp_failed));
}
static void report_session_limit(const struct mansession *s)
{
char session_id[32];
struct ast_security_event_session_limit session_limit = {
.common.event_type = AST_SECURITY_EVENT_SESSION_LIMIT,
.common.version = AST_SECURITY_EVENT_SESSION_LIMIT_VERSION,
.common.service = "AMI",
.common.account_id = s->session->username,
.common.session_tv = &s->session->sessionstart_tv,
.common.local_addr = {
.addr = &s->tcptls_session->parent->local_address,
.transport = mansession_get_transport(s),
},
.common.remote_addr = {
.addr = &s->session->addr,
.transport = mansession_get_transport(s),
},
.common.session_id = session_id,
};
snprintf(session_id, sizeof(session_id), "%p", s->session);
ast_security_event_report(AST_SEC_EVT(&session_limit));
}
/*
* Here we start with action_ handlers for AMI actions,
* and the internal functions used by them.
* Generally, the handlers are called action_foo()
*/
/* helper function for action_login() */
static int authenticate(struct mansession *s, const struct message *m)
{
const char *username = astman_get_header(m, "Username");
const char *password = astman_get_header(m, "Secret");
int error = -1;
struct ast_manager_user *user = NULL;
regex_t *regex_filter;
struct ao2_iterator filter_iter;
if (ast_strlen_zero(username)) { /* missing username */
return -1;
}
/* locate user in locked state */
AST_RWLIST_WRLOCK(&users);
if (!(user = get_manager_by_name_locked(username))) {
report_invalid_user(s, username);
ast_log(LOG_NOTICE, "%s tried to authenticate with nonexistent user '%s'\n", ast_sockaddr_stringify_addr(&s->session->addr), username);
} else if (user->acl && (ast_apply_acl(user->acl, &s->session->addr, "Manager User ACL: ") == AST_SENSE_DENY)) {
report_failed_acl(s, username);
ast_log(LOG_NOTICE, "%s failed to pass IP ACL as '%s'\n", ast_sockaddr_stringify_addr(&s->session->addr), username);
} else if (!strcasecmp(astman_get_header(m, "AuthType"), "MD5")) {
const char *key = astman_get_header(m, "Key");
if (!ast_strlen_zero(key) && !ast_strlen_zero(s->session->challenge) && user->secret) {
int x;
int len = 0;
char md5key[256] = "";
struct MD5Context md5;
unsigned char digest[16];
MD5Init(&md5);
MD5Update(&md5, (unsigned char *) s->session->challenge, strlen(s->session->challenge));
MD5Update(&md5, (unsigned char *) user->secret, strlen(user->secret));
MD5Final(digest, &md5);
for (x = 0; x < 16; x++)
len += sprintf(md5key + len, "%02hhx", digest[x]);
if (!strcmp(md5key, key)) {
error = 0;
} else {
report_failed_challenge_response(s, key, md5key);
}
} else {
ast_debug(1, "MD5 authentication is not possible. challenge: '%s'\n",
S_OR(s->session->challenge, ""));
}
} else if (user->secret) {
if (!strcmp(password, user->secret)) {
error = 0;
} else {
report_inval_password(s, username);
}
}
if (error) {
ast_log(LOG_NOTICE, "%s failed to authenticate as '%s'\n", ast_sockaddr_stringify_addr(&s->session->addr), username);
AST_RWLIST_UNLOCK(&users);
return -1;
}
/* auth complete */
/* All of the user parameters are copied to the session so that in the event
* of a reload and a configuration change, the session parameters are not
* changed. */
ast_copy_string(s->session->username, username, sizeof(s->session->username));
s->session->readperm = user->readperm;
s->session->writeperm = user->writeperm;
s->session->writetimeout = user->writetimeout;
if (user->chanvars) {
s->session->chanvars = ast_variables_dup(user->chanvars);
}
filter_iter = ao2_iterator_init(user->whitefilters, 0);
while ((regex_filter = ao2_iterator_next(&filter_iter))) {
ao2_t_link(s->session->whitefilters, regex_filter, "add white user filter to session");
ao2_t_ref(regex_filter, -1, "remove iterator ref");
}
ao2_iterator_destroy(&filter_iter);
filter_iter = ao2_iterator_init(user->blackfilters, 0);
while ((regex_filter = ao2_iterator_next(&filter_iter))) {
ao2_t_link(s->session->blackfilters, regex_filter, "add black user filter to session");
ao2_t_ref(regex_filter, -1, "remove iterator ref");
}
ao2_iterator_destroy(&filter_iter);
s->session->sessionstart = time(NULL);
s->session->sessionstart_tv = ast_tvnow();
set_eventmask(s, astman_get_header(m, "Events"));
report_auth_success(s);
AST_RWLIST_UNLOCK(&users);
return 0;
}
static int action_ping(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
struct timeval now = ast_tvnow();
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(actionid)){
astman_append(s, "ActionID: %s\r\n", actionid);
}
astman_append(
s,
"Ping: Pong\r\n"
"Timestamp: %ld.%06lu\r\n"
"\r\n",
(long) now.tv_sec, (unsigned long) now.tv_usec);
return 0;
}
void astman_live_dangerously(int new_live_dangerously)
{
if (new_live_dangerously && !live_dangerously)
{
ast_log(LOG_WARNING, "Manager Configuration load protection disabled.\n");
}
if (!new_live_dangerously && live_dangerously)
{
ast_log(LOG_NOTICE, "Manager Configuration load protection enabled.\n");
}
live_dangerously = new_live_dangerously;
}
/**
* \brief Check if a file is restricted or not
*
* \return 0 on success
* \return 1 on restricted file
* \return -1 on failure
*/
static int restrictedFile(const char *filename)
{
char *stripped_filename;
RAII_VAR(char *, path, NULL, ast_free);
RAII_VAR(char *, real_path, NULL, ast_std_free);
if (live_dangerously) {
return 0;
}
stripped_filename = ast_strip(ast_strdupa(filename));
/* If the file path starts with '/', don't prepend ast_config_AST_CONFIG_DIR */
if (stripped_filename[0] == '/') {
real_path = realpath(stripped_filename, NULL);
} else {
if (ast_asprintf(&path, "%s/%s", ast_config_AST_CONFIG_DIR, stripped_filename) == -1) {
return -1;
}
real_path = realpath(path, NULL);
}
if (!real_path) {
return -1;
}
if (!ast_begins_with(real_path, ast_config_AST_CONFIG_DIR)) {
return 1;
}
return 0;
}
static int action_getconfig(struct mansession *s, const struct message *m)
{
struct ast_config *cfg;
const char *fn = astman_get_header(m, "Filename");
const char *category = astman_get_header(m, "Category");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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const char *filter = astman_get_header(m, "Filter");
const char *category_name;
int catcount = 0;
int lineno = 0;
int ret = 0;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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struct ast_category *cur_category = NULL;
struct ast_variable *v;
struct ast_flags config_flags = { CONFIG_FLAG_WITHCOMMENTS | CONFIG_FLAG_NOCACHE };
if (ast_strlen_zero(fn)) {
astman_send_error(s, m, "Filename not specified");
return 0;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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ret = restrictedFile(fn);
if (ret == 1) {
astman_send_error(s, m, "File requires escalated priveledges");
return 0;
} else if (ret == -1) {
astman_send_error(s, m, "Config file not found");
return 0;
}
cfg = ast_config_load2(fn, "manager", config_flags);
if (cfg == CONFIG_STATUS_FILEMISSING) {
astman_send_error(s, m, "Config file not found");
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
astman_send_error(s, m, "Config file has invalid format");
return 0;
}
astman_start_ack(s, m);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
while ((cur_category = ast_category_browse_filtered(cfg, category, cur_category, filter))) {
struct ast_str *templates;
category_name = ast_category_get_name(cur_category);
lineno = 0;
astman_append(s, "Category-%06d: %s\r\n", catcount, category_name);
if (ast_category_is_template(cur_category)) {
astman_append(s, "IsTemplate-%06d: %d\r\n", catcount, 1);
}
if ((templates = ast_category_get_templates(cur_category))
&& ast_str_strlen(templates) > 0) {
astman_append(s, "Templates-%06d: %s\r\n", catcount, ast_str_buffer(templates));
ast_free(templates);
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
for (v = ast_category_first(cur_category); v; v = v->next) {
astman_append(s, "Line-%06d-%06d: %s=%s\r\n", catcount, lineno++, v->name, v->value);
}
catcount++;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!ast_strlen_zero(category) && catcount == 0) { /* TODO: actually, a config with no categories doesn't even get loaded */
astman_append(s, "No categories found\r\n");
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
ast_config_destroy(cfg);
astman_append(s, "\r\n");
return 0;
}
static int action_listcategories(struct mansession *s, const struct message *m)
{
struct ast_config *cfg;
const char *fn = astman_get_header(m, "Filename");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
const char *match = astman_get_header(m, "Match");
struct ast_category *category = NULL;
struct ast_flags config_flags = { CONFIG_FLAG_WITHCOMMENTS | CONFIG_FLAG_NOCACHE };
int catcount = 0;
if (ast_strlen_zero(fn)) {
astman_send_error(s, m, "Filename not specified");
return 0;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!(cfg = ast_config_load2(fn, "manager", config_flags))) {
astman_send_error(s, m, "Config file not found");
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
astman_send_error(s, m, "Config file has invalid format");
return 0;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
astman_start_ack(s, m);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
while ((category = ast_category_browse_filtered(cfg, NULL, category, match))) {
astman_append(s, "Category-%06d: %s\r\n", catcount, ast_category_get_name(category));
catcount++;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (catcount == 0) { /* TODO: actually, a config with no categories doesn't even get loaded */
astman_append(s, "Error: no categories found\r\n");
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
ast_config_destroy(cfg);
astman_append(s, "\r\n");
return 0;
}
/*! The amount of space in out must be at least ( 2 * strlen(in) + 1 ) */
static void json_escape(char *out, const char *in)
{
for (; *in; in++) {
if (*in == '\\' || *in == '\"') {
*out++ = '\\';
}
*out++ = *in;
}
*out = '\0';
}
/*!
* \internal
* \brief Append a JSON escaped string to the manager stream.
*
* \param s AMI stream to append a string.
* \param str String to append to the stream after JSON escaping it.
*/
static void astman_append_json(struct mansession *s, const char *str)
{
char *buf;
buf = ast_alloca(2 * strlen(str) + 1);
json_escape(buf, str);
astman_append(s, "%s", buf);
}
static int action_getconfigjson(struct mansession *s, const struct message *m)
{
struct ast_config *cfg;
const char *fn = astman_get_header(m, "Filename");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
const char *filter = astman_get_header(m, "Filter");
const char *category = astman_get_header(m, "Category");
struct ast_category *cur_category = NULL;
const char *category_name;
struct ast_variable *v;
int comma1 = 0;
struct ast_flags config_flags = { CONFIG_FLAG_WITHCOMMENTS | CONFIG_FLAG_NOCACHE };
if (ast_strlen_zero(fn)) {
astman_send_error(s, m, "Filename not specified");
return 0;
}
if (restrictedFile(fn)) {
astman_send_error(s, m, "File requires escalated priveledges");
return 0;
}
if (!(cfg = ast_config_load2(fn, "manager", config_flags))) {
astman_send_error(s, m, "Config file not found");
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
astman_send_error(s, m, "Config file has invalid format");
return 0;
}
astman_start_ack(s, m);
astman_append(s, "JSON: {");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
while ((cur_category = ast_category_browse_filtered(cfg, category, cur_category, filter))) {
int comma2 = 0;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
struct ast_str *templates;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
category_name = ast_category_get_name(cur_category);
astman_append(s, "%s\"", comma1 ? "," : "");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
astman_append_json(s, category_name);
astman_append(s, "\":{");
comma1 = 1;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (ast_category_is_template(cur_category)) {
astman_append(s, "\"istemplate\":1");
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
comma2 = 1;
}
if ((templates = ast_category_get_templates(cur_category))
&& ast_str_strlen(templates) > 0) {
astman_append(s, "%s", comma2 ? "," : "");
astman_append(s, "\"templates\":\"%s\"", ast_str_buffer(templates));
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
ast_free(templates);
comma2 = 1;
}
for (v = ast_category_first(cur_category); v; v = v->next) {
astman_append(s, "%s\"", comma2 ? "," : "");
astman_append_json(s, v->name);
astman_append(s, "\":\"");
astman_append_json(s, v->value);
astman_append(s, "\"");
comma2 = 1;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
astman_append(s, "}");
}
astman_append(s, "}\r\n\r\n");
ast_config_destroy(cfg);
return 0;
}
/*! \brief helper function for action_updateconfig */
static enum error_type handle_updates(struct mansession *s, const struct message *m, struct ast_config *cfg, const char *dfn)
{
int x;
char hdr[40];
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
const char *action, *cat, *var, *value, *match, *line, *options;
struct ast_variable *v;
struct ast_str *str1 = ast_str_create(16), *str2 = ast_str_create(16);
enum error_type result = 0;
for (x = 0; x < 100000; x++) { /* 100000 = the max number of allowed updates + 1 */
unsigned int object = 0;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
char *dupoptions;
int allowdups = 0;
int istemplate = 0;
int ignoreerror = 0;
RAII_VAR(char *, inherit, NULL, ast_free);
RAII_VAR(char *, catfilter, NULL, ast_free);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
char *token;
int foundvar = 0;
int foundcat = 0;
struct ast_category *category = NULL;
snprintf(hdr, sizeof(hdr), "Action-%06d", x);
action = astman_get_header(m, hdr);
if (ast_strlen_zero(action)) /* breaks the for loop if no action header */
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
break; /* this could cause problems if actions come in misnumbered */
snprintf(hdr, sizeof(hdr), "Cat-%06d", x);
cat = astman_get_header(m, hdr);
if (ast_strlen_zero(cat)) { /* every action needs a category */
result = UNSPECIFIED_CATEGORY;
break;
}
snprintf(hdr, sizeof(hdr), "Var-%06d", x);
var = astman_get_header(m, hdr);
snprintf(hdr, sizeof(hdr), "Value-%06d", x);
value = astman_get_header(m, hdr);
if (!ast_strlen_zero(value) && *value == '>') {
object = 1;
value++;
}
snprintf(hdr, sizeof(hdr), "Match-%06d", x);
match = astman_get_header(m, hdr);
snprintf(hdr, sizeof(hdr), "Line-%06d", x);
line = astman_get_header(m, hdr);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
snprintf(hdr, sizeof(hdr), "Options-%06d", x);
options = astman_get_header(m, hdr);
if (!ast_strlen_zero(options)) {
char copy[strlen(options) + 1];
strcpy(copy, options); /* safe */
dupoptions = copy;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
while ((token = ast_strsep(&dupoptions, ',', AST_STRSEP_STRIP))) {
if (!strcasecmp("allowdups", token)) {
allowdups = 1;
continue;
}
if (!strcasecmp("template", token)) {
istemplate = 1;
continue;
}
if (!strcasecmp("ignoreerror", token)) {
ignoreerror = 1;
continue;
}
if (ast_begins_with(token, "inherit")) {
char *c = ast_strsep(&token, '=', AST_STRSEP_STRIP);
c = ast_strsep(&token, '=', AST_STRSEP_STRIP);
if (c) {
inherit = ast_strdup(c);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
}
continue;
}
if (ast_begins_with(token, "catfilter")) {
char *c = ast_strsep(&token, '=', AST_STRSEP_STRIP);
c = ast_strsep(&token, '=', AST_STRSEP_STRIP);
if (c) {
catfilter = ast_strdup(c);
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
}
continue;
}
}
}
if (!strcasecmp(action, "newcat")) {
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
struct ast_category *template;
char *tmpl_name = NULL;
if (!allowdups) {
if (ast_category_get(cfg, cat, "TEMPLATES=include")) {
if (ignoreerror) {
continue;
} else {
result = FAILURE_NEWCAT; /* already exist */
break;
}
}
}
if (istemplate) {
category = ast_category_new_template(cat, dfn, -1);
} else {
category = ast_category_new(cat, dfn, -1);
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!category) {
result = FAILURE_ALLOCATION;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (inherit) {
while ((tmpl_name = ast_strsep(&inherit, ',', AST_STRSEP_STRIP))) {
if ((template = ast_category_get(cfg, tmpl_name, "TEMPLATES=restrict"))) {
if (ast_category_inherit(category, template)) {
result = FAILURE_ALLOCATION;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
} else {
ast_category_destroy(category);
category = NULL;
result = FAILURE_TEMPLATE; /* template not found */
break;
}
}
}
if (category != NULL) {
if (ast_strlen_zero(match)) {
ast_category_append(cfg, category);
} else {
if (ast_category_insert(cfg, category, match)) {
ast_category_destroy(category);
result = FAILURE_NEWCAT;
break;
}
}
}
} else if (!strcasecmp(action, "renamecat")) {
if (ast_strlen_zero(value)) {
result = UNSPECIFIED_ARGUMENT;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
ast_category_rename(category, value);
foundcat = 1;
}
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
} else if (!strcasecmp(action, "delcat")) {
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
category = ast_category_delete(cfg, category);
foundcat = 1;
}
if (!foundcat && !ignoreerror) {
result = UNKNOWN_CATEGORY;
break;
}
} else if (!strcasecmp(action, "emptycat")) {
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
ast_category_empty(category);
foundcat = 1;
}
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
} else if (!strcasecmp(action, "update")) {
if (ast_strlen_zero(var)) {
result = UNSPECIFIED_ARGUMENT;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
foundvar = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
if (!ast_variable_update(category, var, value, match, object)) {
foundvar = 1;
}
foundcat = 1;
}
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!foundvar) {
result = FAILURE_UPDATE;
break;
}
} else if (!strcasecmp(action, "delete")) {
if ((ast_strlen_zero(var) && ast_strlen_zero(line))) {
result = UNSPECIFIED_ARGUMENT;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
foundvar = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
if (!ast_variable_delete(category, var, match, line)) {
foundvar = 1;
}
foundcat = 1;
}
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!foundvar && !ignoreerror) {
result = FAILURE_UPDATE;
break;
}
} else if (!strcasecmp(action, "append")) {
if (ast_strlen_zero(var)) {
result = UNSPECIFIED_ARGUMENT;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
if (!(v = ast_variable_new(var, value, dfn))) {
result = FAILURE_ALLOCATION;
break;
}
if (object || (match && !strcasecmp(match, "object"))) {
v->object = 1;
}
ast_variable_append(category, v);
foundcat = 1;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
} else if (!strcasecmp(action, "insert")) {
if (ast_strlen_zero(var) || ast_strlen_zero(line)) {
result = UNSPECIFIED_ARGUMENT;
break;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
foundcat = 0;
while ((category = ast_category_browse_filtered(cfg, cat, category, catfilter))) {
if (!(v = ast_variable_new(var, value, dfn))) {
result = FAILURE_ALLOCATION;
break;
}
ast_variable_insert(category, v, line);
foundcat = 1;
}
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
if (!foundcat) {
result = UNKNOWN_CATEGORY;
break;
}
}
else {
ast_log(LOG_WARNING, "Action-%06d: %s not handled\n", x, action);
result = UNKNOWN_ACTION;
break;
}
}
ast_free(str1);
ast_free(str2);
return result;
}
static int action_updateconfig(struct mansession *s, const struct message *m)
{
struct ast_config *cfg;
const char *sfn = astman_get_header(m, "SrcFilename");
const char *dfn = astman_get_header(m, "DstFilename");
int res;
const char *rld = astman_get_header(m, "Reload");
config: Add option to NOT preserve effective context when changing a template Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ ........ Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07 16:56:59 +00:00
int preserve_effective_context = CONFIG_SAVE_FLAG_PRESERVE_EFFECTIVE_CONTEXT;
const char *preserve_effective_context_string = astman_get_header(m, "PreserveEffectiveContext");
struct ast_flags config_flags = { CONFIG_FLAG_WITHCOMMENTS | CONFIG_FLAG_NOCACHE };
enum error_type result;
if (ast_strlen_zero(sfn) || ast_strlen_zero(dfn)) {
astman_send_error(s, m, "Filename not specified");
return 0;
}
if (restrictedFile(sfn) || restrictedFile(dfn)) {
astman_send_error(s, m, "File requires escalated priveledges");
return 0;
}
if (!(cfg = ast_config_load2(sfn, "manager", config_flags))) {
astman_send_error(s, m, "Config file not found");
return 0;
} else if (cfg == CONFIG_STATUS_FILEINVALID) {
astman_send_error(s, m, "Config file has invalid format");
return 0;
}
result = handle_updates(s, m, cfg, dfn);
if (!result) {
ast_include_rename(cfg, sfn, dfn); /* change the include references from dfn to sfn, so things match up */
config: Add option to NOT preserve effective context when changing a template Let's say you have a template T with variable VAR1 = ON and you have a context C(T) that doesn't specify VAR1. If you read C, the effective value of VAR1 is ON. Now you change T VAR1 to OFF and call ast_config_text_file_save. The current behavior is that the file gets re-written with T/VAR1=OFF but C/VAR1=ON is added. Personally, I think this is a bug. It's preserving the effective state of C even though I didn't specify C/VAR1 in th first place. I believe the behavior should be that if I didn't specify C/VAR1 originally, then the effective value of C/VAR1 should continue to follow the inherited state. Now, if I DID explicitly specify C/VAR1, the it should be preserved even if the template changes. Even though I think the existing behavior is a bug, it's been that way forever so I'm not changing it. Instead, I've created ast_config_text_file_save2() that takes a bitmask of flags, one of which is to preserve the effective context (the current behavior). The original ast_config_text_file_save calls *2 with the preserve flag. If you want the new behavior, call *2 directly without a flag. I've also updated Manager UpdateConfig with a new parameter 'PreserveEffectiveContext' whose default is 'yes'. If you want the new behavior with UpdateConfig, set 'PreserveEffectiveContext: no'. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4297/ ........ Merged revisions 430295 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430296 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-07 16:56:59 +00:00
if (!ast_strlen_zero(preserve_effective_context_string) && !ast_true(preserve_effective_context_string)) {
preserve_effective_context = CONFIG_SAVE_FLAG_NONE;
}
res = ast_config_text_file_save2(dfn, cfg, "Manager", preserve_effective_context);
ast_config_destroy(cfg);
if (res) {
astman_send_error(s, m, "Save of config failed");
return 0;
}
astman_send_ack(s, m, NULL);
if (!ast_strlen_zero(rld)) {
if (ast_true(rld)) {
ast_module_reload(NULL); /* Reload everything */
} else if (!ast_false(rld)) {
ast_module_reload(rld); /* Reload the specific module */
}
}
} else {
ast_config_destroy(cfg);
switch(result) {
case UNKNOWN_ACTION:
astman_send_error(s, m, "Unknown action command");
break;
case UNKNOWN_CATEGORY:
astman_send_error(s, m, "Given category does not exist");
break;
case UNSPECIFIED_CATEGORY:
astman_send_error(s, m, "Category not specified");
break;
case UNSPECIFIED_ARGUMENT:
astman_send_error(s, m, "Problem with category, value, or line (if required)");
break;
case FAILURE_ALLOCATION:
astman_send_error(s, m, "Memory allocation failure, this should not happen");
break;
case FAILURE_NEWCAT:
astman_send_error(s, m, "Create category did not complete successfully");
break;
case FAILURE_DELCAT:
astman_send_error(s, m, "Delete category did not complete successfully");
break;
case FAILURE_EMPTYCAT:
astman_send_error(s, m, "Empty category did not complete successfully");
break;
case FAILURE_UPDATE:
astman_send_error(s, m, "Update did not complete successfully");
break;
case FAILURE_DELETE:
astman_send_error(s, m, "Delete did not complete successfully");
break;
case FAILURE_APPEND:
astman_send_error(s, m, "Append did not complete successfully");
break;
manager/config: Support templates and non-unique category names via AMI This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425384 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-13 16:12:17 +00:00
case FAILURE_TEMPLATE:
astman_send_error(s, m, "Template category not found");
break;
}
}
return 0;
}
static int action_createconfig(struct mansession *s, const struct message *m)
{
int fd;
const char *fn = astman_get_header(m, "Filename");
struct ast_str *filepath = ast_str_alloca(PATH_MAX);
ast_str_set(&filepath, 0, "%s/", ast_config_AST_CONFIG_DIR);
ast_str_append(&filepath, 0, "%s", fn);
if ((fd = open(ast_str_buffer(filepath), O_CREAT | O_EXCL, AST_FILE_MODE)) != -1) {
close(fd);
astman_send_ack(s, m, "New configuration file created successfully");
} else {
astman_send_error(s, m, strerror(errno));
}
return 0;
}
static int action_waitevent(struct mansession *s, const struct message *m)
{
const char *timeouts = astman_get_header(m, "Timeout");
int timeout = -1;
int x;
int needexit = 0;
const char *id = astman_get_header(m, "ActionID");
char idText[256];
if (!ast_strlen_zero(id)) {
snprintf(idText, sizeof(idText), "ActionID: %s\r\n", id);
} else {
idText[0] = '\0';
}
if (!ast_strlen_zero(timeouts)) {
sscanf(timeouts, "%30i", &timeout);
if (timeout < -1) {
timeout = -1;
}
/* XXX maybe put an upper bound, or prevent the use of 0 ? */
}
ast_mutex_lock(&s->session->notify_lock);
if (s->session->waiting_thread != AST_PTHREADT_NULL) {
pthread_kill(s->session->waiting_thread, SIGURG);
}
ast_mutex_unlock(&s->session->notify_lock);
ao2_lock(s->session);
if (s->session->managerid) { /* AMI-over-HTTP session */
/*
* Make sure the timeout is within the expire time of the session,
* as the client will likely abort the request if it does not see
* data coming after some amount of time.
*/
time_t now = time(NULL);
int max = s->session->sessiontimeout - now - 10;
if (max < 0) { /* We are already late. Strange but possible. */
max = 0;
}
if (timeout < 0 || timeout > max) {
timeout = max;
}
if (!s->session->send_events) { /* make sure we record events */
s->session->send_events = -1;
}
}
ao2_unlock(s->session);
ast_mutex_lock(&s->session->notify_lock);
s->session->waiting_thread = pthread_self(); /* let new events wake up this thread */
ast_mutex_unlock(&s->session->notify_lock);
ast_debug(1, "Starting waiting for an event!\n");
for (x = 0; x < timeout || timeout < 0; x++) {
ao2_lock(s->session);
if (AST_RWLIST_NEXT(s->session->last_ev, eq_next)) {
needexit = 1;
}
if (s->session->needdestroy) {
needexit = 1;
}
ao2_unlock(s->session);
/* We can have multiple HTTP session point to the same mansession entry.
* The way we deal with it is not very nice: newcomers kick out the previous
* HTTP session. XXX this needs to be improved.
*/
ast_mutex_lock(&s->session->notify_lock);
if (s->session->waiting_thread != pthread_self()) {
needexit = 1;
}
ast_mutex_unlock(&s->session->notify_lock);
if (needexit) {
break;
}
if (s->session->managerid == 0) { /* AMI session */
if (ast_wait_for_input(ast_iostream_get_fd(s->session->stream), 1000)) {
break;
}
} else { /* HTTP session */
sleep(1);
}
}
ast_debug(1, "Finished waiting for an event!\n");
ast_mutex_lock(&s->session->notify_lock);
if (s->session->waiting_thread == pthread_self()) {
struct eventqent *eqe = s->session->last_ev;
s->session->waiting_thread = AST_PTHREADT_NULL;
ast_mutex_unlock(&s->session->notify_lock);
ao2_lock(s->session);
astman_send_response(s, m, "Success", "Waiting for Event completed.");
while ((eqe = advance_event(eqe))) {
if (((s->session->readperm & eqe->category) == eqe->category)
&& ((s->session->send_events & eqe->category) == eqe->category)
&& match_filter(s, eqe->eventdata)) {
astman_append(s, "%s", eqe->eventdata);
}
s->session->last_ev = eqe;
}
astman_append(s,
"Event: WaitEventComplete\r\n"
"%s"
"\r\n", idText);
ao2_unlock(s->session);
} else {
ast_mutex_unlock(&s->session->notify_lock);
ast_debug(1, "Abandoning event request!\n");
}
return 0;
}
static int action_listcommands(struct mansession *s, const struct message *m)
{
struct manager_action *cur;
struct ast_str *temp = ast_str_alloca(MAX_AUTH_PERM_STRING);
astman_start_ack(s, m);
AST_RWLIST_RDLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, cur, list) {
if ((s->session->writeperm & cur->authority) || cur->authority == 0) {
astman_append(s, "%s: %s (Priv: %s)\r\n",
cur->action, cur->synopsis, authority_to_str(cur->authority, &temp));
}
}
AST_RWLIST_UNLOCK(&actions);
astman_append(s, "\r\n");
return 0;
}
static int action_events(struct mansession *s, const struct message *m)
{
const char *mask = astman_get_header(m, "EventMask");
int res, x;
const char *id = astman_get_header(m, "ActionID");
char id_text[256];
if (!ast_strlen_zero(id)) {
snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
} else {
id_text[0] = '\0';
}
res = set_eventmask(s, mask);
if (broken_events_action) {
/* if this option is set we should not return a response on
* error, or when all events are set */
if (res > 0) {
for (x = 0; x < ARRAY_LEN(perms); x++) {
if (!strcasecmp(perms[x].label, "all") && res == perms[x].num) {
return 0;
}
}
astman_append(s, "Response: Success\r\n%s"
"Events: On\r\n\r\n", id_text);
} else if (res == 0)
astman_append(s, "Response: Success\r\n%s"
"Events: Off\r\n\r\n", id_text);
return 0;
}
if (res > 0)
astman_append(s, "Response: Success\r\n%s"
"Events: On\r\n\r\n", id_text);
else if (res == 0)
astman_append(s, "Response: Success\r\n%s"
"Events: Off\r\n\r\n", id_text);
else
astman_send_error(s, m, "Invalid event mask");
return 0;
}
static int action_logoff(struct mansession *s, const struct message *m)
{
astman_send_response(s, m, "Goodbye", "Thanks for all the fish.");
return -1;
}
static int action_login(struct mansession *s, const struct message *m)
{
/* still authenticated - don't process again */
if (s->session->authenticated) {
astman_send_ack(s, m, "Already authenticated");
return 0;
}
if (authenticate(s, m)) {
sleep(1);
astman_send_error(s, m, "Authentication failed");
return -1;
}
s->session->authenticated = 1;
ast_atomic_fetchadd_int(&unauth_sessions, -1);
if (manager_displayconnects(s->session)) {
ast_verb(2, "%sManager '%s' logged on from %s\n", (s->session->managerid ? "HTTP " : ""), s->session->username, ast_sockaddr_stringify_addr(&s->session->addr));
}
astman_send_ack(s, m, "Authentication accepted");
if ((s->session->send_events & EVENT_FLAG_SYSTEM)
&& (s->session->readperm & EVENT_FLAG_SYSTEM)
&& ast_test_flag(&ast_options, AST_OPT_FLAG_FULLY_BOOTED)) {
struct ast_str *auth = ast_str_alloca(MAX_AUTH_PERM_STRING);
const char *cat_str = authority_to_str(EVENT_FLAG_SYSTEM, &auth);
long uptime = 0;
long lastreloaded = 0;
struct timeval tmp;
struct timeval curtime = ast_tvnow();
if (ast_startuptime.tv_sec) {
tmp = ast_tvsub(curtime, ast_startuptime);
uptime = tmp.tv_sec;
}
if (ast_lastreloadtime.tv_sec) {
tmp = ast_tvsub(curtime, ast_lastreloadtime);
lastreloaded = tmp.tv_sec;
}
astman_append(s, "Event: FullyBooted\r\n"
"Privilege: %s\r\n"
"Uptime: %ld\r\n"
"LastReload: %ld\r\n"
"Status: Fully Booted\r\n\r\n", cat_str, uptime, lastreloaded);
}
return 0;
}
static int action_challenge(struct mansession *s, const struct message *m)
{
const char *authtype = astman_get_header(m, "AuthType");
if (!strcasecmp(authtype, "MD5")) {
if (ast_strlen_zero(s->session->challenge)) {
snprintf(s->session->challenge, sizeof(s->session->challenge), "%ld", ast_random());
}
mansession_lock(s);
astman_start_ack(s, m);
astman_append(s, "Challenge: %s\r\n\r\n", s->session->challenge);
mansession_unlock(s);
} else {
astman_send_error(s, m, "Must specify AuthType");
}
return 0;
}
int ast_manager_hangup_helper(struct mansession *s,
const struct message *m, manager_hangup_handler_t hangup_handler,
manager_hangup_cause_validator_t cause_validator)
{
struct ast_channel *c = NULL;
int causecode = 0; /* all values <= 0 mean 'do not set hangupcause in channel' */
const char *id = astman_get_header(m, "ActionID");
const char *name_or_regex = astman_get_header(m, "Channel");
const char *cause = astman_get_header(m, "Cause");
char idText[256];
regex_t regexbuf;
struct ast_channel_iterator *iter = NULL;
struct ast_str *regex_string;
int channels_matched = 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 (ast_strlen_zero(name_or_regex)) {
astman_send_error(s, m, "No channel specified");
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 (!ast_strlen_zero(id)) {
snprintf(idText, sizeof(idText), "ActionID: %s\r\n", id);
} else {
idText[0] = '\0';
}
if (cause_validator) {
causecode = cause_validator(name_or_regex, cause);
} else if (!ast_strlen_zero(cause)) {
char *endptr;
causecode = strtol(cause, &endptr, 10);
if (causecode < 0 || causecode > 127 || *endptr != '\0') {
ast_log(LOG_NOTICE, "Invalid 'Cause: %s' in manager action Hangup\n", cause);
/* keep going, better to hangup without cause than to not hang up at all */
causecode = 0; /* do not set channel's hangupcause */
}
}
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
/************************************************/
/* Regular explicit match channel byname hangup */
if (name_or_regex[0] != '/') {
if (!(c = ast_channel_get_by_name(name_or_regex))) {
ast_log(LOG_NOTICE, "Request to hangup non-existent channel: %s\n",
name_or_regex);
astman_send_error(s, m, "No such channel");
return 0;
}
ast_verb(3, "%sManager '%s' from %s, hanging up channel: %s\n",
(s->session->managerid ? "HTTP " : ""),
s->session->username,
ast_sockaddr_stringify_addr(&s->session->addr),
ast_channel_name(c));
hangup_handler(c, causecode);
c = ast_channel_unref(c);
astman_send_ack(s, m, "Channel Hungup");
return 0;
}
/***********************************************/
/* find and hangup any channels matching regex */
regex_string = ast_str_create(strlen(name_or_regex));
if (!regex_string) {
astman_send_error(s, m, "Memory Allocation Failure");
return 0;
}
/* Make "/regex/" into "regex" */
if (ast_regex_string_to_regex_pattern(name_or_regex, &regex_string) != 0) {
astman_send_error(s, m, "Regex format invalid, Channel param should be /regex/");
ast_free(regex_string);
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 regex compilation fails, hangup fails */
if (regcomp(&regexbuf, ast_str_buffer(regex_string), REG_EXTENDED | REG_NOSUB)) {
astman_send_error_va(s, m, "Regex compile failed on: %s", name_or_regex);
ast_free(regex_string);
return 0;
}
astman_send_listack(s, m, "Channels hung up will follow", "start");
iter = ast_channel_iterator_all_new();
if (iter) {
for (; (c = ast_channel_iterator_next(iter)); ast_channel_unref(c)) {
if (regexec(&regexbuf, ast_channel_name(c), 0, NULL, 0)) {
continue;
}
ast_verb(3, "%sManager '%s' from %s, hanging up channel: %s\n",
(s->session->managerid ? "HTTP " : ""),
s->session->username,
ast_sockaddr_stringify_addr(&s->session->addr),
ast_channel_name(c));
hangup_handler(c, causecode);
channels_matched++;
astman_append(s,
"Event: ChannelHungup\r\n"
"Channel: %s\r\n"
"%s"
"\r\n", ast_channel_name(c), idText);
}
ast_channel_iterator_destroy(iter);
}
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
regfree(&regexbuf);
ast_free(regex_string);
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
astman_send_list_complete(s, m, "ChannelsHungupListComplete", channels_matched);
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
return 0;
}
static int action_hangup(struct mansession *s, const struct message *m)
{
return ast_manager_hangup_helper(s, m,
ast_channel_softhangup_withcause_locked, NULL);
}
static int action_setvar(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
const char *name = astman_get_header(m, "Channel");
const char *varname = astman_get_header(m, "Variable");
const char *varval = astman_get_header(m, "Value");
int res = 0;
if (ast_strlen_zero(varname)) {
astman_send_error(s, m, "No variable specified");
return 0;
}
if (!ast_strlen_zero(name)) {
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(name))) {
astman_send_error(s, m, "No such channel");
return 0;
}
}
res = pbx_builtin_setvar_helper(c, varname, S_OR(varval, ""));
if (c) {
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
c = ast_channel_unref(c);
}
if (res == 0) {
astman_send_ack(s, m, "Variable Set");
} else {
astman_send_error(s, m, "Variable not set");
}
return 0;
}
static int action_getvar(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
const char *name = astman_get_header(m, "Channel");
const char *varname = astman_get_header(m, "Variable");
char *varval;
Merged revisions 337974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-26 19:40:12 +00:00
char workspace[1024];
if (ast_strlen_zero(varname)) {
astman_send_error(s, m, "No variable specified");
return 0;
}
/* We don't want users with insufficient permissions using certain functions. */
if (!(function_capable_string_allowed_with_auths(varname, s->session->writeperm))) {
astman_send_error(s, m, "GetVar Access Forbidden: Variable");
return 0;
}
if (!ast_strlen_zero(name)) {
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(name))) {
astman_send_error(s, m, "No such channel");
return 0;
}
}
Merged revisions 337974 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r337974 | rmudgett | 2011-09-26 14:35:23 -0500 (Mon, 26 Sep 2011) | 37 lines Merged revisions 337973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337973 | rmudgett | 2011-09-26 14:30:39 -0500 (Mon, 26 Sep 2011) | 30 lines Fix deadlock when using dummy channels. Dummy channels created by ast_dummy_channel_alloc() should be destoyed by ast_channel_unref(). Using ast_channel_release() needlessly grabs the channel container lock and can cause a deadlock as a result. * Analyzed use of ast_dummy_channel_alloc() and made use ast_channel_unref() when done with the dummy channel. (Primary reason for the reported deadlock.) * Made app_dial.c:dial_exec_full() not call ast_call() holding any channel locks. Chan_local could not perform deadlock avoidance correctly. (Potential deadlock exposed by this issue. Secondary reason for the reported deadlock since the held lock was part of the deadlock chain.) * Fixed some uses of ast_dummy_channel_alloc() not checking the returned channel pointer for failure. * Fixed some potential chan=NULL pointer usage in func_odbc.c. Protected by testing the bogus_chan value. * Fixed needlessly clearing a 1024 char auto array when setting the first char to zero is enough in manager.c:action_getvar(). (closes issue ASTERISK-18613) Reported by: Thomas Arimont Patches: jira_asterisk_18613_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: Thomas Arimont ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-09-26 19:40:12 +00:00
workspace[0] = '\0';
if (varname[strlen(varname) - 1] == ')') {
if (!c) {
c = ast_dummy_channel_alloc();
if (c) {
ast_func_read(c, (char *) varname, workspace, sizeof(workspace));
} else
ast_log(LOG_ERROR, "Unable to allocate bogus channel for variable substitution. Function results may be blank.\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
} else {
ast_func_read(c, (char *) varname, workspace, sizeof(workspace));
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
}
varval = workspace;
} else {
pbx_retrieve_variable(c, varname, &varval, workspace, sizeof(workspace), NULL);
}
if (c) {
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
c = ast_channel_unref(c);
}
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
astman_start_ack(s, m);
astman_append(s, "Variable: %s\r\nValue: %s\r\n\r\n", varname, S_OR(varval, ""));
return 0;
}
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
static void generate_status(struct mansession *s, struct ast_channel *chan, char **vars, int varc, int all_variables, char *id_text, int *count)
{
struct timeval now;
long elapsed_seconds;
struct ast_bridge *bridge;
RAII_VAR(struct ast_str *, variable_str, NULL, ast_free);
struct ast_str *write_transpath = ast_str_alloca(256);
struct ast_str *read_transpath = ast_str_alloca(256);
struct ast_str *codec_buf = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
struct ast_party_id effective_id;
int i;
RAII_VAR(struct ast_channel_snapshot *, snapshot,
ast_channel_snapshot_get_latest(ast_channel_uniqueid(chan)),
ao2_cleanup);
RAII_VAR(struct ast_str *, snapshot_str, NULL, ast_free);
if (!snapshot) {
return;
}
snapshot_str = ast_manager_build_channel_state_string(snapshot);
if (!snapshot_str) {
return;
}
if (all_variables) {
variable_str = ast_str_create(2048);
} else {
variable_str = ast_str_create(1024);
}
if (!variable_str) {
return;
}
now = ast_tvnow();
elapsed_seconds = ast_tvdiff_sec(now, ast_channel_creationtime(chan));
/* Even if all_variables has been specified, explicitly requested variables
* may be global variables or dialplan functions */
for (i = 0; i < varc; i++) {
char valbuf[512], *ret = NULL;
if (vars[i][strlen(vars[i]) - 1] == ')') {
if (ast_func_read(chan, vars[i], valbuf, sizeof(valbuf)) < 0) {
valbuf[0] = '\0';
}
ret = valbuf;
} else {
pbx_retrieve_variable(chan, vars[i], &ret, valbuf, sizeof(valbuf), NULL);
}
ast_str_append(&variable_str, 0, "Variable: %s=%s\r\n", vars[i], ret);
}
/* Walk all channel variables and add them */
if (all_variables) {
struct ast_var_t *variables;
AST_LIST_TRAVERSE(ast_channel_varshead(chan), variables, entries) {
ast_str_append(&variable_str, 0, "Variable: %s=%s\r\n",
ast_var_name(variables), ast_var_value(variables));
}
}
bridge = ast_channel_get_bridge(chan);
effective_id = ast_channel_connected_effective_id(chan);
astman_append(s,
"Event: Status\r\n"
"Privilege: Call\r\n"
"%s"
"Type: %s\r\n"
"DNID: %s\r\n"
"EffectiveConnectedLineNum: %s\r\n"
"EffectiveConnectedLineName: %s\r\n"
"TimeToHangup: %ld\r\n"
"BridgeID: %s\r\n"
"Application: %s\r\n"
"Data: %s\r\n"
"Nativeformats: %s\r\n"
"Readformat: %s\r\n"
"Readtrans: %s\r\n"
"Writeformat: %s\r\n"
"Writetrans: %s\r\n"
"Callgroup: %llu\r\n"
"Pickupgroup: %llu\r\n"
"Seconds: %ld\r\n"
"%s"
"%s"
"\r\n",
ast_str_buffer(snapshot_str),
ast_channel_tech(chan)->type,
S_OR(ast_channel_dialed(chan)->number.str, ""),
S_COR(effective_id.number.valid, effective_id.number.str, "<unknown>"),
S_COR(effective_id.name.valid, effective_id.name.str, "<unknown>"),
(long)ast_channel_whentohangup(chan)->tv_sec,
bridge ? bridge->uniqueid : "",
ast_channel_appl(chan),
ast_channel_data(chan),
ast_format_cap_get_names(ast_channel_nativeformats(chan), &codec_buf),
ast_format_get_name(ast_channel_readformat(chan)),
ast_translate_path_to_str(ast_channel_readtrans(chan), &read_transpath),
ast_format_get_name(ast_channel_writeformat(chan)),
ast_translate_path_to_str(ast_channel_writetrans(chan), &write_transpath),
ast_channel_callgroup(chan),
ast_channel_pickupgroup(chan),
(long)elapsed_seconds,
ast_str_buffer(variable_str),
id_text);
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
++*count;
ao2_cleanup(bridge);
}
/*! \brief Manager "status" command to show channels */
static int action_status(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, "Channel");
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
const char *chan_variables = astman_get_header(m, "Variables");
const char *all_chan_variables = astman_get_header(m, "AllVariables");
int all_variables = 0;
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
const char *id = astman_get_header(m, "ActionID");
char *variables = ast_strdupa(S_OR(chan_variables, ""));
struct ast_channel *chan;
int channels = 0;
int all = ast_strlen_zero(name); /* set if we want all channels */
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
char id_text[256];
struct ast_channel_iterator *it_chans = NULL;
AST_DECLARE_APP_ARGS(vars,
AST_APP_ARG(name)[100];
);
if (!ast_strlen_zero(all_chan_variables)) {
all_variables = ast_true(all_chan_variables);
}
if (!(function_capable_string_allowed_with_auths(variables, s->session->writeperm))) {
astman_send_error(s, m, "Status Access Forbidden: Variables");
return 0;
}
if (all) {
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
if (!(it_chans = ast_channel_iterator_all_new())) {
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
astman_send_error(s, m, "Memory Allocation Failure");
return 1;
}
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
chan = ast_channel_iterator_next(it_chans);
} else {
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
chan = ast_channel_get_by_name(name);
if (!chan) {
astman_send_error(s, m, "No such channel");
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
astman_send_listack(s, m, "Channel status will follow", "start");
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
if (!ast_strlen_zero(id)) {
snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", id);
} else {
id_text[0] = '\0';
}
if (!ast_strlen_zero(chan_variables)) {
AST_STANDARD_APP_ARGS(vars, variables);
}
/* if we look by name, we break after the first iteration */
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
for (; chan; all ? chan = ast_channel_iterator_next(it_chans) : 0) {
ast_channel_lock(chan);
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
generate_status(s, chan, vars.name, vars.argc, all_variables, id_text, &channels);
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_unlock(chan);
chan = ast_channel_unref(chan);
}
if (it_chans) {
ast_channel_iterator_destroy(it_chans);
}
AMI: Make AMI actions that generate event lists consistent. * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ ........ Merged revisions 430434 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-09 18:16:54 +00:00
astman_send_list_complete_start(s, m, "StatusComplete", channels);
astman_append(s, "Items: %d\r\n", channels);
astman_send_list_complete_end(s);
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
return 0;
}
/*!
* \brief Queue a given read action containing a payload onto a channel
*
* This queues a READ_ACTION control frame that contains a given "payload", or
* data to be triggered and handled on the channel's read side. This ensures
* the "action" is handled by the channel's media reading thread.
*
* \param chan The channel to queue the action on
* \param payload The read action's payload
* \param payload_size The size of the given payload
* \param action The type of read action to queue
*
* \retval -1 on error
* \retval 0 on success
*/
static int queue_read_action_payload(struct ast_channel *chan, const unsigned char *payload,
size_t payload_size, enum ast_frame_read_action action)
{
struct ast_control_read_action_payload *obj;
size_t obj_size;
int res;
obj_size = payload_size + sizeof(*obj);
obj = ast_malloc(obj_size);
if (!obj) {
return -1;
}
obj->action = action;
obj->payload_size = payload_size;
memcpy(obj->payload, payload, payload_size);
res = ast_queue_control_data(chan, AST_CONTROL_READ_ACTION, obj, obj_size);
ast_free(obj);
return res;
}
/*!
* \brief Queue a read action to send a text message
*
* \param chan The channel to queue the action on
* \param body The body of the message
*
* \retval -1 on error
* \retval 0 on success
*/
static int queue_sendtext(struct ast_channel *chan, const char *body)
{
return queue_read_action_payload(chan, (const unsigned char *)body,
strlen(body) + 1, AST_FRAME_READ_ACTION_SEND_TEXT);
}
/*!
* \brief Queue a read action to send a text data message
*
* \param chan The channel to queue the action on
* \param body The body of the message
* \param content_type The message's content type
*
* \retval -1 on error
* \retval 0 on success
*/
static int queue_sendtext_data(struct ast_channel *chan, const char *body,
const char *content_type)
{
int res;
struct ast_msg_data *obj;
obj = ast_msg_data_alloc2(AST_MSG_DATA_SOURCE_TYPE_UNKNOWN,
NULL, NULL, content_type, body);
if (!obj) {
return -1;
}
res = queue_read_action_payload(chan, (const unsigned char *)obj,
ast_msg_data_get_length(obj), AST_FRAME_READ_ACTION_SEND_TEXT_DATA);
ast_free(obj);
return res;
}
static int action_sendtext(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
const char *textmsg = astman_get_header(m, "Message");
const char *content_type = astman_get_header(m, "Content-Type");
int res;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return 0;
}
if (ast_strlen_zero(textmsg)) {
astman_send_error(s, m, "No Message specified");
return 0;
}
c = ast_channel_get_by_name(name);
if (!c) {
astman_send_error(s, m, "No such channel");
return 0;
}
/*
* If the "extra" data is not available, then send using "string" only.
* Doing such maintains backward compatibilities.
*/
res = ast_strlen_zero(content_type) ? queue_sendtext(c, textmsg) :
queue_sendtext_data(c, textmsg, content_type);
ast_channel_unref(c);
if (res >= 0) {
astman_send_ack(s, m, "Success");
} else {
astman_send_error(s, m, "Failure");
}
return 0;
}
/*! \brief action_redirect: The redirect manager command */
static int action_redirect(struct mansession *s, const struct message *m)
{
char buf[256];
const char *name = astman_get_header(m, "Channel");
const char *name2 = astman_get_header(m, "ExtraChannel");
const char *exten = astman_get_header(m, "Exten");
const char *exten2 = astman_get_header(m, "ExtraExten");
const char *context = astman_get_header(m, "Context");
const char *context2 = astman_get_header(m, "ExtraContext");
const char *priority = astman_get_header(m, "Priority");
const char *priority2 = astman_get_header(m, "ExtraPriority");
struct ast_channel *chan;
struct ast_channel *chan2;
int pi = 0;
int pi2 = 0;
int res;
int chan1_wait = 0;
int chan2_wait = 0;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "Channel not specified");
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 (ast_strlen_zero(context)) {
astman_send_error(s, m, "Context not specified");
return 0;
}
if (ast_strlen_zero(exten)) {
astman_send_error(s, m, "Exten not specified");
return 0;
}
if (ast_strlen_zero(priority)) {
astman_send_error(s, m, "Priority not specified");
return 0;
}
if (sscanf(priority, "%30d", &pi) != 1) {
pi = ast_findlabel_extension(NULL, context, exten, priority, NULL);
}
if (pi < 1) {
astman_send_error(s, m, "Priority is invalid");
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 (!ast_strlen_zero(name2) && !ast_strlen_zero(context2)) {
/* We have an ExtraChannel and an ExtraContext */
if (ast_strlen_zero(exten2)) {
astman_send_error(s, m, "ExtraExten not specified");
return 0;
}
if (ast_strlen_zero(priority2)) {
astman_send_error(s, m, "ExtraPriority not specified");
return 0;
}
if (sscanf(priority2, "%30d", &pi2) != 1) {
pi2 = ast_findlabel_extension(NULL, context2, exten2, priority2, NULL);
}
if (pi2 < 1) {
astman_send_error(s, m, "ExtraPriority is invalid");
return 0;
}
}
chan = ast_channel_get_by_name(name);
if (!chan) {
snprintf(buf, sizeof(buf), "Channel does not exist: %s", name);
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
if (ast_check_hangup_locked(chan)) {
astman_send_error(s, m, "Redirect failed, channel not up.");
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
if (ast_strlen_zero(name2)) {
/* Single channel redirect in progress. */
res = ast_async_goto(chan, context, exten, pi);
if (!res) {
astman_send_ack(s, m, "Redirect successful");
} else {
astman_send_error(s, m, "Redirect failed");
}
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
}
chan2 = ast_channel_get_by_name(name2);
if (!chan2) {
snprintf(buf, sizeof(buf), "ExtraChannel does not exist: %s", name2);
astman_send_error(s, m, buf);
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;
}
if (ast_check_hangup_locked(chan2)) {
astman_send_error(s, m, "Redirect failed, extra channel not up.");
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_unref(chan2);
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
/* Dual channel redirect in progress. */
ast_channel_lock(chan);
if (ast_channel_is_bridged(chan)) {
ast_set_flag(ast_channel_flags(chan), AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT);
chan1_wait = 1;
Merged revisions 172030 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172030 | murf | 2009-01-28 11:51:16 -0700 (Wed, 28 Jan 2009) | 46 lines This patch fixes h-exten running misbehavior in manager-redirected situations. What it does: 1. A new Flag value is defined in include/asterisk/channel.h, AST_FLAG_BRIDGE_HANGUP_DONT, which used as a messenge to the bridge hangup exten code not to run the h-exten there (nor publish the bridge cdr there). It will done at the pbx-loop level instead. 2. In the manager Redirect code, I set this flag on the channel if the channel has a non-null pbx pointer. I did the same for the second (chan2) channel, which gets run if name2 is set... and the first succeeds. 3. I restored the ending of the cdr for the pbx loop h-exten running code. Don't know why it was removed in the first place. 4. The first attempt at the fix for this bug was to place code directly in the async_goto routine, which was called from a large number of places, and could affect a large number of cases, so I tested that fix against a fair number of transfer scenarios, both with and without the patch. In the process, I saw that putting the fix in async_goto seemed not to affect any of the blind or attended scenarios, but still, I was was highly concerned that some other scenarios I had not tested might be negatively impacted, so I refined the patch to its current scope, and jmls tested both. In the process, tho, I saw that blind xfers in one situation, when the one-touch blind-xfer feature is used by the peer, we got strange h-exten behavior. So, I inserted code to swap CDRs and to set the HANGUP_DONT field, to get uniform behavior. 5. I added code to the bridge to obey the HANGUP_DONT flag, skipping both publishing the bridge CDR, and running the h-exten; they will be done at the pbx-loop (higher) level instead. 6. I removed all the debug logs from the patch before committing. 7. I moved the AUTOLOOP set/reset in the h-exten code in res_features so it's only done if the h-exten is going to be run. A very minor performance improvement, but technically correct. (closes issue #14241) Reported by: jmls Patches: 14241_redirect_no_bridgeCDR_or_h_exten_via_transfer uploaded by murf (license 17) Tested by: murf, jmls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 20:31:06 +00:00
}
ast_channel_unlock(chan);
ast_channel_lock(chan2);
if (ast_channel_is_bridged(chan2)) {
ast_set_flag(ast_channel_flags(chan2), AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT);
chan2_wait = 1;
}
ast_channel_unlock(chan2);
res = ast_async_goto(chan, context, exten, pi);
if (!res) {
if (!ast_strlen_zero(context2)) {
res = ast_async_goto(chan2, context2, exten2, pi2);
} else {
res = ast_async_goto(chan2, context, exten, pi);
}
if (!res) {
astman_send_ack(s, m, "Dual Redirect successful");
} else {
astman_send_error(s, m, "Secondary redirect failed");
}
} else {
astman_send_error(s, m, "Redirect failed");
}
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
/* Release the bridge wait. */
if (chan1_wait) {
ast_channel_clear_flag(chan, AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT);
}
if (chan2_wait) {
ast_channel_clear_flag(chan2, AST_FLAG_BRIDGE_DUAL_REDIRECT_WAIT);
}
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_unref(chan2);
chan = ast_channel_unref(chan);
return 0;
}
static int action_blind_transfer(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, "Channel");
const char *exten = astman_get_header(m, "Exten");
const char *context = astman_get_header(m, "Context");
struct ast_channel *chan;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return 0;
}
if (ast_strlen_zero(exten)) {
astman_send_error(s, m, "No extension specified");
return 0;
}
chan = ast_channel_get_by_name(name);
if (!chan) {
astman_send_error(s, m, "Channel specified does not exist");
return 0;
}
if (ast_strlen_zero(context)) {
context = ast_channel_context(chan);
}
switch (ast_bridge_transfer_blind(1, chan, exten, context, NULL, NULL)) {
case AST_BRIDGE_TRANSFER_NOT_PERMITTED:
astman_send_error(s, m, "Transfer not permitted");
break;
case AST_BRIDGE_TRANSFER_INVALID:
astman_send_error(s, m, "Transfer invalid");
break;
case AST_BRIDGE_TRANSFER_FAIL:
astman_send_error(s, m, "Transfer failed");
break;
case AST_BRIDGE_TRANSFER_SUCCESS:
astman_send_ack(s, m, "Transfer succeeded");
break;
}
ast_channel_unref(chan);
return 0;
}
static int action_atxfer(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, "Channel");
const char *exten = astman_get_header(m, "Exten");
const char *context = astman_get_header(m, "Context");
struct ast_channel *chan = NULL;
char feature_code[AST_FEATURE_MAX_LEN];
const char *digit;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return 0;
}
if (ast_strlen_zero(exten)) {
astman_send_error(s, m, "No extension specified");
return 0;
}
if (!(chan = ast_channel_get_by_name(name))) {
astman_send_error(s, m, "Channel specified does not exist");
return 0;
}
ast_channel_lock(chan);
if (ast_get_builtin_feature(chan, "atxfer", feature_code, sizeof(feature_code)) ||
ast_strlen_zero(feature_code)) {
ast_channel_unlock(chan);
astman_send_error(s, m, "No attended transfer feature code found");
ast_channel_unref(chan);
return 0;
}
ast_channel_unlock(chan);
if (!ast_strlen_zero(context)) {
pbx_builtin_setvar_helper(chan, "TRANSFER_CONTEXT", context);
}
for (digit = feature_code; *digit; ++digit) {
struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit };
ast_queue_frame(chan, &f);
}
for (digit = exten; *digit; ++digit) {
struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit };
ast_queue_frame(chan, &f);
}
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, "Atxfer successfully queued");
return 0;
}
static int action_cancel_atxfer(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, "Channel");
struct ast_channel *chan = NULL;
char *feature_code;
const char *digit;
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
return 0;
}
if (!(chan = ast_channel_get_by_name(name))) {
astman_send_error(s, m, "Channel specified does not exist");
return 0;
}
ast_channel_lock(chan);
feature_code = ast_get_chan_features_atxferabort(chan);
ast_channel_unlock(chan);
if (!feature_code) {
astman_send_error(s, m, "No disconnect feature code found");
ast_channel_unref(chan);
return 0;
}
for (digit = feature_code; *digit; ++digit) {
struct ast_frame f = { AST_FRAME_DTMF, .subclass.integer = *digit };
ast_queue_frame(chan, &f);
}
ast_free(feature_code);
chan = ast_channel_unref(chan);
astman_send_ack(s, m, "CancelAtxfer successfully queued");
return 0;
}
static int check_blacklist(const char *cmd)
{
char *cmd_copy, *cur_cmd;
char *cmd_words[AST_MAX_CMD_LEN] = { NULL, };
int i;
cmd_copy = ast_strdupa(cmd);
for (i = 0; i < MAX_BLACKLIST_CMD_LEN && (cur_cmd = strsep(&cmd_copy, " ")); i++) {
cur_cmd = ast_strip(cur_cmd);
if (ast_strlen_zero(cur_cmd)) {
i--;
continue;
}
cmd_words[i] = cur_cmd;
}
for (i = 0; i < ARRAY_LEN(command_blacklist); i++) {
int j, match = 1;
for (j = 0; command_blacklist[i].words[j]; j++) {
if (ast_strlen_zero(cmd_words[j]) || strcasecmp(cmd_words[j], command_blacklist[i].words[j])) {
match = 0;
break;
}
}
if (match) {
return 1;
}
}
return 0;
}
/*! \brief Manager command "command" - execute CLI command */
static int action_command(struct mansession *s, const struct message *m)
{
const char *cmd = astman_get_header(m, "Command");
char *buf = NULL, *final_buf = NULL, *delim, *output;
char template[] = "/tmp/ast-ami-XXXXXX"; /* template for temporary file */
int fd, ret;
off_t len;
if (ast_strlen_zero(cmd)) {
astman_send_error(s, m, "No command provided");
return 0;
}
if (check_blacklist(cmd)) {
astman_send_error(s, m, "Command blacklisted");
return 0;
}
if ((fd = mkstemp(template)) < 0) {
astman_send_error_va(s, m, "Failed to create temporary file: %s", strerror(errno));
return 0;
}
ret = ast_cli_command(fd, cmd);
astman_send_response_full(s, m, ret == RESULT_SUCCESS ? "Success" : "Error", MSG_MOREDATA, NULL);
/* Determine number of characters available */
if ((len = lseek(fd, 0, SEEK_END)) < 0) {
astman_append(s, "Message: Failed to determine number of characters: %s\r\n", strerror(errno));
goto action_command_cleanup;
}
/* This has a potential to overflow the stack. Hence, use the heap. */
buf = ast_malloc(len + 1);
final_buf = ast_malloc(len + 1);
if (!buf || !final_buf) {
astman_append(s, "Message: Memory allocation failure\r\n");
goto action_command_cleanup;
}
if (lseek(fd, 0, SEEK_SET) < 0) {
astman_append(s, "Message: Failed to set position on temporary file: %s\r\n", strerror(errno));
goto action_command_cleanup;
}
if (read(fd, buf, len) < 0) {
astman_append(s, "Message: Failed to read from temporary file: %s\r\n", strerror(errno));
goto action_command_cleanup;
}
buf[len] = '\0';
term_strip(final_buf, buf, len);
final_buf[len] = '\0';
/* Trim trailing newline */
if (len && final_buf[len - 1] == '\n') {
final_buf[len - 1] = '\0';
}
astman_append(s, "Message: Command output follows\r\n");
delim = final_buf;
while ((output = strsep(&delim, "\n"))) {
astman_append(s, "Output: %s\r\n", output);
}
action_command_cleanup:
astman_append(s, "\r\n");
close(fd);
unlink(template);
ast_free(buf);
ast_free(final_buf);
return 0;
}
/*! \brief helper function for originate */
struct fast_originate_helper {
int timeout;
struct ast_format_cap *cap; /*!< Codecs used for a call */
int early_media;
AST_DECLARE_STRING_FIELDS (
AST_STRING_FIELD(tech);
/*! data can contain a channel name, extension number, username, password, etc. */
AST_STRING_FIELD(data);
AST_STRING_FIELD(app);
AST_STRING_FIELD(appdata);
AST_STRING_FIELD(cid_name);
AST_STRING_FIELD(cid_num);
AST_STRING_FIELD(context);
AST_STRING_FIELD(exten);
AST_STRING_FIELD(idtext);
AST_STRING_FIELD(account);
AST_STRING_FIELD(channelid);
AST_STRING_FIELD(otherchannelid);
);
int priority;
struct ast_variable *vars;
};
/*!
* \internal
*
* \param doomed Struct to destroy.
*/
static void destroy_fast_originate_helper(struct fast_originate_helper *doomed)
{
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_cleanup(doomed->cap);
ast_variables_destroy(doomed->vars);
ast_string_field_free_memory(doomed);
ast_free(doomed);
}
static void *fast_originate(void *data)
{
struct fast_originate_helper *in = data;
int res;
int reason = 0;
struct ast_channel *chan = NULL, *chans[1];
Merged revisions 47986,47995,47997,48001,48003-48004,48008-48014,48016,48018-48019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r47986 | oej | 2006-11-24 07:00:19 -0700 (Fri, 24 Nov 2006) | 6 lines Doxygen update - Document cause codes - Document a bit more on channel variables - global, predefined and local - Fix some doxygen in channel.h. Adding one comment for two definitions does not work. They won't be copied to each. ................ r47995 | murf | 2006-11-24 10:40:49 -0700 (Fri, 24 Nov 2006) | 1 line This fix inspired by a patch supplied in bug 8189, which points out problems with the PLC code ................ r47997 | murf | 2006-11-24 11:17:25 -0700 (Fri, 24 Nov 2006) | 1 line removed the svnmerge-integrated property from trunk; it's confusing svnmerge in newly created branches ................ r48001 | rizzo | 2006-11-25 02:02:42 -0700 (Sat, 25 Nov 2006) | 5 lines set pointers to NULL after freeing memory to avoid multiple free() probably 1.4/1.2 issue as well if someone can look into that. ................ r48003 | oej | 2006-11-25 02:45:57 -0700 (Sat, 25 Nov 2006) | 9 lines - Adding comment on suspicious memory allocation. Seems like it's never freed, but I don't have a clear understanding of the frame allocation/deallocation, so I just mark this for investigation. (Reported by Ed Guy). We're trying to see if a free() hurts... - Doxygen comments on p2p rtp bridge stuff. I am a bit worried about shortcutting rtcp this way, but will need feedback from rtcp gurus. This should work for video calls too, and possibly UDPTL. ................ r48004 | oej | 2006-11-25 02:48:30 -0700 (Sat, 25 Nov 2006) | 2 lines Changing ERROR to lesser level. Imported from 1.2/1.4 ................ r48008 | rizzo | 2006-11-25 10:37:04 -0700 (Sat, 25 Nov 2006) | 7 lines generalize a bit the functions used to create an tcp socket and then run a service on it. The code in manager.c does essentially the same things, so we will be able to reuse the code in here (probably moving it to netsock.c or another appropriate library file). ................ r48009 | mattf | 2006-11-25 13:30:04 -0700 (Sat, 25 Nov 2006) | 1 line Updates to show linkset command ................ r48010 | mattf | 2006-11-25 13:54:38 -0700 (Sat, 25 Nov 2006) | 2 lines Add ss7 show linkset command ................ r48011 | mattf | 2006-11-25 14:32:33 -0700 (Sat, 25 Nov 2006) | 1 line Make sure we don't send a group reset on a group larger than 32 CICs ................ r48012 | mattf | 2006-11-25 14:35:23 -0700 (Sat, 25 Nov 2006) | 1 line bug fix ................ r48013 | mattf | 2006-11-25 14:46:58 -0700 (Sat, 25 Nov 2006) | 1 line Make compiler happier ................ r48014 | mattf | 2006-11-25 14:50:42 -0700 (Sat, 25 Nov 2006) | 1 line Little fix so we use the right message ................ r48016 | murf | 2006-11-25 17:15:42 -0700 (Sat, 25 Nov 2006) | 9 lines Merged revisions 48015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48015 | murf | 2006-11-25 17:01:34 -0700 (Sat, 25 Nov 2006) | 1 line A little bit of func_cdr documentation upgrade-- no bug# involved, although 8221 may have inspired it. ........ ................ r48018 | murf | 2006-11-25 17:31:13 -0700 (Sat, 25 Nov 2006) | 9 lines Merged revisions 48017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r48017 | murf | 2006-11-25 17:26:16 -0700 (Sat, 25 Nov 2006) | 1 line might as well also document the raw values of the flag vars ........ ................ r48019 | russell | 2006-11-25 23:55:33 -0700 (Sat, 25 Nov 2006) | 6 lines - Add some comments on thread storage with a brief explanation of what it is as well as what the motivation is for using it. - Add a comment by the declaration of ast_inet_ntoa() noting that this function is not reentrant, and the result of a previous call to the function is no longer valid after calling it again. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-07 16:03:56 +00:00
char requested_channel[AST_CHANNEL_NAME];
struct ast_assigned_ids assignedids = {
.uniqueid = in->channelid,
.uniqueid2 = in->otherchannelid
};
if (!ast_strlen_zero(in->app)) {
res = ast_pbx_outgoing_app(in->tech, in->cap, in->data,
in->timeout, in->app, in->appdata, &reason,
AST_OUTGOING_WAIT,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan, &assignedids);
} else {
res = ast_pbx_outgoing_exten(in->tech, in->cap, in->data,
in->timeout, in->context, in->exten, in->priority, &reason,
AST_OUTGOING_WAIT,
S_OR(in->cid_num, NULL),
S_OR(in->cid_name, NULL),
in->vars, in->account, &chan, in->early_media, &assignedids);
}
if (!chan) {
snprintf(requested_channel, AST_CHANNEL_NAME, "%s/%s", in->tech, in->data);
}
/* Tell the manager what happened with the channel */
chans[0] = chan;
if (!ast_strlen_zero(in->app)) {
ast_manager_event_multichan(EVENT_FLAG_CALL, "OriginateResponse", chan ? 1 : 0, chans,
"%s"
"Response: %s\r\n"
"Channel: %s\r\n"
"Application: %s\r\n"
"Data: %s\r\n"
"Reason: %d\r\n"
"Uniqueid: %s\r\n"
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n",
in->idtext, res ? "Failure" : "Success",
chan ? ast_channel_name(chan) : requested_channel,
in->app, in->appdata, reason,
chan ? ast_channel_uniqueid(chan) : S_OR(in->channelid, "<unknown>"),
S_OR(in->cid_num, "<unknown>"),
S_OR(in->cid_name, "<unknown>")
);
} else {
ast_manager_event_multichan(EVENT_FLAG_CALL, "OriginateResponse", chan ? 1 : 0, chans,
"%s"
"Response: %s\r\n"
"Channel: %s\r\n"
"Context: %s\r\n"
"Exten: %s\r\n"
"Reason: %d\r\n"
"Uniqueid: %s\r\n"
"CallerIDNum: %s\r\n"
"CallerIDName: %s\r\n",
in->idtext, res ? "Failure" : "Success",
chan ? ast_channel_name(chan) : requested_channel,
in->context, in->exten, reason,
chan ? ast_channel_uniqueid(chan) : S_OR(in->channelid, "<unknown>"),
S_OR(in->cid_num, "<unknown>"),
S_OR(in->cid_name, "<unknown>")
);
}
/* Locked and ref'd by ast_pbx_outgoing_exten or ast_pbx_outgoing_app */
if (chan) {
ast_channel_unlock(chan);
ast_channel_unref(chan);
}
destroy_fast_originate_helper(in);
return NULL;
}
static int aocmessage_get_unit_entry(const struct message *m, struct ast_aoc_unit_entry *entry, unsigned int entry_num)
{
const char *unitamount;
const char *unittype;
struct ast_str *str = ast_str_alloca(32);
memset(entry, 0, sizeof(*entry));
ast_str_set(&str, 0, "UnitAmount(%u)", entry_num);
unitamount = astman_get_header(m, ast_str_buffer(str));
ast_str_set(&str, 0, "UnitType(%u)", entry_num);
unittype = astman_get_header(m, ast_str_buffer(str));
if (!ast_strlen_zero(unitamount) && (sscanf(unitamount, "%30u", &entry->amount) == 1)) {
entry->valid_amount = 1;
}
if (!ast_strlen_zero(unittype) && sscanf(unittype, "%30u", &entry->type) == 1) {
entry->valid_type = 1;
}
return 0;
}
static struct ast_aoc_decoded *action_aoc_de_message(struct mansession *s, const struct message *m)
{
const char *msgtype = astman_get_header(m, "MsgType");
const char *chargetype = astman_get_header(m, "ChargeType");
const char *currencyname = astman_get_header(m, "CurrencyName");
const char *currencyamount = astman_get_header(m, "CurrencyAmount");
const char *mult = astman_get_header(m, "CurrencyMultiplier");
const char *totaltype = astman_get_header(m, "TotalType");
const char *aocbillingid = astman_get_header(m, "AOCBillingId");
const char *association_id= astman_get_header(m, "ChargingAssociationId");
const char *association_num = astman_get_header(m, "ChargingAssociationNumber");
const char *association_plan = astman_get_header(m, "ChargingAssociationPlan");
enum ast_aoc_type _msgtype;
enum ast_aoc_charge_type _chargetype;
enum ast_aoc_currency_multiplier _mult = AST_AOC_MULT_ONE;
enum ast_aoc_total_type _totaltype = AST_AOC_TOTAL;
enum ast_aoc_billing_id _billingid = AST_AOC_BILLING_NA;
unsigned int _currencyamount = 0;
int _association_id = 0;
unsigned int _association_plan = 0;
struct ast_aoc_decoded *decoded = NULL;
if (ast_strlen_zero(chargetype)) {
astman_send_error(s, m, "ChargeType not specified");
goto aocmessage_cleanup;
}
_msgtype = strcasecmp(msgtype, "d") ? AST_AOC_E : AST_AOC_D;
if (!strcasecmp(chargetype, "NA")) {
_chargetype = AST_AOC_CHARGE_NA;
} else if (!strcasecmp(chargetype, "Free")) {
_chargetype = AST_AOC_CHARGE_FREE;
} else if (!strcasecmp(chargetype, "Currency")) {
_chargetype = AST_AOC_CHARGE_CURRENCY;
} else if (!strcasecmp(chargetype, "Unit")) {
_chargetype = AST_AOC_CHARGE_UNIT;
} else {
astman_send_error(s, m, "Invalid ChargeType");
goto aocmessage_cleanup;
}
if (_chargetype == AST_AOC_CHARGE_CURRENCY) {
if (ast_strlen_zero(currencyamount) || (sscanf(currencyamount, "%30u", &_currencyamount) != 1)) {
astman_send_error(s, m, "Invalid CurrencyAmount, CurrencyAmount is a required when ChargeType is Currency");
goto aocmessage_cleanup;
}
if (ast_strlen_zero(mult)) {
astman_send_error(s, m, "ChargeMultiplier unspecified, ChargeMultiplier is required when ChargeType is Currency.");
goto aocmessage_cleanup;
} else if (!strcasecmp(mult, "onethousandth")) {
_mult = AST_AOC_MULT_ONETHOUSANDTH;
} else if (!strcasecmp(mult, "onehundredth")) {
_mult = AST_AOC_MULT_ONEHUNDREDTH;
} else if (!strcasecmp(mult, "onetenth")) {
_mult = AST_AOC_MULT_ONETENTH;
} else if (!strcasecmp(mult, "one")) {
_mult = AST_AOC_MULT_ONE;
} else if (!strcasecmp(mult, "ten")) {
_mult = AST_AOC_MULT_TEN;
} else if (!strcasecmp(mult, "hundred")) {
_mult = AST_AOC_MULT_HUNDRED;
} else if (!strcasecmp(mult, "thousand")) {
_mult = AST_AOC_MULT_THOUSAND;
} else {
astman_send_error(s, m, "Invalid ChargeMultiplier");
goto aocmessage_cleanup;
}
}
/* create decoded object and start setting values */
if (!(decoded = ast_aoc_create(_msgtype, _chargetype, 0))) {
astman_send_error(s, m, "Message Creation Failed");
goto aocmessage_cleanup;
}
if (_msgtype == AST_AOC_D) {
if (!ast_strlen_zero(totaltype) && !strcasecmp(totaltype, "subtotal")) {
_totaltype = AST_AOC_SUBTOTAL;
}
if (ast_strlen_zero(aocbillingid)) {
/* ignore this is optional */
} else if (!strcasecmp(aocbillingid, "Normal")) {
_billingid = AST_AOC_BILLING_NORMAL;
} else if (!strcasecmp(aocbillingid, "ReverseCharge")) {
_billingid = AST_AOC_BILLING_REVERSE_CHARGE;
} else if (!strcasecmp(aocbillingid, "CreditCard")) {
_billingid = AST_AOC_BILLING_CREDIT_CARD;
} else {
astman_send_error(s, m, "Invalid AOC-D AOCBillingId");
goto aocmessage_cleanup;
}
} else {
if (ast_strlen_zero(aocbillingid)) {
/* ignore this is optional */
} else if (!strcasecmp(aocbillingid, "Normal")) {
_billingid = AST_AOC_BILLING_NORMAL;
} else if (!strcasecmp(aocbillingid, "ReverseCharge")) {
_billingid = AST_AOC_BILLING_REVERSE_CHARGE;
} else if (!strcasecmp(aocbillingid, "CreditCard")) {
_billingid = AST_AOC_BILLING_CREDIT_CARD;
} else if (!strcasecmp(aocbillingid, "CallFwdUnconditional")) {
_billingid = AST_AOC_BILLING_CALL_FWD_UNCONDITIONAL;
} else if (!strcasecmp(aocbillingid, "CallFwdBusy")) {
_billingid = AST_AOC_BILLING_CALL_FWD_BUSY;
} else if (!strcasecmp(aocbillingid, "CallFwdNoReply")) {
_billingid = AST_AOC_BILLING_CALL_FWD_NO_REPLY;
} else if (!strcasecmp(aocbillingid, "CallDeflection")) {
_billingid = AST_AOC_BILLING_CALL_DEFLECTION;
} else if (!strcasecmp(aocbillingid, "CallTransfer")) {
_billingid = AST_AOC_BILLING_CALL_TRANSFER;
} else {
astman_send_error(s, m, "Invalid AOC-E AOCBillingId");
goto aocmessage_cleanup;
}
if (!ast_strlen_zero(association_id) && (sscanf(association_id, "%30d", &_association_id) != 1)) {
astman_send_error(s, m, "Invalid ChargingAssociationId");
goto aocmessage_cleanup;
}
if (!ast_strlen_zero(association_plan) && (sscanf(association_plan, "%30u", &_association_plan) != 1)) {
astman_send_error(s, m, "Invalid ChargingAssociationPlan");
goto aocmessage_cleanup;
}
if (_association_id) {
ast_aoc_set_association_id(decoded, _association_id);
} else if (!ast_strlen_zero(association_num)) {
ast_aoc_set_association_number(decoded, association_num, _association_plan);
}
}
if (_chargetype == AST_AOC_CHARGE_CURRENCY) {
ast_aoc_set_currency_info(decoded, _currencyamount, _mult, ast_strlen_zero(currencyname) ? NULL : currencyname);
} else if (_chargetype == AST_AOC_CHARGE_UNIT) {
struct ast_aoc_unit_entry entry;
int i;
/* multiple unit entries are possible, lets get them all */
for (i = 0; i < 32; i++) {
if (aocmessage_get_unit_entry(m, &entry, i)) {
break; /* that's the end then */
}
ast_aoc_add_unit_entry(decoded, entry.valid_amount, entry.amount, entry.valid_type, entry.type);
}
/* at least one unit entry is required */
if (!i) {
astman_send_error(s, m, "Invalid UnitAmount(0), At least one valid unit entry is required when ChargeType is set to Unit");
goto aocmessage_cleanup;
}
}
ast_aoc_set_billing_id(decoded, _billingid);
ast_aoc_set_total_type(decoded, _totaltype);
return decoded;
aocmessage_cleanup:
ast_aoc_destroy_decoded(decoded);
return NULL;
}
static int action_aoc_s_submessage(struct mansession *s, const struct message *m,
struct ast_aoc_decoded *decoded)
{
const char *chargeditem = __astman_get_header(m, "ChargedItem", GET_HEADER_LAST_MATCH);
const char *ratetype = __astman_get_header(m, "RateType", GET_HEADER_LAST_MATCH);
const char *currencyname = __astman_get_header(m, "CurrencyName", GET_HEADER_LAST_MATCH);
const char *currencyamount = __astman_get_header(m, "CurrencyAmount", GET_HEADER_LAST_MATCH);
const char *mult = __astman_get_header(m, "CurrencyMultiplier", GET_HEADER_LAST_MATCH);
const char *time = __astman_get_header(m, "Time", GET_HEADER_LAST_MATCH);
const char *timescale = __astman_get_header(m, "TimeScale", GET_HEADER_LAST_MATCH);
const char *granularity = __astman_get_header(m, "Granularity", GET_HEADER_LAST_MATCH);
const char *granularitytimescale = __astman_get_header(m, "GranularityTimeScale", GET_HEADER_LAST_MATCH);
const char *chargingtype = __astman_get_header(m, "ChargingType", GET_HEADER_LAST_MATCH);
const char *volumeunit = __astman_get_header(m, "VolumeUnit", GET_HEADER_LAST_MATCH);
const char *code = __astman_get_header(m, "Code", GET_HEADER_LAST_MATCH);
enum ast_aoc_s_charged_item _chargeditem;
enum ast_aoc_s_rate_type _ratetype;
enum ast_aoc_currency_multiplier _mult = AST_AOC_MULT_ONE;
unsigned int _currencyamount = 0;
unsigned int _code;
unsigned int _time = 0;
enum ast_aoc_time_scale _scale = 0;
unsigned int _granularity = 0;
enum ast_aoc_time_scale _granularity_time_scale = AST_AOC_TIME_SCALE_MINUTE;
int _step = 0;
enum ast_aoc_volume_unit _volumeunit = 0;
if (ast_strlen_zero(chargeditem)) {
astman_send_error(s, m, "ChargedItem not specified");
goto aocmessage_cleanup;
}
if (ast_strlen_zero(ratetype)) {
astman_send_error(s, m, "RateType not specified");
goto aocmessage_cleanup;
}
if (!strcasecmp(chargeditem, "NA")) {
_chargeditem = AST_AOC_CHARGED_ITEM_NA;
} else if (!strcasecmp(chargeditem, "SpecialArrangement")) {
_chargeditem = AST_AOC_CHARGED_ITEM_SPECIAL_ARRANGEMENT;
} else if (!strcasecmp(chargeditem, "BasicCommunication")) {
_chargeditem = AST_AOC_CHARGED_ITEM_BASIC_COMMUNICATION;
} else if (!strcasecmp(chargeditem, "CallAttempt")) {
_chargeditem = AST_AOC_CHARGED_ITEM_CALL_ATTEMPT;
} else if (!strcasecmp(chargeditem, "CallSetup")) {
_chargeditem = AST_AOC_CHARGED_ITEM_CALL_SETUP;
} else if (!strcasecmp(chargeditem, "UserUserInfo")) {
_chargeditem = AST_AOC_CHARGED_ITEM_USER_USER_INFO;
} else if (!strcasecmp(chargeditem, "SupplementaryService")) {
_chargeditem = AST_AOC_CHARGED_ITEM_SUPPLEMENTARY_SERVICE;
} else {
astman_send_error(s, m, "Invalid ChargedItem");
goto aocmessage_cleanup;
}
if (!strcasecmp(ratetype, "NA")) {
_ratetype = AST_AOC_RATE_TYPE_NA;
} else if (!strcasecmp(ratetype, "Free")) {
_ratetype = AST_AOC_RATE_TYPE_FREE;
} else if (!strcasecmp(ratetype, "FreeFromBeginning")) {
_ratetype = AST_AOC_RATE_TYPE_FREE_FROM_BEGINNING;
} else if (!strcasecmp(ratetype, "Duration")) {
_ratetype = AST_AOC_RATE_TYPE_DURATION;
} else if (!strcasecmp(ratetype, "Flat")) {
_ratetype = AST_AOC_RATE_TYPE_FLAT;
} else if (!strcasecmp(ratetype, "Volume")) {
_ratetype = AST_AOC_RATE_TYPE_VOLUME;
} else if (!strcasecmp(ratetype, "SpecialCode")) {
_ratetype = AST_AOC_RATE_TYPE_SPECIAL_CODE;
} else {
astman_send_error(s, m, "Invalid RateType");
goto aocmessage_cleanup;
}
if (_ratetype > AST_AOC_RATE_TYPE_FREE_FROM_BEGINNING) {
if (ast_strlen_zero(currencyamount) || (sscanf(currencyamount, "%30u",
&_currencyamount) != 1)) {
astman_send_error(s, m, "Invalid CurrencyAmount, CurrencyAmount is a required when RateType is non-free");
goto aocmessage_cleanup;
}
if (ast_strlen_zero(mult)) {
astman_send_error(s, m, "ChargeMultiplier unspecified, ChargeMultiplier is required when ChargeType is Currency.");
goto aocmessage_cleanup;
} else if (!strcasecmp(mult, "onethousandth")) {
_mult = AST_AOC_MULT_ONETHOUSANDTH;
} else if (!strcasecmp(mult, "onehundredth")) {
_mult = AST_AOC_MULT_ONEHUNDREDTH;
} else if (!strcasecmp(mult, "onetenth")) {
_mult = AST_AOC_MULT_ONETENTH;
} else if (!strcasecmp(mult, "one")) {
_mult = AST_AOC_MULT_ONE;
} else if (!strcasecmp(mult, "ten")) {
_mult = AST_AOC_MULT_TEN;
} else if (!strcasecmp(mult, "hundred")) {
_mult = AST_AOC_MULT_HUNDRED;
} else if (!strcasecmp(mult, "thousand")) {
_mult = AST_AOC_MULT_THOUSAND;
} else {
astman_send_error(s, m, "Invalid ChargeMultiplier");
goto aocmessage_cleanup;
}
}
if (_ratetype == AST_AOC_RATE_TYPE_DURATION) {
if (ast_strlen_zero(timescale)) {
astman_send_error(s, m, "TimeScale unspecified, TimeScale is required when RateType is Duration.");
goto aocmessage_cleanup;
} else if (!strcasecmp(timescale, "onehundredthsecond")) {
_scale = AST_AOC_TIME_SCALE_HUNDREDTH_SECOND;
} else if (!strcasecmp(timescale, "onetenthsecond")) {
_scale = AST_AOC_TIME_SCALE_TENTH_SECOND;
} else if (!strcasecmp(timescale, "second")) {
_scale = AST_AOC_TIME_SCALE_SECOND;
} else if (!strcasecmp(timescale, "tenseconds")) {
_scale = AST_AOC_TIME_SCALE_TEN_SECOND;
} else if (!strcasecmp(timescale, "minute")) {
_scale = AST_AOC_TIME_SCALE_MINUTE;
} else if (!strcasecmp(timescale, "hour")) {
_scale = AST_AOC_TIME_SCALE_HOUR;
} else if (!strcasecmp(timescale, "day")) {
_scale = AST_AOC_TIME_SCALE_DAY;
} else {
astman_send_error(s, m, "Invalid TimeScale");
goto aocmessage_cleanup;
}
if (ast_strlen_zero(time) || (sscanf(time, "%30u", &_time) != 1)) {
astman_send_error(s, m, "Invalid Time, Time is a required when RateType is Duration");
goto aocmessage_cleanup;
}
if (!ast_strlen_zero(granularity)) {
if ((sscanf(time, "%30u", &_granularity) != 1)) {
astman_send_error(s, m, "Invalid Granularity");
goto aocmessage_cleanup;
}
if (ast_strlen_zero(granularitytimescale)) {
astman_send_error(s, m, "Invalid GranularityTimeScale, GranularityTimeScale is a required when Granularity is specified");
} else if (!strcasecmp(granularitytimescale, "onehundredthsecond")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_HUNDREDTH_SECOND;
} else if (!strcasecmp(granularitytimescale, "onetenthsecond")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_TENTH_SECOND;
} else if (!strcasecmp(granularitytimescale, "second")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_SECOND;
} else if (!strcasecmp(granularitytimescale, "tenseconds")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_TEN_SECOND;
} else if (!strcasecmp(granularitytimescale, "minute")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_MINUTE;
} else if (!strcasecmp(granularitytimescale, "hour")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_HOUR;
} else if (!strcasecmp(granularitytimescale, "day")) {
_granularity_time_scale = AST_AOC_TIME_SCALE_DAY;
} else {
astman_send_error(s, m, "Invalid GranularityTimeScale");
goto aocmessage_cleanup;
}
}
if (ast_strlen_zero(chargingtype) || strcasecmp(chargingtype, "continuouscharging") == 0) {
_step = 0;
} else if (strcasecmp(chargingtype, "stepfunction") == 0 ) {
_step = 1;
} else {
astman_send_error(s, m, "Invalid ChargingType");
goto aocmessage_cleanup;
}
}
if (_ratetype == AST_AOC_RATE_TYPE_VOLUME) {
if (ast_strlen_zero(volumeunit)) {
astman_send_error(s, m, "VolumeUnit unspecified, VolumeUnit is required when RateType is Volume.");
goto aocmessage_cleanup;
} else if (!strcasecmp(timescale, "octet")) {
_volumeunit = AST_AOC_VOLUME_UNIT_OCTET;
} else if (!strcasecmp(timescale, "segment")) {
_volumeunit = AST_AOC_VOLUME_UNIT_SEGMENT;
} else if (!strcasecmp(timescale, "message")) {
_volumeunit = AST_AOC_VOLUME_UNIT_MESSAGE;
}else {
astman_send_error(s, m, "Invalid VolumeUnit");
goto aocmessage_cleanup;
}
}
if (_chargeditem == AST_AOC_CHARGED_ITEM_SPECIAL_ARRANGEMENT
|| _ratetype == AST_AOC_RATE_TYPE_SPECIAL_CODE) {
if (ast_strlen_zero(code) || (sscanf(code, "%30u", &_code) != 1)) {
astman_send_error(s, m, "Invalid Code, Code is a required when ChargedItem is SpecialArrangement and when RateType is SpecialCode");
goto aocmessage_cleanup;
}
}
if (_chargeditem == AST_AOC_CHARGED_ITEM_SPECIAL_ARRANGEMENT) {
ast_aoc_s_add_special_arrangement(decoded, _code);
} else if (_ratetype == AST_AOC_RATE_TYPE_DURATION) {
ast_aoc_s_add_rate_duration(decoded, _chargeditem, _currencyamount, _mult,
currencyname, _time, _scale, _granularity, _granularity_time_scale, _step);
} else if (_ratetype == AST_AOC_RATE_TYPE_FLAT) {
ast_aoc_s_add_rate_flat(decoded, _chargeditem, _currencyamount, _mult,
currencyname);
} else if (_ratetype == AST_AOC_RATE_TYPE_VOLUME) {
ast_aoc_s_add_rate_volume(decoded, _chargeditem, _volumeunit, _currencyamount,
_mult, currencyname);
} else if (_ratetype == AST_AOC_RATE_TYPE_SPECIAL_CODE) {
ast_aoc_s_add_rate_special_charge_code(decoded, _chargeditem, _code);
} else if (_ratetype == AST_AOC_RATE_TYPE_FREE) {
ast_aoc_s_add_rate_free(decoded, _chargeditem, 0);
} else if (_ratetype == AST_AOC_RATE_TYPE_FREE_FROM_BEGINNING) {
ast_aoc_s_add_rate_free(decoded, _chargeditem, 1);
} else if (_ratetype == AST_AOC_RATE_TYPE_NA) {
ast_aoc_s_add_rate_na(decoded, _chargeditem);
}
return 0;
aocmessage_cleanup:
return -1;
}
static struct ast_aoc_decoded *action_aoc_s_message(struct mansession *s,
const struct message *m)
{
struct ast_aoc_decoded *decoded = NULL;
int hdrlen;
int x;
static const char hdr[] = "ChargedItem:";
struct message sm = { 0 };
int rates = 0;
if (!(decoded = ast_aoc_create(AST_AOC_S, 0, 0))) {
astman_send_error(s, m, "Message Creation Failed");
goto aocmessage_cleanup;
}
hdrlen = strlen(hdr);
for (x = 0; x < m->hdrcount; x++) {
if (strncasecmp(hdr, m->headers[x], hdrlen) == 0) {
if (rates > ast_aoc_s_get_count(decoded)) {
if (action_aoc_s_submessage(s, &sm, decoded) == -1) {
goto aocmessage_cleanup;
}
}
++rates;
}
sm.headers[sm.hdrcount] = m->headers[x];
++sm.hdrcount;
}
if (rates > ast_aoc_s_get_count(decoded)) {
if (action_aoc_s_submessage(s, &sm, decoded) == -1) {
goto aocmessage_cleanup;
}
}
return decoded;
aocmessage_cleanup:
ast_aoc_destroy_decoded(decoded);
return NULL;
}
static int action_aocmessage(struct mansession *s, const struct message *m)
{
const char *msgtype = astman_get_header(m, "MsgType");
const char *channel = astman_get_header(m, "Channel");
const char *pchannel = astman_get_header(m, "ChannelPrefix");
struct ast_channel *chan = NULL;
struct ast_aoc_decoded *decoded = NULL;
struct ast_aoc_encoded *encoded = NULL;
size_t encoded_size = 0;
if (ast_strlen_zero(channel) && ast_strlen_zero(pchannel)) {
astman_send_error(s, m, "Channel and PartialChannel are not specified. Specify at least one of these.");
goto aocmessage_cleanup;
}
if (!(chan = ast_channel_get_by_name(channel)) && !ast_strlen_zero(pchannel)) {
chan = ast_channel_get_by_name_prefix(pchannel, strlen(pchannel));
}
if (!chan) {
astman_send_error(s, m, "No such channel");
goto aocmessage_cleanup;
}
if (strcasecmp(msgtype, "d") == 0 || strcasecmp(msgtype, "e") == 0) {
decoded = action_aoc_de_message(s, m);
}
else if (strcasecmp(msgtype, "s") == 0) {
decoded = action_aoc_s_message(s, m);
}
else {
astman_send_error(s, m, "Invalid MsgType");
goto aocmessage_cleanup;
}
if (!decoded) {
goto aocmessage_cleanup;
}
if ((encoded = ast_aoc_encode(decoded, &encoded_size, chan))
&& !ast_indicate_data(chan, AST_CONTROL_AOC, encoded, encoded_size)) {
astman_send_ack(s, m, "AOC Message successfully queued on channel");
} else {
astman_send_error(s, m, "Error encoding AOC message, could not queue onto channel");
}
aocmessage_cleanup:
ast_aoc_destroy_decoded(decoded);
ast_aoc_destroy_encoded(encoded);
if (chan) {
chan = ast_channel_unref(chan);
}
return 0;
}
static int action_originate(struct mansession *s, const struct message *m)
{
const char *name = astman_get_header(m, "Channel");
const char *exten = astman_get_header(m, "Exten");
const char *context = astman_get_header(m, "Context");
const char *priority = astman_get_header(m, "Priority");
const char *timeout = astman_get_header(m, "Timeout");
const char *callerid = astman_get_header(m, "CallerID");
const char *account = astman_get_header(m, "Account");
const char *app = astman_get_header(m, "Application");
const char *appdata = astman_get_header(m, "Data");
const char *async = astman_get_header(m, "Async");
const char *id = astman_get_header(m, "ActionID");
const char *codecs = astman_get_header(m, "Codecs");
const char *early_media = astman_get_header(m, "Earlymedia");
struct ast_assigned_ids assignedids = {
.uniqueid = astman_get_header(m, "ChannelId"),
.uniqueid2 = astman_get_header(m, "OtherChannelId"),
};
struct ast_variable *vars = NULL;
char *tech, *data;
char *l = NULL, *n = NULL;
int pi = 0;
int res;
int to = 30000;
int reason = 0;
char tmp[256];
char tmp2[256];
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_cap *cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
pthread_t th;
int bridge_early = 0;
if (!cap) {
astman_send_error(s, m, "Internal Error. Memory allocation failure.");
return 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
ast_format_cap_append(cap, ast_format_slin, 0);
if ((assignedids.uniqueid && AST_MAX_PUBLIC_UNIQUEID < strlen(assignedids.uniqueid))
|| (assignedids.uniqueid2 && AST_MAX_PUBLIC_UNIQUEID < strlen(assignedids.uniqueid2))) {
astman_send_error_va(s, m, "Uniqueid length exceeds maximum of %d\n",
AST_MAX_PUBLIC_UNIQUEID);
res = 0;
goto fast_orig_cleanup;
}
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "Channel not specified");
res = 0;
goto fast_orig_cleanup;
}
if (!ast_strlen_zero(priority) && (sscanf(priority, "%30d", &pi) != 1)) {
if ((pi = ast_findlabel_extension(NULL, context, exten, priority, NULL)) < 1) {
astman_send_error(s, m, "Invalid priority");
res = 0;
goto fast_orig_cleanup;
}
}
if (!ast_strlen_zero(timeout) && (sscanf(timeout, "%30d", &to) != 1)) {
astman_send_error(s, m, "Invalid timeout");
res = 0;
goto fast_orig_cleanup;
}
ast_copy_string(tmp, name, sizeof(tmp));
tech = tmp;
data = strchr(tmp, '/');
if (!data) {
astman_send_error(s, m, "Invalid channel");
res = 0;
goto fast_orig_cleanup;
}
*data++ = '\0';
ast_copy_string(tmp2, callerid, sizeof(tmp2));
ast_callerid_parse(tmp2, &n, &l);
if (n) {
if (ast_strlen_zero(n)) {
n = NULL;
}
}
if (l) {
ast_shrink_phone_number(l);
if (ast_strlen_zero(l)) {
l = NULL;
}
}
if (!ast_strlen_zero(codecs)) {
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_remove_by_type(cap, AST_MEDIA_TYPE_UNKNOWN);
ast_format_cap_update_by_allow_disallow(cap, codecs, 1);
}
if (!ast_strlen_zero(app) && s->session) {
int bad_appdata = 0;
/* To run the System application (or anything else that goes to
* shell), you must have the additional System privilege */
if (!(s->session->writeperm & EVENT_FLAG_SYSTEM)
&& (
strcasestr(app, "system") || /* System(rm -rf /)
TrySystem(rm -rf /) */
strcasestr(app, "exec") || /* Exec(System(rm -rf /))
TryExec(System(rm -rf /)) */
strcasestr(app, "agi") || /* AGI(/bin/rm,-rf /)
EAGI(/bin/rm,-rf /) */
strcasestr(app, "mixmonitor") || /* MixMonitor(blah,,rm -rf) */
AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR The AMI Originate action can allow a remote user to specify information that can be used to execute shell commands on the system hosting Asterisk. This can result in an unwanted escalation of permissions, as the Originate action, which requires the "originate" class authorization, can be used to perform actions that would typically require the "system" class authorization. Previous attempts to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought to do so by inspecting the names of applications and functions passed in with the Originate action and, if those applications/functions matched a predefined set of values, rejecting the command if the user lacked the "system" class authorization. As noted by IBM X-Force Research, the "ExternalIVR" application is not listed in the predefined set of values. The solution for this particular vulnerability is to include the "ExternalIVR" application in the set of defined applications/functions that require "system" class authorization. Unfortunately, the approach of inspecting fields in the Originate action against known applications/functions has a significant flaw. The predefined set of values can be bypassed by creative use of the Originate action or by certain dialplan configurations, which is beyond the ability of Asterisk to analyze at run-time. Attempting to work around these scenarios would result in severely restricting the applications or functions and prevent their usage for legitimate means. As such, any additional security vulnerabilities, where an application/function that would normally require the "system" class authorization can be executed by users with the "originate" class authorization, will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has been updated to reflect that the AMI Originate action can result in commands requiring the "system" class authorization to be executed. Proper system configuration can limit the impact of such scenarios. (closes issue ASTERISK-20132) Reported by: Zubair Ashraf of IBM X-Force Research ........ Merged revisions 371998 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371999 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372000 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 16:14:26 +00:00
strcasestr(app, "externalivr") || /* ExternalIVR(rm -rf) */
strcasestr(app, "originate") || /* Originate(Local/1234,app,System,rm -rf) */
(strstr(appdata, "SHELL") && (bad_appdata = 1)) || /* NoOp(${SHELL(rm -rf /)}) */
(strstr(appdata, "EVAL") && (bad_appdata = 1)) /* NoOp(${EVAL(${some_var_containing_SHELL})}) */
)) {
char error_buf[64];
snprintf(error_buf, sizeof(error_buf), "Originate Access Forbidden: %s", bad_appdata ? "Data" : "Application");
astman_send_error(s, m, error_buf);
res = 0;
goto fast_orig_cleanup;
}
}
/* Check early if the extension exists. If not, we need to bail out here. */
if (exten && context && pi) {
if (! ast_exists_extension(NULL, context, exten, pi, l)) {
/* The extension does not exist. */
astman_send_error(s, m, "Extension does not exist.");
res = 0;
goto fast_orig_cleanup;
}
}
/* Allocate requested channel variables */
vars = astman_get_variables(m);
if (s->session && s->session->chanvars) {
struct ast_variable *v, *old;
old = vars;
vars = NULL;
/* The variables in the AMI originate action are appended at the end of the list, to override any user variables that apply */
vars = ast_variables_dup(s->session->chanvars);
if (old) {
for (v = vars; v->next; v = v->next );
v->next = old; /* Append originate variables at end of list */
}
}
/* For originate async - we can bridge in early media stage */
bridge_early = ast_true(early_media);
if (ast_true(async)) {
struct fast_originate_helper *fast;
fast = ast_calloc(1, sizeof(*fast));
if (!fast || ast_string_field_init(fast, 252)) {
ast_free(fast);
ast_variables_destroy(vars);
res = -1;
} else {
if (!ast_strlen_zero(id)) {
ast_string_field_build(fast, idtext, "ActionID: %s\r\n", id);
}
ast_string_field_set(fast, tech, tech);
ast_string_field_set(fast, data, data);
ast_string_field_set(fast, app, app);
ast_string_field_set(fast, appdata, appdata);
ast_string_field_set(fast, cid_num, l);
ast_string_field_set(fast, cid_name, n);
ast_string_field_set(fast, context, context);
ast_string_field_set(fast, exten, exten);
ast_string_field_set(fast, account, account);
ast_string_field_set(fast, channelid, assignedids.uniqueid);
ast_string_field_set(fast, otherchannelid, assignedids.uniqueid2);
fast->vars = vars;
fast->cap = cap;
cap = NULL; /* transfered originate helper the capabilities structure. It is now responsible for freeing it. */
fast->timeout = to;
fast->early_media = bridge_early;
fast->priority = pi;
if (ast_pthread_create_detached(&th, NULL, fast_originate, fast)) {
destroy_fast_originate_helper(fast);
res = -1;
} else {
res = 0;
}
}
} else if (!ast_strlen_zero(app)) {
res = ast_pbx_outgoing_app(tech, cap, data, to, app, appdata, &reason,
AST_OUTGOING_WAIT, l, n, vars, account, NULL,
assignedids.uniqueid ? &assignedids : NULL);
ast_variables_destroy(vars);
} else {
if (exten && context && pi) {
res = ast_pbx_outgoing_exten(tech, cap, data, to,
context, exten, pi, &reason, AST_OUTGOING_WAIT,
l, n, vars, account, NULL, bridge_early,
assignedids.uniqueid ? &assignedids : NULL);
ast_variables_destroy(vars);
} else {
astman_send_error(s, m, "Originate with 'Exten' requires 'Context' and 'Priority'");
ast_variables_destroy(vars);
res = 0;
goto fast_orig_cleanup;
}
}
if (!res) {
astman_send_ack(s, m, "Originate successfully queued");
} else {
astman_send_error(s, m, "Originate failed");
}
fast_orig_cleanup:
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_cleanup(cap);
return 0;
}
static int action_mailboxstatus(struct mansession *s, const struct message *m)
{
const char *mailbox = astman_get_header(m, "Mailbox");
int ret;
if (ast_strlen_zero(mailbox)) {
astman_send_error(s, m, "Mailbox not specified");
return 0;
}
ret = ast_app_has_voicemail(mailbox, NULL);
astman_start_ack(s, m);
astman_append(s, "Message: Mailbox Status\r\n"
"Mailbox: %s\r\n"
"Waiting: %d\r\n\r\n", mailbox, ret);
return 0;
}
static int action_mailboxcount(struct mansession *s, const struct message *m)
{
const char *mailbox = astman_get_header(m, "Mailbox");
int newmsgs = 0, oldmsgs = 0, urgentmsgs = 0;;
if (ast_strlen_zero(mailbox)) {
astman_send_error(s, m, "Mailbox not specified");
return 0;
}
ast_app_inboxcount2(mailbox, &urgentmsgs, &newmsgs, &oldmsgs);
astman_start_ack(s, m);
astman_append(s, "Message: Mailbox Message Count\r\n"
"Mailbox: %s\r\n"
"UrgMessages: %d\r\n"
"NewMessages: %d\r\n"
"OldMessages: %d\r\n"
"\r\n",
mailbox, urgentmsgs, newmsgs, oldmsgs);
return 0;
}
static int action_extensionstate(struct mansession *s, const struct message *m)
{
const char *exten = astman_get_header(m, "Exten");
const char *context = astman_get_header(m, "Context");
char hint[256];
int status;
if (ast_strlen_zero(exten)) {
astman_send_error(s, m, "Extension not specified");
return 0;
}
if (ast_strlen_zero(context)) {
context = "default";
}
status = ast_extension_state(NULL, context, exten);
hint[0] = '\0';
ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, context, exten);
astman_start_ack(s, m);
astman_append(s, "Message: Extension Status\r\n"
"Exten: %s\r\n"
"Context: %s\r\n"
"Hint: %s\r\n"
"Status: %d\r\n"
"StatusText: %s\r\n"
"\r\n",
exten, context, hint, status,
ast_extension_state2str(status));
return 0;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
static int action_presencestate(struct mansession *s, const struct message *m)
{
const char *provider = astman_get_header(m, "Provider");
enum ast_presence_state state;
char *subtype;
char *message;
if (ast_strlen_zero(provider)) {
astman_send_error(s, m, "No provider specified");
return 0;
}
state = ast_presence_state(provider, &subtype, &message);
if (state == AST_PRESENCE_INVALID) {
astman_send_error_va(s, m, "Invalid provider %s or provider in invalid state", provider);
return 0;
}
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
astman_start_ack(s, m);
astman_append(s, "Message: Presence State\r\n"
"State: %s\r\n", ast_presence_state2str(state));
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
if (!ast_strlen_zero(subtype)) {
astman_append(s, "Subtype: %s\r\n", subtype);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
if (!ast_strlen_zero(message)) {
/* XXX The Message header here is deprecated as it
* duplicates the action response header 'Message'.
* Remove it in the next major revision of AMI.
*/
astman_append(s, "Message: %s\r\n"
"PresenceMessage: %s\r\n",
message, message);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
}
astman_append(s, "\r\n");
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
return 0;
}
static int action_timeout(struct mansession *s, const struct message *m)
{
struct ast_channel *c;
const char *name = astman_get_header(m, "Channel");
double timeout = atof(astman_get_header(m, "Timeout"));
struct timeval when = { timeout, 0 };
if (ast_strlen_zero(name)) {
astman_send_error(s, m, "No channel specified");
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 (!timeout || timeout < 0) {
astman_send_error(s, m, "No timeout specified");
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 (!(c = ast_channel_get_by_name(name))) {
astman_send_error(s, m, "No such channel");
return 0;
}
when.tv_usec = (timeout - when.tv_sec) * 1000000.0;
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(c);
ast_channel_setwhentohangup_tv(c, when);
ast_channel_unlock(c);
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
c = ast_channel_unref(c);
astman_send_ack(s, m, "Timeout Set");
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
return 0;
}
static int whitefilter_cmp_fn(void *obj, void *arg, void *data, int flags)
{
regex_t *regex_filter = obj;
const char *eventdata = arg;
int *result = data;
if (!regexec(regex_filter, eventdata, 0, NULL, 0)) {
*result = 1;
return (CMP_MATCH | CMP_STOP);
}
return 0;
}
static int blackfilter_cmp_fn(void *obj, void *arg, void *data, int flags)
{
regex_t *regex_filter = obj;
const char *eventdata = arg;
int *result = data;
if (!regexec(regex_filter, eventdata, 0, NULL, 0)) {
*result = 0;
return (CMP_MATCH | CMP_STOP);
}
*result = 1;
return 0;
}
/*!
* \brief Manager command to add an event filter to a manager session
* \see For more details look at manager_add_filter
*/
static int action_filter(struct mansession *s, const struct message *m)
{
const char *filter = astman_get_header(m, "Filter");
const char *operation = astman_get_header(m, "Operation");
int res;
if (!strcasecmp(operation, "Add")) {
res = manager_add_filter(filter, s->session->whitefilters, s->session->blackfilters);
if (res != FILTER_SUCCESS) {
if (res == FILTER_ALLOC_FAILED) {
astman_send_error(s, m, "Internal Error. Failed to allocate regex for filter");
return 0;
} else if (res == FILTER_COMPILE_FAIL) {
astman_send_error(s, m, "Filter did not compile. Check the syntax of the filter given.");
return 0;
} else {
astman_send_error(s, m, "Internal Error. Failed adding filter.");
return 0;
}
}
astman_send_ack(s, m, "Success");
return 0;
}
astman_send_error(s, m, "Unknown operation");
return 0;
}
/*!
* \brief Add an event filter to a manager session
*
* \param filter_pattern Filter syntax to add, see below for syntax
* \param whitefilters, blackfilters
*
* \return FILTER_ALLOC_FAILED Memory allocation failure
* \return FILTER_COMPILE_FAIL If the filter did not compile
* \return FILTER_SUCCESS Success
*
* Filter will be used to match against each line of a manager event
* Filter can be any valid regular expression
* Filter can be a valid regular expression prefixed with !, which will add the filter as a black filter
*
* Examples:
* \code
* filter_pattern = "Event: Newchannel"
* filter_pattern = "Event: New.*"
* filter_pattern = "!Channel: DAHDI.*"
* \endcode
*
*/
static enum add_filter_result manager_add_filter(const char *filter_pattern, struct ao2_container *whitefilters, struct ao2_container *blackfilters) {
regex_t *new_filter = ao2_t_alloc(sizeof(*new_filter), event_filter_destructor, "event_filter allocation");
int is_blackfilter;
if (!new_filter) {
return FILTER_ALLOC_FAILED;
}
if (filter_pattern[0] == '!') {
is_blackfilter = 1;
filter_pattern++;
} else {
is_blackfilter = 0;
}
if (regcomp(new_filter, filter_pattern, REG_EXTENDED | REG_NOSUB)) {
ao2_t_ref(new_filter, -1, "failed to make regex");
return FILTER_COMPILE_FAIL;
}
if (is_blackfilter) {
ao2_t_link(blackfilters, new_filter, "link new filter into black user container");
} else {
ao2_t_link(whitefilters, new_filter, "link new filter into white user container");
}
ao2_ref(new_filter, -1);
return FILTER_SUCCESS;
}
static int match_filter(struct mansession *s, char *eventdata)
{
int result = 0;
if (manager_debug) {
ast_verbose("<-- Examining AMI event: -->\n%s\n", eventdata);
} else {
ast_debug(4, "Examining AMI event:\n%s\n", eventdata);
}
if (!ao2_container_count(s->session->whitefilters) && !ao2_container_count(s->session->blackfilters)) {
return 1; /* no filtering means match all */
} else if (ao2_container_count(s->session->whitefilters) && !ao2_container_count(s->session->blackfilters)) {
/* white filters only: implied black all filter processed first, then white filters */
ao2_t_callback_data(s->session->whitefilters, OBJ_NODATA, whitefilter_cmp_fn, eventdata, &result, "find filter in session filter container");
} else if (!ao2_container_count(s->session->whitefilters) && ao2_container_count(s->session->blackfilters)) {
/* black filters only: implied white all filter processed first, then black filters */
ao2_t_callback_data(s->session->blackfilters, OBJ_NODATA, blackfilter_cmp_fn, eventdata, &result, "find filter in session filter container");
} else {
/* white and black filters: implied black all filter processed first, then white filters, and lastly black filters */
ao2_t_callback_data(s->session->whitefilters, OBJ_NODATA, whitefilter_cmp_fn, eventdata, &result, "find filter in session filter container");
if (result) {
result = 0;
ao2_t_callback_data(s->session->blackfilters, OBJ_NODATA, blackfilter_cmp_fn, eventdata, &result, "find filter in session filter container");
}
}
return result;
}
/*!
* Send any applicable events to the client listening on this socket.
* Wait only for a finite time on each event, and drop all events whether
* they are successfully sent or not.
*/
static int process_events(struct mansession *s)
{
int ret = 0;
ao2_lock(s->session);
if (s->session->stream != NULL) {
struct eventqent *eqe = s->session->last_ev;
while ((eqe = advance_event(eqe))) {
if (eqe->category == EVENT_FLAG_SHUTDOWN) {
ast_debug(3, "Received CloseSession event\n");
ret = -1;
}
if (!ret && s->session->authenticated &&
(s->session->readperm & eqe->category) == eqe->category &&
(s->session->send_events & eqe->category) == eqe->category) {
if (match_filter(s, eqe->eventdata)) {
if (send_string(s, eqe->eventdata) < 0 || s->write_error)
ret = -1; /* don't send more */
}
}
s->session->last_ev = eqe;
}
}
ao2_unlock(s->session);
return ret;
}
static int action_userevent(struct mansession *s, const struct message *m)
{
const char *event = astman_get_header(m, "UserEvent");
struct ast_str *body = ast_str_thread_get(&userevent_buf, 16);
int x;
ast_str_reset(body);
for (x = 0; x < m->hdrcount; x++) {
if (strncasecmp("UserEvent:", m->headers[x], strlen("UserEvent:")) &&
strncasecmp("Action:", m->headers[x], strlen("Action:"))) {
ast_str_append(&body, 0, "%s\r\n", m->headers[x]);
}
}
astman_send_ack(s, m, "Event Sent");
manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", event, ast_str_buffer(body));
return 0;
}
/*! \brief Show PBX core settings information */
static int action_coresettings(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
char idText[150];
if (!ast_strlen_zero(actionid)) {
snprintf(idText, sizeof(idText), "ActionID: %s\r\n", actionid);
} else {
idText[0] = '\0';
}
astman_append(s, "Response: Success\r\n"
"%s"
"AMIversion: %s\r\n"
"AsteriskVersion: %s\r\n"
"SystemName: %s\r\n"
"CoreMaxCalls: %d\r\n"
"CoreMaxLoadAvg: %f\r\n"
"CoreRunUser: %s\r\n"
"CoreRunGroup: %s\r\n"
"CoreMaxFilehandles: %d\r\n"
"CoreRealTimeEnabled: %s\r\n"
"CoreCDRenabled: %s\r\n"
"CoreHTTPenabled: %s\r\n"
"SoundsSearchCustomDir: %s\r\n"
"\r\n",
idText,
AMI_VERSION,
ast_get_version(),
ast_config_AST_SYSTEM_NAME,
ast_option_maxcalls,
ast_option_maxload,
ast_config_AST_RUN_USER,
ast_config_AST_RUN_GROUP,
ast_option_maxfiles,
AST_CLI_YESNO(ast_realtime_enabled()),
Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
AST_CLI_YESNO(ast_cdr_is_enabled()),
AST_CLI_YESNO(ast_webmanager_check_enabled()),
AST_CLI_YESNO(ast_opt_sounds_search_custom)
);
return 0;
}
/*! \brief Show PBX core status information */
static int action_corestatus(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
char idText[150];
char startuptime[150], startupdate[150];
char reloadtime[150], reloaddate[150];
struct ast_tm tm;
if (!ast_strlen_zero(actionid)) {
snprintf(idText, sizeof(idText), "ActionID: %s\r\n", actionid);
} else {
idText[0] = '\0';
}
ast_localtime(&ast_startuptime, &tm, NULL);
ast_strftime(startuptime, sizeof(startuptime), "%H:%M:%S", &tm);
ast_strftime(startupdate, sizeof(startupdate), "%Y-%m-%d", &tm);
ast_localtime(&ast_lastreloadtime, &tm, NULL);
ast_strftime(reloadtime, sizeof(reloadtime), "%H:%M:%S", &tm);
ast_strftime(reloaddate, sizeof(reloaddate), "%Y-%m-%d", &tm);
astman_append(s, "Response: Success\r\n"
"%s"
"CoreStartupDate: %s\r\n"
"CoreStartupTime: %s\r\n"
"CoreReloadDate: %s\r\n"
"CoreReloadTime: %s\r\n"
"CoreCurrentCalls: %d\r\n"
"\r\n",
idText,
startupdate,
startuptime,
reloaddate,
reloadtime,
ast_active_channels()
);
return 0;
}
/*! \brief Send a reload event */
static int action_reload(struct mansession *s, const struct message *m)
{
const char *module = astman_get_header(m, "Module");
enum ast_module_reload_result res = ast_module_reload(S_OR(module, NULL));
switch (res) {
case AST_MODULE_RELOAD_NOT_FOUND:
astman_send_error(s, m, "No such module");
break;
case AST_MODULE_RELOAD_NOT_IMPLEMENTED:
astman_send_error(s, m, "Module does not support reload");
break;
case AST_MODULE_RELOAD_ERROR:
astman_send_error(s, m, "An unknown error occurred");
break;
case AST_MODULE_RELOAD_IN_PROGRESS:
astman_send_error(s, m, "A reload is in progress");
break;
case AST_MODULE_RELOAD_UNINITIALIZED:
astman_send_error(s, m, "Module not initialized");
break;
case AST_MODULE_RELOAD_QUEUED:
case AST_MODULE_RELOAD_SUCCESS:
/* Treat a queued request as success */
astman_send_ack(s, m, "Module Reloaded");
break;
}
return 0;
}
/*! \brief Manager command "CoreShowChannels" - List currently defined channels
* and some information about them. */
static int action_coreshowchannels(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
char idText[256];
int numchans = 0;
struct ao2_container *channels;
struct ao2_iterator it_chans;
struct ast_channel_snapshot *cs;
if (!ast_strlen_zero(actionid)) {
snprintf(idText, sizeof(idText), "ActionID: %s\r\n", actionid);
} else {
idText[0] = '\0';
}
channels = ast_channel_cache_by_name();
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
astman_send_listack(s, m, "Channels will follow", "start");
it_chans = ao2_iterator_init(channels, 0);
for (; (cs = ao2_iterator_next(&it_chans)); ao2_ref(cs, -1)) {
struct ast_str *built = ast_manager_build_channel_state_string_prefix(cs, "");
char durbuf[16] = "";
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 (!built) {
continue;
}
if (!ast_tvzero(cs->base->creationtime)) {
int duration, durh, durm, durs;
duration = (int)(ast_tvdiff_ms(ast_tvnow(), cs->base->creationtime) / 1000);
durh = duration / 3600;
durm = (duration % 3600) / 60;
durs = duration % 60;
snprintf(durbuf, sizeof(durbuf), "%02d:%02d:%02d", durh, durm, durs);
}
astman_append(s,
"Event: CoreShowChannel\r\n"
"%s"
"%s"
"Application: %s\r\n"
"ApplicationData: %s\r\n"
"Duration: %s\r\n"
"BridgeId: %s\r\n"
"\r\n",
idText,
ast_str_buffer(built),
cs->dialplan->appl,
cs->dialplan->data,
durbuf,
cs->bridge->id);
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
numchans++;
ast_free(built);
}
ao2_iterator_destroy(&it_chans);
astman_send_list_complete(s, m, "CoreShowChannelsComplete", numchans);
ao2_ref(channels, -1);
return 0;
}
/*! \brief Helper function to add a channel name to the vector */
static int coreshowchannelmap_add_to_map(struct ao2_container *c, const char *s)
{
char *str;
str = ast_strdup(s);
if (!str) {
ast_log(LOG_ERROR, "Unable to append channel to channel map\n");
return 1;
}
/* If this is a duplicate, it will be ignored */
ast_str_container_add(c, str);
return 0;
}
/*! \brief Recursive function to get all channels in a bridge. Follow local channels as well */
static int coreshowchannelmap_add_connected_channels(struct ao2_container *channel_map,
struct ast_channel_snapshot *channel_snapshot, struct ast_bridge_snapshot *bridge_snapshot)
{
int res = 0;
struct ao2_iterator iter;
char *current_channel_uid;
iter = ao2_iterator_init(bridge_snapshot->channels, 0);
while ((current_channel_uid = ao2_iterator_next(&iter))) {
struct ast_channel_snapshot *current_channel_snapshot;
int add_channel_res;
/* Don't add the original channel to the list - it's either already in there,
* or it's the channel we want the map for */
if (!strcmp(current_channel_uid, channel_snapshot->base->uniqueid)) {
ao2_ref(current_channel_uid, -1);
continue;
}
current_channel_snapshot = ast_channel_snapshot_get_latest(current_channel_uid);
if (!current_channel_snapshot) {
ast_debug(5, "Unable to get channel snapshot\n");
ao2_ref(current_channel_uid, -1);
continue;
}
add_channel_res = coreshowchannelmap_add_to_map(channel_map, current_channel_snapshot->base->name);
if (add_channel_res) {
res = 1;
ao2_ref(current_channel_snapshot, -1);
ao2_ref(current_channel_uid, -1);
break;
}
/* If this is a local channel that we haven't seen yet, let's go ahead and find out what else is connected to it */
if (ast_begins_with(current_channel_snapshot->base->name, "Local")) {
struct ast_channel_snapshot *other_local_snapshot;
struct ast_bridge_snapshot *other_bridge_snapshot;
int size = strlen(current_channel_snapshot->base->name);
char other_local[size + 1];
/* Don't copy the trailing number - set it to 1 or 2, whichever one it currently is not */
ast_copy_string(other_local, current_channel_snapshot->base->name, size);
other_local[size - 1] = ast_ends_with(current_channel_snapshot->base->name, "1") ? '2' : '1';
other_local[size] = '\0';
other_local_snapshot = ast_channel_snapshot_get_latest_by_name(other_local);
if (!other_local_snapshot) {
ast_debug(5, "Unable to get other local channel snapshot\n");
ao2_ref(current_channel_snapshot, -1);
ao2_ref(current_channel_uid, -1);
continue;
}
if (coreshowchannelmap_add_to_map(channel_map, other_local_snapshot->base->name)) {
res = 1;
ao2_ref(current_channel_snapshot, -1);
ao2_ref(current_channel_uid, -1);
ao2_ref(other_local_snapshot, -1);
break;
}
other_bridge_snapshot = ast_bridge_get_snapshot_by_uniqueid(other_local_snapshot->bridge->id);
if (other_bridge_snapshot) {
res = coreshowchannelmap_add_connected_channels(channel_map, other_local_snapshot, other_bridge_snapshot);
}
ao2_ref(current_channel_snapshot, -1);
ao2_ref(current_channel_uid, -1);
ao2_ref(other_local_snapshot, -1);
ao2_ref(other_bridge_snapshot, -1);
if (res) {
break;
}
}
}
ao2_iterator_destroy(&iter);
return res;
}
/*! \brief Manager command "CoreShowChannelMap" - Lists all channels connected to
* the specified channel. */
static int action_coreshowchannelmap(struct mansession *s, const struct message *m)
{
const char *actionid = astman_get_header(m, "ActionID");
const char *channel_name = astman_get_header(m, "Channel");
char *current_channel_name;
char id_text[256];
int total = 0;
struct ao2_container *channel_map;
struct ao2_iterator i;
RAII_VAR(struct ast_bridge_snapshot *, bridge_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, channel_snapshot, NULL, ao2_cleanup);
if (!ast_strlen_zero(actionid)) {
snprintf(id_text, sizeof(id_text), "ActionID: %s\r\n", actionid);
} else {
id_text[0] = '\0';
}
if (ast_strlen_zero(channel_name)) {
astman_send_error(s, m, "CoreShowChannelMap requires a channel.\n");
return 0;
}
channel_snapshot = ast_channel_snapshot_get_latest_by_name(channel_name);
if (!channel_snapshot) {
astman_send_error(s, m, "Could not get channel snapshot\n");
return 0;
}
bridge_snapshot = ast_bridge_get_snapshot_by_uniqueid(channel_snapshot->bridge->id);
if (!bridge_snapshot) {
astman_send_listack(s, m, "Channel map will follow", "start");
astman_send_list_complete_start(s, m, "CoreShowChannelMapComplete", 0);
astman_send_list_complete_end(s);
return 0;
}
channel_map = ast_str_container_alloc_options(AO2_ALLOC_OPT_LOCK_NOLOCK | AO2_CONTAINER_ALLOC_OPT_DUPS_OBJ_REJECT, 1);
if (!channel_map) {
astman_send_error(s, m, "Could not create channel map\n");
return 0;
}
astman_send_listack(s, m, "Channel map will follow", "start");
if (coreshowchannelmap_add_connected_channels(channel_map, channel_snapshot, bridge_snapshot)) {
astman_send_error(s, m, "Could not complete channel map\n");
ao2_ref(channel_map, -1);
return 0;
}
i = ao2_iterator_init(channel_map, 0);
while ((current_channel_name = ao2_iterator_next(&i))) {
astman_append(s,
"Event: CoreShowChannelMap\r\n"
"%s"
"Channel: %s\r\n"
"ConnectedChannel: %s\r\n\n",
id_text,
channel_name,
current_channel_name);
total++;
}
ao2_iterator_destroy(&i);
ao2_ref(channel_map, -1);
astman_send_list_complete_start(s, m, "CoreShowChannelMapComplete", total);
astman_send_list_complete_end(s);
return 0;
}
/*! \brief Manager command "LoggerRotate" - reloads and rotates the logger in
* the same manner as the CLI command 'logger rotate'. */
static int action_loggerrotate(struct mansession *s, const struct message *m)
{
if (ast_logger_rotate()) {
astman_send_error(s, m, "Failed to reload the logger and rotate log files");
return 0;
}
astman_send_ack(s, m, "Reloaded the logger and rotated log files");
return 0;
}
/*! \brief Manager function to check if module is loaded */
static int manager_modulecheck(struct mansession *s, const struct message *m)
{
const char *module = astman_get_header(m, "Module");
const char *id = astman_get_header(m, "ActionID");
ast_debug(1, "**** ModuleCheck .so file %s\n", module);
if (!ast_module_check(module)) {
astman_send_error(s, m, "Module not loaded");
return 0;
}
astman_append(s, "Response: Success\r\n");
if (!ast_strlen_zero(id)) {
astman_append(s, "ActionID: %s\r\n", id);
}
#if !defined(LOW_MEMORY)
/* When we switched from subversion to git we lost the ability to
* retrieve the 'ASTERISK_FILE_VERSION' from that file, but we retain
* the response header here for backwards compatibility. */
astman_append(s, "Version: \r\n");
#endif
astman_append(s, "\r\n");
return 0;
}
static int manager_moduleload(struct mansession *s, const struct message *m)
{
int res;
const char *module = astman_get_header(m, "Module");
const char *loadtype = astman_get_header(m, "LoadType");
if (!loadtype || strlen(loadtype) == 0) {
astman_send_error(s, m, "Incomplete ModuleLoad action.");
}
if ((!module || strlen(module) == 0) && strcasecmp(loadtype, "reload") != 0) {
astman_send_error(s, m, "Need module name");
}
if (!strcasecmp(loadtype, "load")) {
res = ast_load_resource(module);
if (res) {
astman_send_error(s, m, "Could not load module.");
} else {
astman_send_ack(s, m, "Module loaded.");
}
} else if (!strcasecmp(loadtype, "unload")) {
res = ast_unload_resource(module, AST_FORCE_SOFT);
if (res) {
astman_send_error(s, m, "Could not unload module.");
} else {
astman_send_ack(s, m, "Module unloaded.");
}
} else if (!strcasecmp(loadtype, "reload")) {
/* TODO: Unify the ack/error messages here with action_reload */
if (!ast_strlen_zero(module)) {
enum ast_module_reload_result reload_res = ast_module_reload(module);
switch (reload_res) {
case AST_MODULE_RELOAD_NOT_FOUND:
astman_send_error(s, m, "No such module.");
break;
case AST_MODULE_RELOAD_NOT_IMPLEMENTED:
astman_send_error(s, m, "Module does not support reload action.");
break;
case AST_MODULE_RELOAD_ERROR:
astman_send_error(s, m, "An unknown error occurred");
break;
case AST_MODULE_RELOAD_IN_PROGRESS:
astman_send_error(s, m, "A reload is in progress");
break;
case AST_MODULE_RELOAD_UNINITIALIZED:
astman_send_error(s, m, "Module not initialized");
break;
case AST_MODULE_RELOAD_QUEUED:
case AST_MODULE_RELOAD_SUCCESS:
/* Treat a queued request as success */
astman_send_ack(s, m, "Module reloaded.");
break;
}
} else {
ast_module_reload(NULL); /* Reload all modules */
astman_send_ack(s, m, "All modules reloaded");
}
} else
astman_send_error(s, m, "Incomplete ModuleLoad action.");
return 0;
}
static void log_action(const struct message *m, const char *action)
{
struct ast_str *buf;
int x;
if (!manager_debug) {
return;
}
buf = ast_str_create(256);
if (!buf) {
return;
}
for (x = 0; x < m->hdrcount; ++x) {
if (!strncasecmp(m->headers[x], "Secret", 6)) {
ast_str_append(&buf, 0, "Secret: <redacted from logging>\n");
} else {
ast_str_append(&buf, 0, "%s\n", m->headers[x]);
}
}
ast_verbose("<--- Examining AMI action: -->\n%s\n", ast_str_buffer(buf));
ast_free(buf);
}
/*
* Done with the action handlers here, we start with the code in charge
* of accepting connections and serving them.
* accept_thread() forks a new thread for each connection, session_do(),
* which in turn calls get_input() repeatedly until a full message has
* been accumulated, and then invokes process_message() to pass it to
* the appropriate handler.
*/
/*! \brief
* Process an AMI message, performing desired action.
* Return 0 on success, -1 on error that require the session to be destroyed.
*/
static int process_message(struct mansession *s, const struct message *m)
{
int ret = 0;
struct manager_action *act_found;
struct ast_manager_user *user = NULL;
const char *username;
const char *action;
action = __astman_get_header(m, "Action", GET_HEADER_SKIP_EMPTY);
if (ast_strlen_zero(action)) {
report_req_bad_format(s, "NONE");
mansession_lock(s);
astman_send_error(s, m, "Missing action in request");
mansession_unlock(s);
return 0;
}
log_action(m, action);
if (ast_shutting_down()) {
ast_log(LOG_ERROR, "Unable to process manager action '%s'. Asterisk is shutting down.\n", action);
mansession_lock(s);
astman_send_error(s, m, "Asterisk is shutting down");
mansession_unlock(s);
return 0;
}
if (!s->session->authenticated
&& strcasecmp(action, "Login")
&& strcasecmp(action, "Logoff")
&& strcasecmp(action, "Challenge")) {
if (!s->session->authenticated) {
report_req_not_allowed(s, action);
}
mansession_lock(s);
astman_send_error(s, m, "Permission denied");
mansession_unlock(s);
return 0;
}
if (!s->session->authenticated
&& (!strcasecmp(action, "Login")
|| !strcasecmp(action, "Challenge"))) {
username = astman_get_header(m, "Username");
if (!ast_strlen_zero(username) && check_manager_session_inuse(username)) {
AST_RWLIST_WRLOCK(&users);
user = get_manager_by_name_locked(username);
if (user && !user->allowmultiplelogin) {
AST_RWLIST_UNLOCK(&users);
report_session_limit(s);
sleep(1);
mansession_lock(s);
astman_send_error(s, m, "Login Already In Use");
mansession_unlock(s);
return -1;
}
AST_RWLIST_UNLOCK(&users);
}
}
act_found = action_find(action);
if (act_found) {
/* Found the requested AMI action. */
int acted = 0;
if ((s->session->writeperm & act_found->authority)
|| act_found->authority == 0) {
/* We have the authority to execute the action. */
ret = -1;
ao2_lock(act_found);
if (act_found->registered && act_found->func) {
struct ast_module *mod_ref = ast_module_running_ref(act_found->module);
ao2_unlock(act_found);
if (mod_ref || !act_found->module) {
ast_debug(1, "Running action '%s'\n", act_found->action);
ret = act_found->func(s, m);
acted = 1;
ast_module_unref(mod_ref);
}
} else {
ao2_unlock(act_found);
}
}
if (!acted) {
/*
* We did not execute the action because access was denied, it
* was no longer registered, or no action was really registered.
* Complain about it and leave.
*/
report_req_not_allowed(s, action);
mansession_lock(s);
astman_send_error(s, m, "Permission denied");
mansession_unlock(s);
}
ao2_t_ref(act_found, -1, "done with found action object");
} else {
char buf[512];
report_req_bad_format(s, action);
snprintf(buf, sizeof(buf), "Invalid/unknown command: %s. Use Action: ListCommands to show available commands.", action);
mansession_lock(s);
astman_send_error(s, m, buf);
mansession_unlock(s);
}
if (ret) {
return ret;
}
/* Once done with our message, deliver any pending events unless the
requester doesn't want them as part of this response.
*/
if (ast_strlen_zero(astman_get_header(m, "SuppressEvents"))) {
return process_events(s);
} else {
return ret;
}
}
/*!
* Read one full line (including crlf) from the manager socket.
* \note \verbatim
* \r\n is the only valid terminator for the line.
* (Note that, later, '\0' will be considered as the end-of-line marker,
* so everything between the '\0' and the '\r\n' will not be used).
* Also note that we assume output to have at least "maxlen" space.
* \endverbatim
*/
static int get_input(struct mansession *s, char *output)
{
int res, x;
int maxlen = sizeof(s->session->inbuf) - 1;
char *src = s->session->inbuf;
int timeout = -1;
time_t now;
/*
* Look for \r\n within the buffer. If found, copy to the output
* buffer and return, trimming the \r\n (not used afterwards).
*/
for (x = 0; x < s->session->inlen; x++) {
int cr; /* set if we have \r */
if (src[x] == '\r' && x+1 < s->session->inlen && src[x + 1] == '\n') {
cr = 2; /* Found. Update length to include \r\n */
} else if (src[x] == '\n') {
cr = 1; /* also accept \n only */
} else {
continue;
}
memmove(output, src, x); /*... but trim \r\n */
output[x] = '\0'; /* terminate the string */
x += cr; /* number of bytes used */
s->session->inlen -= x; /* remaining size */
memmove(src, src + x, s->session->inlen); /* remove used bytes */
return 1;
}
if (s->session->inlen >= maxlen) {
/* no crlf found, and buffer full - sorry, too long for us
* keep the last character in case we are in the middle of a CRLF. */
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
ast_log(LOG_WARNING, "Discarding message from %s. Line too long: %.25s...\n", ast_sockaddr_stringify_addr(&s->session->addr), src);
src[0] = src[s->session->inlen - 1];
s->session->inlen = 1;
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
s->parsing = MESSAGE_LINE_TOO_LONG;
}
res = 0;
while (res == 0) {
/* calculate a timeout if we are not authenticated */
if (!s->session->authenticated) {
if(time(&now) == -1) {
ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
return -1;
}
timeout = (authtimeout - (now - s->session->authstart)) * 1000;
if (timeout < 0) {
/* we have timed out */
return 0;
}
}
ast_mutex_lock(&s->session->notify_lock);
if (s->session->pending_event) {
s->session->pending_event = 0;
ast_mutex_unlock(&s->session->notify_lock);
return 0;
}
s->session->waiting_thread = pthread_self();
ast_mutex_unlock(&s->session->notify_lock);
res = ast_wait_for_input(ast_iostream_get_fd(s->session->stream), timeout);
ast_mutex_lock(&s->session->notify_lock);
s->session->waiting_thread = AST_PTHREADT_NULL;
ast_mutex_unlock(&s->session->notify_lock);
}
if (res < 0) {
/* If we get a signal from some other thread (typically because
* there are new events queued), return 0 to notify the caller.
*/
if (errno == EINTR || errno == EAGAIN) {
return 0;
}
ast_log(LOG_WARNING, "poll() returned error: %s\n", strerror(errno));
return -1;
}
ao2_lock(s->session);
res = ast_iostream_read(s->session->stream, src + s->session->inlen, maxlen - s->session->inlen);
if (res < 1) {
res = -1; /* error return */
} else {
s->session->inlen += res;
src[s->session->inlen] = '\0';
res = 0;
}
ao2_unlock(s->session);
return res;
}
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
/*!
* \internal
* \brief Error handling for sending parse errors. This function handles locking, and clearing the
* parse error flag.
*
* \param s AMI session to process action request.
* \param m Message that's in error.
* \param error Error message to send.
*/
static void handle_parse_error(struct mansession *s, struct message *m, char *error)
{
mansession_lock(s);
astman_send_error(s, m, error);
s->parsing = MESSAGE_OKAY;
mansession_unlock(s);
}
/*!
* \internal
* \brief Read and process an AMI action request.
*
* \param s AMI session to process action request.
*
* \retval 0 Retain AMI connection for next command.
* \retval -1 Drop AMI connection due to logoff or connection error.
*/
static int do_message(struct mansession *s)
{
struct message m = { 0 };
char header_buf[sizeof(s->session->inbuf)] = { '\0' };
int res;
int hdr_loss;
time_t now;
hdr_loss = 0;
for (;;) {
/* Check if any events are pending and do them if needed */
if (process_events(s)) {
res = -1;
break;
}
res = get_input(s, header_buf);
if (res == 0) {
/* No input line received. */
if (!s->session->authenticated) {
if (time(&now) == -1) {
ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
res = -1;
break;
}
if (now - s->session->authstart > authtimeout) {
if (displayconnects) {
ast_verb(2, "Client from %s, failed to authenticate in %d seconds\n", ast_sockaddr_stringify_addr(&s->session->addr), authtimeout);
}
res = -1;
break;
}
}
continue;
} else if (res > 0) {
/* Input line received. */
if (ast_strlen_zero(header_buf)) {
if (hdr_loss) {
mansession_lock(s);
astman_send_error(s, &m, "Too many lines in message or allocation failure");
mansession_unlock(s);
res = 0;
} else {
Multiple revisions 374570,374581 ........ r374570 | dlee | 2012-10-05 15:14:41 -0500 (Fri, 05 Oct 2012) | 22 lines Improve AMI long line error handling In AMI's parser, when it receives a long line (> 1024 characters), it discards that line, but continues to process the message normally. Typically, this is not a problem because a) who has lines that long and b) usually a discarded line results in an invalid message. But if that line is specifying an optional field, then the message will be processed, you get a 'Response: Success', but things don't work the way you expected them to. This patch changes the behavior when a line-too-long parse error occurs. * Changes the log message to avoid way-too-long (and truncated anyways) log messages * Adds a 'parsing' status flag to Response: Success * Sets parsing = MESSAGE_LINE_TOO_LONG if, well, a line is too long * Responds with an appropriate error if parsing != MESSAGE_OKAY (closes issue AST-961) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2142/ ........ r374581 | dlee | 2012-10-05 15:20:28 -0500 (Fri, 05 Oct 2012) | 1 line I've committed too much. Reverting part of r374570. ........ Merged revisions 374570,374581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374586 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 374587 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-05 20:33:56 +00:00
switch (s->parsing) {
case MESSAGE_OKAY:
res = process_message(s, &m) ? -1 : 0;
break;
case MESSAGE_LINE_TOO_LONG:
handle_parse_error(s, &m, "Failed to parse message: line too long");
res = 0;
break;
}
}
break;
} else if (m.hdrcount < ARRAY_LEN(m.headers)) {
m.headers[m.hdrcount] = ast_strdup(header_buf);
if (!m.headers[m.hdrcount]) {
/* Allocation failure. */
hdr_loss = 1;
} else {
++m.hdrcount;
}
} else {
/* Too many lines in message. */
hdr_loss = 1;
}
} else {
/* Input error. */
break;
}
}
astman_free_headers(&m);
return res;
}
/*! \brief The body of the individual manager session.
* Call get_input() to read one line at a time
* (or be woken up on new events), collect the lines in a
* message until found an empty line, and execute the request.
* In any case, deliver events asynchronously through process_events()
* (called from here if no line is available, or at the end of
* process_message(). )
*/
static void *session_do(void *data)
{
struct ast_tcptls_session_instance *ser = data;
struct mansession_session *session;
struct mansession s = {
.tcptls_session = data,
};
int res;
int arg = 1;
struct ast_sockaddr ser_remote_address_tmp;
if (ast_atomic_fetchadd_int(&unauth_sessions, +1) >= authlimit) {
ast_atomic_fetchadd_int(&unauth_sessions, -1);
goto done;
}
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
session = build_mansession(&ser_remote_address_tmp);
if (session == NULL) {
ast_atomic_fetchadd_int(&unauth_sessions, -1);
goto done;
}
/* here we set TCP_NODELAY on the socket to disable Nagle's algorithm.
* This is necessary to prevent delays (caused by buffering) as we
* write to the socket in bits and pieces. */
if (setsockopt(ast_iostream_get_fd(ser->stream), IPPROTO_TCP, TCP_NODELAY, (char *) &arg, sizeof(arg)) < 0) {
ast_log(LOG_WARNING, "Failed to set TCP_NODELAY on manager connection: %s\n", strerror(errno));
}
ast_iostream_nonblock(ser->stream);
ao2_lock(session);
/* Hook to the tail of the event queue */
session->last_ev = grab_last();
ast_mutex_init(&s.lock);
/* these fields duplicate those in the 'ser' structure */
session->stream = s.stream = ser->stream;
ast_sockaddr_copy(&session->addr, &ser_remote_address_tmp);
s.session = session;
AST_LIST_HEAD_INIT_NOLOCK(&session->datastores);
if(time(&session->authstart) == -1) {
ast_log(LOG_ERROR, "error executing time(): %s; disconnecting client\n", strerror(errno));
ast_atomic_fetchadd_int(&unauth_sessions, -1);
ao2_unlock(session);
session_destroy(session);
goto done;
}
ao2_unlock(session);
/*
* We cannot let the stream exclusively wait for data to arrive.
* We have to wake up the task to send async events.
*/
ast_iostream_set_exclusive_input(ser->stream, 0);
ast_iostream_set_timeout_sequence(ser->stream,
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
ast_tvnow(), authtimeout * 1000);
astman_append(&s, "Asterisk Call Manager/%s\r\n", AMI_VERSION); /* welcome prompt */
for (;;) {
if ((res = do_message(&s)) < 0 || s.write_error) {
break;
}
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
if (session->authenticated) {
ast_iostream_set_timeout_disable(ser->stream);
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
}
}
/* session is over, explain why and terminate */
if (session->authenticated) {
if (manager_displayconnects(session)) {
ast_verb(2, "Manager '%s' logged off from %s\n", session->username, ast_sockaddr_stringify_addr(&session->addr));
}
} else {
ast_atomic_fetchadd_int(&unauth_sessions, -1);
if (displayconnects) {
ast_verb(2, "Connect attempt from '%s' unable to authenticate\n", ast_sockaddr_stringify_addr(&session->addr));
}
}
session_destroy(session);
ast_mutex_destroy(&s.lock);
done:
ao2_ref(ser, -1);
ser = NULL;
return NULL;
}
/*! \brief remove at most n_max stale session from the list. */
static int purge_sessions(int n_max)
{
struct ao2_container *sessions;
struct mansession_session *session;
time_t now = time(NULL);
struct ao2_iterator i;
int purged = 0;
sessions = ao2_global_obj_ref(mgr_sessions);
if (!sessions) {
return 0;
}
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i)) && n_max > 0) {
ao2_lock(session);
if (session->sessiontimeout && (now > session->sessiontimeout) && !session->inuse) {
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
if (session->authenticated
&& VERBOSITY_ATLEAST(2)
&& manager_displayconnects(session)) {
ast_verb(2, "HTTP Manager '%s' timed out from %s\n",
session->username, ast_sockaddr_stringify_addr(&session->addr));
}
ao2_unlock(session);
session_destroy(session);
n_max--;
purged++;
} else {
ao2_unlock(session);
unref_mansession(session);
}
}
ao2_iterator_destroy(&i);
return purged;
}
/*! \brief
* events are appended to a queue from where they
* can be dispatched to clients.
*/
static int append_event(const char *str, int category)
{
struct eventqent *tmp = ast_malloc(sizeof(*tmp) + strlen(str));
static int seq; /* sequence number */
if (!tmp) {
return -1;
}
/* need to init all fields, because ast_malloc() does not */
tmp->usecount = 0;
tmp->category = category;
tmp->seq = ast_atomic_fetchadd_int(&seq, 1);
tmp->tv = ast_tvnow();
AST_RWLIST_NEXT(tmp, eq_next) = NULL;
strcpy(tmp->eventdata, str);
AST_RWLIST_WRLOCK(&all_events);
AST_RWLIST_INSERT_TAIL(&all_events, tmp, eq_next);
AST_RWLIST_UNLOCK(&all_events);
return 0;
}
static void append_channel_vars(struct ast_str **pbuf, struct ast_channel *chan)
{
struct varshead *vars;
struct ast_var_t *var;
vars = ast_channel_get_manager_vars(chan);
if (!vars) {
return;
}
AST_LIST_TRAVERSE(vars, var, entries) {
ast_str_append(pbuf, 0, "ChanVariable(%s): %s=%s\r\n", ast_channel_name(chan), var->name, var->value);
}
ao2_ref(vars, -1);
}
/* XXX see if can be moved inside the function */
AST_THREADSTORAGE(manager_event_buf);
#define MANAGER_EVENT_BUF_INITSIZE 256
static int __attribute__((format(printf, 9, 0))) __manager_event_sessions_va(
struct ao2_container *sessions,
int category,
const char *event,
int chancount,
struct ast_channel **chans,
const char *file,
int line,
const char *func,
const char *fmt,
va_list ap)
{
struct ast_str *auth = ast_str_alloca(MAX_AUTH_PERM_STRING);
const char *cat_str;
struct timeval now;
struct ast_str *buf;
int i;
if (!ast_strlen_zero(manager_disabledevents)) {
if (ast_in_delimited_string(event, manager_disabledevents, ',')) {
ast_debug(3, "AMI Event '%s' is globally disabled, skipping\n", event);
/* Event is globally disabled */
return -1;
}
}
buf = ast_str_thread_get(&manager_event_buf, MANAGER_EVENT_BUF_INITSIZE);
if (!buf) {
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
return -1;
}
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
cat_str = authority_to_str(category, &auth);
ast_str_set(&buf, 0,
"Event: %s\r\n"
"Privilege: %s\r\n",
event, cat_str);
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
if (timestampevents) {
now = ast_tvnow();
ast_str_append(&buf, 0,
"Timestamp: %ld.%06lu\r\n",
(long)now.tv_sec, (unsigned long) now.tv_usec);
}
if (manager_debug) {
static int seq;
ast_str_append(&buf, 0,
"SequenceNumber: %d\r\n",
ast_atomic_fetchadd_int(&seq, 1));
ast_str_append(&buf, 0,
"File: %s\r\n"
"Line: %d\r\n"
"Func: %s\r\n",
file, line, func);
}
if (!ast_strlen_zero(ast_config_AST_SYSTEM_NAME)) {
ast_str_append(&buf, 0,
"SystemName: %s\r\n",
ast_config_AST_SYSTEM_NAME);
}
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
ast_str_append_va(&buf, 0, fmt, ap);
for (i = 0; i < chancount; i++) {
append_channel_vars(&buf, chans[i]);
}
ast_str_append(&buf, 0, "\r\n");
append_event(ast_str_buffer(buf), category);
/* Wake up any sleeping sessions */
if (sessions) {
struct ao2_iterator iter;
struct mansession_session *session;
iter = ao2_iterator_init(sessions, 0);
while ((session = ao2_iterator_next(&iter))) {
ast_mutex_lock(&session->notify_lock);
if (session->waiting_thread != AST_PTHREADT_NULL) {
pthread_kill(session->waiting_thread, SIGURG);
} else {
/* We have an event to process, but the mansession is
* not waiting for it. We still need to indicate that there
* is an event waiting so that get_input processes the pending
* event instead of polling.
*/
session->pending_event = 1;
}
ast_mutex_unlock(&session->notify_lock);
unref_mansession(session);
}
ao2_iterator_destroy(&iter);
}
if (category != EVENT_FLAG_SHUTDOWN && !AST_RWLIST_EMPTY(&manager_hooks)) {
struct manager_custom_hook *hook;
AST_RWLIST_RDLOCK(&manager_hooks);
AST_RWLIST_TRAVERSE(&manager_hooks, hook, list) {
hook->helper(category, event, ast_str_buffer(buf));
}
AST_RWLIST_UNLOCK(&manager_hooks);
}
return 0;
}
static int __attribute__((format(printf, 9, 0))) __manager_event_sessions(
struct ao2_container *sessions,
int category,
const char *event,
int chancount,
struct ast_channel **chans,
const char *file,
int line,
const char *func,
const char *fmt,
...)
{
va_list ap;
int res;
va_start(ap, fmt);
res = __manager_event_sessions_va(sessions, category, event, chancount, chans,
file, line, func, fmt, ap);
va_end(ap);
return res;
}
int __ast_manager_event_multichan(int category, const char *event, int chancount,
struct ast_channel **chans, const char *file, int line, const char *func,
const char *fmt, ...)
{
struct ao2_container *sessions = ao2_global_obj_ref(mgr_sessions);
va_list ap;
int res;
if (!any_manager_listeners(sessions)) {
/* Nobody is listening */
ao2_cleanup(sessions);
return 0;
}
va_start(ap, fmt);
res = __manager_event_sessions_va(sessions, category, event, chancount, chans,
file, line, func, fmt, ap);
va_end(ap);
ao2_cleanup(sessions);
return res;
}
/*! \brief
* support functions to register/unregister AMI action handlers,
*/
int ast_manager_unregister(const char *action)
{
struct manager_action *cur;
AST_RWLIST_WRLOCK(&actions);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&actions, cur, list) {
if (!strcasecmp(action, cur->action)) {
AST_RWLIST_REMOVE_CURRENT(list);
break;
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&actions);
if (cur) {
/*
* We have removed the action object from the container so we
* are no longer in a hurry.
*/
ao2_lock(cur);
cur->registered = 0;
ao2_unlock(cur);
ao2_t_ref(cur, -1, "action object removed from list");
ast_verb(5, "Manager unregistered action %s\n", action);
}
return 0;
}
static int manager_state_cb(const char *context, const char *exten, struct ast_state_cb_info *info, void *data)
{
/* Notify managers of change */
char hint[512];
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
hint[0] = '\0';
ast_get_hint(hint, sizeof(hint), NULL, 0, NULL, context, exten);
switch(info->reason) {
case AST_HINT_UPDATE_DEVICE:
manager_event(EVENT_FLAG_CALL, "ExtensionStatus",
"Exten: %s\r\n"
"Context: %s\r\n"
"Hint: %s\r\n"
"Status: %d\r\n"
"StatusText: %s\r\n",
exten,
context,
hint,
info->exten_state,
ast_extension_state2str(info->exten_state));
break;
case AST_HINT_UPDATE_PRESENCE:
manager_event(EVENT_FLAG_CALL, "PresenceStatus",
"Exten: %s\r\n"
"Context: %s\r\n"
"Hint: %s\r\n"
"Status: %s\r\n"
"Subtype: %s\r\n"
"Message: %s\r\n",
exten,
context,
hint,
ast_presence_state2str(info->presence_state),
info->presence_subtype,
info->presence_message);
break;
}
return 0;
}
static int ast_manager_register_struct(struct manager_action *act)
{
struct manager_action *cur, *prev = NULL;
AST_RWLIST_WRLOCK(&actions);
AST_RWLIST_TRAVERSE(&actions, cur, list) {
int ret;
ret = strcasecmp(cur->action, act->action);
if (ret == 0) {
ast_log(LOG_WARNING, "Manager: Action '%s' already registered\n", act->action);
AST_RWLIST_UNLOCK(&actions);
return -1;
}
if (ret > 0) { /* Insert these alphabetically */
break;
}
prev = cur;
}
ao2_t_ref(act, +1, "action object added to list");
act->registered = 1;
if (prev) {
AST_RWLIST_INSERT_AFTER(&actions, prev, act, list);
} else {
AST_RWLIST_INSERT_HEAD(&actions, act, list);
}
ast_verb(5, "Manager registered action %s\n", act->action);
AST_RWLIST_UNLOCK(&actions);
return 0;
}
/*!
* \internal
* \brief Destroy the registered AMI action object.
*
* \param obj Object to destroy.
*/
static void action_destroy(void *obj)
{
struct manager_action *doomed = obj;
if (doomed->synopsis) {
/* The string fields were initialized. */
ast_string_field_free_memory(doomed);
}
ao2_cleanup(doomed->final_response);
ao2_cleanup(doomed->list_responses);
}
/*! \brief register a new command with manager, including online help. This is
the preferred way to register a manager command */
int ast_manager_register2(const char *action, int auth, int (*func)(struct mansession *s, const struct message *m), struct ast_module *module, const char *synopsis, const char *description)
{
struct manager_action *cur;
cur = ao2_t_alloc(sizeof(*cur), action_destroy, action);
if (!cur) {
return -1;
}
if (ast_string_field_init(cur, 128)) {
ao2_t_ref(cur, -1, "action object creation failed");
return -1;
}
cur->action = action;
cur->authority = auth;
cur->func = func;
cur->module = module;
#ifdef AST_XML_DOCS
if (ast_strlen_zero(synopsis) && ast_strlen_zero(description)) {
char *tmpxml;
tmpxml = ast_xmldoc_build_synopsis("manager", action, NULL);
ast_string_field_set(cur, synopsis, tmpxml);
ast_free(tmpxml);
tmpxml = ast_xmldoc_build_syntax("manager", action, NULL);
ast_string_field_set(cur, syntax, tmpxml);
ast_free(tmpxml);
tmpxml = ast_xmldoc_build_description("manager", action, NULL);
ast_string_field_set(cur, description, tmpxml);
ast_free(tmpxml);
tmpxml = ast_xmldoc_build_seealso("manager", action, NULL);
ast_string_field_set(cur, seealso, tmpxml);
ast_free(tmpxml);
tmpxml = ast_xmldoc_build_arguments("manager", action, NULL);
ast_string_field_set(cur, arguments, tmpxml);
ast_free(tmpxml);
cur->final_response = ast_xmldoc_build_final_response("manager", action, NULL);
cur->list_responses = ast_xmldoc_build_list_responses("manager", action, NULL);
cur->docsrc = AST_XML_DOC;
} else
#endif
{
ast_string_field_set(cur, synopsis, synopsis);
ast_string_field_set(cur, description, description);
#ifdef AST_XML_DOCS
cur->docsrc = AST_STATIC_DOC;
#endif
}
if (ast_manager_register_struct(cur)) {
ao2_t_ref(cur, -1, "action object registration failed");
return -1;
}
ao2_t_ref(cur, -1, "action object registration successful");
return 0;
}
/*! @}
END Doxygen group */
/*
* The following are support functions for AMI-over-http.
* The common entry point is generic_http_callback(),
* which extracts HTTP header and URI fields and reformats
* them into AMI messages, locates a proper session
* (using the mansession_id Cookie or GET variable),
* and calls process_message() as for regular AMI clients.
* When done, the output (which goes to a temporary file)
* is read back into a buffer and reformatted as desired,
* then fed back to the client over the original socket.
*/
enum output_format {
FORMAT_RAW,
FORMAT_HTML,
FORMAT_XML,
};
static const char * const contenttype[] = {
[FORMAT_RAW] = "plain",
[FORMAT_HTML] = "html",
[FORMAT_XML] = "xml",
};
/*!
* locate an http session in the list. The search key (ident) is
* the value of the mansession_id cookie (0 is not valid and means
* a session on the AMI socket).
*/
static struct mansession_session *find_session(uint32_t ident, int incinuse)
{
struct ao2_container *sessions;
struct mansession_session *session;
struct ao2_iterator i;
if (ident == 0) {
return NULL;
}
sessions = ao2_global_obj_ref(mgr_sessions);
if (!sessions) {
return NULL;
}
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i))) {
ao2_lock(session);
if (session->managerid == ident && !session->needdestroy) {
ast_atomic_fetchadd_int(&session->inuse, incinuse ? 1 : 0);
break;
}
ao2_unlock(session);
unref_mansession(session);
}
ao2_iterator_destroy(&i);
return session;
}
/*!
* locate an http session in the list.
* The search keys (nonce) and (username) is value from received
* "Authorization" http header.
* As well as in find_session() function, the value of the nonce can't be zero.
* (0 meansi, that the session used for AMI socket connection).
* Flag (stale) is set, if client used valid, but old, nonce value.
*
*/
static struct mansession_session *find_session_by_nonce(const char *username, unsigned long nonce, int *stale)
{
struct mansession_session *session;
struct ao2_container *sessions;
struct ao2_iterator i;
if (nonce == 0 || username == NULL || stale == NULL) {
return NULL;
}
sessions = ao2_global_obj_ref(mgr_sessions);
if (!sessions) {
return NULL;
}
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i))) {
ao2_lock(session);
if (!strcasecmp(session->username, username) && session->managerid == nonce) {
*stale = 0;
break;
} else if (!strcasecmp(session->username, username) && session->oldnonce == nonce) {
*stale = 1;
break;
}
ao2_unlock(session);
unref_mansession(session);
}
ao2_iterator_destroy(&i);
return session;
}
int astman_is_authed(uint32_t ident)
{
int authed;
struct mansession_session *session;
if (!(session = find_session(ident, 0)))
return 0;
authed = (session->authenticated != 0);
ao2_unlock(session);
unref_mansession(session);
return authed;
}
int astman_verify_session_readpermissions(uint32_t ident, int perm)
{
int result = 0;
struct mansession_session *session;
struct ao2_container *sessions;
struct ao2_iterator i;
if (ident == 0) {
return 0;
}
sessions = ao2_global_obj_ref(mgr_sessions);
if (!sessions) {
return 0;
}
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i))) {
ao2_lock(session);
if ((session->managerid == ident) && (session->readperm & perm)) {
result = 1;
ao2_unlock(session);
unref_mansession(session);
break;
}
ao2_unlock(session);
unref_mansession(session);
}
ao2_iterator_destroy(&i);
return result;
}
int astman_verify_session_writepermissions(uint32_t ident, int perm)
{
int result = 0;
struct mansession_session *session;
struct ao2_container *sessions;
struct ao2_iterator i;
if (ident == 0) {
return 0;
}
sessions = ao2_global_obj_ref(mgr_sessions);
if (!sessions) {
return 0;
}
i = ao2_iterator_init(sessions, 0);
ao2_ref(sessions, -1);
while ((session = ao2_iterator_next(&i))) {
ao2_lock(session);
if ((session->managerid == ident) && (session->writeperm & perm)) {
result = 1;
ao2_unlock(session);
unref_mansession(session);
break;
}
ao2_unlock(session);
unref_mansession(session);
}
ao2_iterator_destroy(&i);
return result;
}
/*
* convert to xml with various conversion:
* mode & 1 -> lowercase;
* mode & 2 -> replace non-alphanumeric chars with underscore
*/
static void xml_copy_escape(struct ast_str **out, const char *src, int mode)
{
/* store in a local buffer to avoid calling ast_str_append too often */
char buf[256];
char *dst = buf;
const char *save = src;
int space = sizeof(buf);
/* repeat until done and nothing to flush */
for ( ; *src || dst != buf ; src++) {
if (*src == '\0' || space < 10) { /* flush */
*dst++ = '\0';
ast_str_append(out, 0, "%s", buf);
dst = buf;
space = sizeof(buf);
if (*src == '\0') {
break;
}
}
if (mode & 2) {
if (save == src && isdigit(*src)) {
/* The first character of an XML attribute cannot be a digit */
*dst++ = '_';
*dst++ = *src;
space -= 2;
continue;
} else if (!isalnum(*src)) {
/* Replace non-alphanumeric with an underscore */
*dst++ = '_';
space--;
continue;
}
}
switch (*src) {
case '<':
strcpy(dst, "&lt;");
dst += 4;
space -= 4;
break;
case '>':
strcpy(dst, "&gt;");
dst += 4;
space -= 4;
break;
case '\"':
strcpy(dst, "&quot;");
dst += 6;
space -= 6;
break;
case '\'':
strcpy(dst, "&apos;");
dst += 6;
space -= 6;
break;
case '&':
strcpy(dst, "&amp;");
dst += 5;
space -= 5;
break;
default:
*dst++ = mode ? tolower(*src) : *src;
space--;
}
}
}
struct variable_count {
char *varname;
int count;
};
static int variable_count_hash_fn(const void *vvc, const int flags)
{
const struct variable_count *vc = vvc;
return ast_str_hash(vc->varname);
}
static int variable_count_cmp_fn(void *obj, void *vstr, int flags)
{
/* Due to the simplicity of struct variable_count, it makes no difference
* if you pass in objects or strings, the same operation applies. This is
* due to the fact that the hash occurs on the first element, which means
* the address of both the struct and the string are exactly the same. */
struct variable_count *vc = obj;
char *str = vstr;
return !strcmp(vc->varname, str) ? CMP_MATCH | CMP_STOP : 0;
}
/*! \brief Convert the input into XML or HTML.
* The input is supposed to be a sequence of lines of the form
* Name: value
* optionally followed by a blob of unformatted text.
* A blank line is a section separator. Basically, this is a
* mixture of the format of Manager Interface and CLI commands.
* The unformatted text is considered as a single value of a field
* named 'Opaque-data'.
*
* At the moment the output format is the following (but it may
* change depending on future requirements so don't count too
* much on it when writing applications):
*
* General: the unformatted text is used as a value of
* XML output: to be completed
*
* \verbatim
* Each section is within <response type="object" id="xxx">
* where xxx is taken from ajaxdest variable or defaults to unknown
* Each row is reported as an attribute Name="value" of an XML
* entity named from the variable ajaxobjtype, default to "generic"
* \endverbatim
*
* HTML output:
* each Name-value pair is output as a single row of a two-column table.
* Sections (blank lines in the input) are separated by a <HR>
*
*/
static void xml_translate(struct ast_str **out, char *in, struct ast_variable *get_vars, enum output_format format)
{
struct ast_variable *v;
const char *dest = NULL;
char *var, *val;
const char *objtype = NULL;
int in_data = 0; /* parsing data */
int inobj = 0;
int xml = (format == FORMAT_XML);
struct variable_count *vc = NULL;
struct ao2_container *vco = NULL;
if (xml) {
/* dest and objtype need only for XML format */
for (v = get_vars; v; v = v->next) {
if (!strcasecmp(v->name, "ajaxdest")) {
dest = v->value;
} else if (!strcasecmp(v->name, "ajaxobjtype")) {
objtype = v->value;
}
}
if (ast_strlen_zero(dest)) {
dest = "unknown";
}
if (ast_strlen_zero(objtype)) {
objtype = "generic";
}
}
/* we want to stop when we find an empty line */
while (in && *in) {
val = strsep(&in, "\r\n"); /* mark start and end of line */
if (in && *in == '\n') { /* remove trailing \n if any */
in++;
}
ast_trim_blanks(val);
ast_debug(5, "inobj %d in_data %d line <%s>\n", inobj, in_data, val);
if (ast_strlen_zero(val)) {
/* empty line */
if (in_data) {
/* close data in Opaque mode */
ast_str_append(out, 0, xml ? "'" : "</td></tr>\n");
in_data = 0;
}
if (inobj) {
/* close block */
ast_str_append(out, 0, xml ? " /></response>\n" :
"<tr><td colspan=\"2\"><hr></td></tr>\r\n");
inobj = 0;
ao2_ref(vco, -1);
vco = NULL;
}
continue;
}
if (!inobj) {
/* start new block */
if (xml) {
ast_str_append(out, 0, "<response type='object' id='%s'><%s", dest, objtype);
}
vco = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0, 37,
variable_count_hash_fn, NULL, variable_count_cmp_fn);
inobj = 1;
}
if (in_data) {
/* Process data field in Opaque mode. This is a
* followup, so we re-add line feeds. */
ast_str_append(out, 0, xml ? "\n" : "<br>\n");
xml_copy_escape(out, val, 0); /* data field */
continue;
}
/* We expect "Name: value" line here */
var = strsep(&val, ":");
if (val) {
/* found the field name */
val = ast_skip_blanks(val);
ast_trim_blanks(var);
} else {
/* field name not found, switch to opaque mode */
val = var;
var = "Opaque-data";
in_data = 1;
}
ast_str_append(out, 0, xml ? " " : "<tr><td>");
if ((vc = ao2_find(vco, var, 0))) {
vc->count++;
} else {
/* Create a new entry for this one */
vc = ao2_alloc(sizeof(*vc), NULL);
vc->varname = var;
vc->count = 1;
ao2_link(vco, vc);
}
xml_copy_escape(out, var, xml ? 1 | 2 : 0); /* data name */
if (vc->count > 1) {
ast_str_append(out, 0, "-%d", vc->count);
}
ao2_ref(vc, -1);
ast_str_append(out, 0, xml ? "='" : "</td><td>");
xml_copy_escape(out, val, 0); /* data field */
if (!in_data || !*in) {
ast_str_append(out, 0, xml ? "'" : "</td></tr>\n");
}
}
if (inobj) {
ast_str_append(out, 0, xml ? " /></response>\n" :
"<tr><td colspan=\"2\"><hr></td></tr>\r\n");
ao2_ref(vco, -1);
}
}
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
static void close_mansession_file(struct mansession *s)
{
if (s->stream) {
ast_iostream_close(s->stream);
s->stream = NULL;
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
} else {
ast_log(LOG_ERROR, "Attempted to close file/file descriptor on mansession without a valid file or file descriptor.\n");
}
}
static void process_output(struct mansession *s, struct ast_str **out, struct ast_variable *params, enum output_format format)
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
{
char *buf;
off_t l;
int fd;
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
if (!s->stream)
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
return;
/* Ensure buffer is NULL-terminated */
ast_iostream_write(s->stream, "", 1);
fd = ast_iostream_get_fd(s->stream);
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
l = lseek(fd, 0, SEEK_CUR);
if (l > 0) {
if (MAP_FAILED == (buf = mmap(NULL, l, PROT_READ | PROT_WRITE, MAP_PRIVATE, fd, 0))) {
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
ast_log(LOG_WARNING, "mmap failed. Manager output was not processed\n");
} else {
if (format == FORMAT_XML || format == FORMAT_HTML) {
xml_translate(out, buf, params, format);
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
} else {
ast_str_append(out, 0, "%s", buf);
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
}
munmap(buf, l);
}
} else if (format == FORMAT_XML || format == FORMAT_HTML) {
xml_translate(out, "", params, format);
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
}
AST-2014-007: Fix DOS by consuming the number of allowed HTTP connections. Simply establishing a TCP connection and never sending anything to the configured HTTP port in http.conf will tie up a HTTP connection. Since there is a maximum number of open HTTP sessions allowed at a time you can block legitimate connections. A similar problem exists if a HTTP request is started but never finished. * Added http.conf session_inactivity timer option to close HTTP connections that aren't doing anything. Defaults to 30000 ms. * Removed the undocumented manager.conf block-sockets option. It interferes with TCP/TLS inactivity timeouts. * AMI and SIP TLS connections now have better authentication timeout protection. Though I didn't remove the bizzare TLS timeout polling code from chan_sip. * chan_sip can now handle SSL certificate renegotiations in the middle of a session. It couldn't do that before because the socket was non-blocking and the SSL calls were not restarted as documented by the OpenSSL documentation. * Fixed an off nominal leak of the ssl struct in handle_tcptls_connection() if the FILE stream failed to open and the SSL certificate negotiations failed. The patch creates a custom FILE stream handler to give the created FILE streams inactivity timeout and timeout after a specific moment in time capability. This approach eliminates the need for code using the FILE stream to be redesigned to deal with the timeouts. This patch indirectly fixes most of ASTERISK-18345 by fixing the usage of the SSL_read/SSL_write operations. ASTERISK-23673 #close Reported by: Richard Mudgett ........ Merged revisions 415841 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415854 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415896 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415907 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12 17:00:08 +00:00
close_mansession_file(s);
Merged revisions 323608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-15 15:33:57 +00:00
}
static int generic_http_callback(struct ast_tcptls_session_instance *ser,
enum ast_http_method method,
enum output_format format,
const struct ast_sockaddr *remote_address, const char *uri,
struct ast_variable *get_params,
struct ast_variable *headers)
{
struct mansession s = { .session = NULL, .tcptls_session = ser };
struct mansession_session *session = NULL;
uint32_t ident;
int fd;
int blastaway = 0;
struct ast_variable *params = get_params;
char template[] = "/tmp/ast-http-XXXXXX"; /* template for temporary file */
struct ast_str *http_header = NULL, *out = NULL;
struct message m = { 0 };
if (method != AST_HTTP_GET && method != AST_HTTP_HEAD && method != AST_HTTP_POST) {
ast_http_error(ser, 501, "Not Implemented", "Attempt to use unimplemented / unsupported method");
return 0;
}
ident = ast_http_manid_from_vars(headers);
if (!(session = find_session(ident, 1))) {
/**/
/* Create new session.
* While it is not in the list we don't need any locking
*/
if (!(session = build_mansession(remote_address))) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (out of memory)");
return 0;
}
ao2_lock(session);
session->send_events = 0;
session->inuse = 1;
/*!
* \note There is approximately a 1 in 1.8E19 chance that the following
* calculation will produce 0, which is an invalid ID, but due to the
* properties of the rand() function (and the constancy of s), that
* won't happen twice in a row.
*/
while ((session->managerid = ast_random() ^ (unsigned long) session) == 0) {
}
session->last_ev = grab_last();
AST_LIST_HEAD_INIT_NOLOCK(&session->datastores);
}
ao2_unlock(session);
http_header = ast_str_create(128);
out = ast_str_create(2048);
ast_mutex_init(&s.lock);
if (http_header == NULL || out == NULL) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (ast_str_create() out of memory)");
goto generic_callback_out;
}
s.session = session;
fd = mkstemp(template); /* create a temporary file for command output */
unlink(template);
if (fd <= -1) {
ast_http_error(ser, 500, "Server Error", "Internal Server Error (mkstemp failed)");
goto generic_callback_out;
}
s.stream = ast_iostream_from_fd(&fd);
if (!s.stream) {
ast_log(LOG_WARNING, "HTTP Manager, fdopen failed: %s!\n", strerror(errno));
ast_http_error(ser, 500, "Server Error", "Internal Server Error (fdopen failed)");
close(fd);
goto generic_callback_out;
}
if (method == AST_HTTP_POST) {
params = ast_http_get_post_vars(ser, headers);
if (!params) {
switch (errno) {
case EFBIG:
ast_http_error(ser, 413, "Request Entity Too Large", "Body too large");
close_mansession_file(&s);
goto generic_callback_out;
case ENOMEM:
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Out of memory");
close_mansession_file(&s);
goto generic_callback_out;
case EIO:
ast_http_error(ser, 400, "Bad Request", "Error parsing request body");
close_mansession_file(&s);
goto generic_callback_out;
}
}
}
astman_append_headers(&m, params);
if (process_message(&s, &m)) {
if (session->authenticated) {
if (manager_displayconnects(session)) {
ast_verb(2, "HTTP Manager '%s' logged off from %s\n", session->username, ast_sockaddr_stringify_addr(&session->addr));
}
} else {
if (displayconnects) {
ast_verb(2, "HTTP Connect attempt from '%s' unable to authenticate\n", ast_sockaddr_stringify_addr(&session->addr));
}
}
session->needdestroy = 1;
}
astman_free_headers(&m);
ast_str_append(&http_header, 0,
"Content-type: text/%s\r\n"
"Set-Cookie: mansession_id=\"%08x\"; Version=1; Max-Age=%d\r\n"
"Pragma: SuppressEvents\r\n",
contenttype[format],
session->managerid, httptimeout);
if (format == FORMAT_XML) {
ast_str_append(&out, 0, "<ajax-response>\n");
} else if (format == FORMAT_HTML) {
/*
* When handling AMI-over-HTTP in HTML format, we provide a simple form for
* debugging purposes. This HTML code should not be here, we
* should read from some config file...
*/
#define ROW_FMT "<tr><td colspan=\"2\" bgcolor=\"#f1f1ff\">%s</td></tr>\r\n"
#define TEST_STRING \
"<form action=\"manager\" method=\"post\">\n\
Action: <select name=\"action\">\n\
<option value=\"\">-----&gt;</option>\n\
<option value=\"login\">login</option>\n\
<option value=\"command\">Command</option>\n\
<option value=\"waitevent\">waitevent</option>\n\
<option value=\"listcommands\">listcommands</option>\n\
</select>\n\
or <input name=\"action\"><br/>\n\
CLI Command <input name=\"command\"><br>\n\
user <input name=\"username\"> pass <input type=\"password\" name=\"secret\"><br>\n\
<input type=\"submit\">\n</form>\n"
ast_str_append(&out, 0, "<title>Asterisk&trade; Manager Interface</title>");
ast_str_append(&out, 0, "<body bgcolor=\"#ffffff\"><table align=center bgcolor=\"#f1f1f1\" width=\"500\">\r\n");
ast_str_append(&out, 0, ROW_FMT, "<h1>Manager Tester</h1>");
ast_str_append(&out, 0, ROW_FMT, TEST_STRING);
}
process_output(&s, &out, params, format);
if (format == FORMAT_XML) {
ast_str_append(&out, 0, "</ajax-response>\n");
} else if (format == FORMAT_HTML) {
ast_str_append(&out, 0, "</table></body>\r\n");
}
ao2_lock(session);
/* Reset HTTP timeout. If we're not authenticated, keep it extremely short */
session->sessiontimeout = time(NULL) + ((session->authenticated || httptimeout < 5) ? httptimeout : 5);
if (session->needdestroy) {
if (session->inuse == 1) {
ast_debug(1, "Need destroy, doing it now!\n");
blastaway = 1;
} else {
ast_debug(1, "Need destroy, but can't do it yet!\n");
ast_mutex_lock(&session->notify_lock);
if (session->waiting_thread != AST_PTHREADT_NULL) {
pthread_kill(session->waiting_thread, SIGURG);
}
ast_mutex_unlock(&session->notify_lock);
session->inuse--;
}
} else {
session->inuse--;
}
ao2_unlock(session);
ast_http_send(ser, method, 200, NULL, http_header, out, 0, 0);
http_header = NULL;
out = NULL;
generic_callback_out:
ast_mutex_destroy(&s.lock);
/* Clear resource */
if (method == AST_HTTP_POST && params) {
ast_variables_destroy(params);
}
ast_free(http_header);
ast_free(out);
if (session) {
if (blastaway) {
session_destroy(session);
} else {
if (session->stream) {
ast_iostream_close(session->stream);
session->stream = NULL;
}
unref_mansession(session);
}
}
return 0;
}
static int auth_http_callback(struct ast_tcptls_session_instance *ser,
enum ast_http_method method,
enum output_format format,
const struct ast_sockaddr *remote_address, const char *uri,
struct ast_variable *get_params,
struct ast_variable *headers)
{
struct mansession_session *session = NULL;
struct mansession s = { .session = NULL, .tcptls_session = ser };
struct ast_variable *v, *params = get_params;
char template[] = "/tmp/ast-http-XXXXXX"; /* template for temporary file */
struct ast_str *http_header = NULL, *out = NULL;
size_t result_size;
struct message m = { 0 };
int fd;
time_t time_now = time(NULL);
unsigned long nonce = 0, nc;
struct ast_http_digest d = { NULL, };
struct ast_manager_user *user = NULL;
int stale = 0;
char resp_hash[256]="";
/* Cache for user data */
char u_username[80];
int u_readperm;
int u_writeperm;
int u_writetimeout;
int u_displayconnects;
if (method != AST_HTTP_GET && method != AST_HTTP_HEAD && method != AST_HTTP_POST) {
ast_http_error(ser, 501, "Not Implemented", "Attempt to use unimplemented / unsupported method");
return 0;
}
/* Find "Authorization: " header */
for (v = headers; v; v = v->next) {
if (!strcasecmp(v->name, "Authorization")) {
break;
}
}
if (!v || ast_strlen_zero(v->value)) {
goto out_401; /* Authorization Header not present - send auth request */
}
/* Digest found - parse */
if (ast_string_field_init(&d, 128)) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (out of memory)");
return 0;
}
if (ast_parse_digest(v->value, &d, 0, 1)) {
/* Error in Digest - send new one */
nonce = 0;
goto out_401;
}
if (sscanf(d.nonce, "%30lx", &nonce) != 1) {
ast_log(LOG_WARNING, "Received incorrect nonce in Digest <%s>\n", d.nonce);
nonce = 0;
goto out_401;
}
AST_RWLIST_WRLOCK(&users);
user = get_manager_by_name_locked(d.username);
if(!user) {
AST_RWLIST_UNLOCK(&users);
ast_log(LOG_NOTICE, "%s tried to authenticate with nonexistent user '%s'\n", ast_sockaddr_stringify_addr(&session->addr), d.username);
nonce = 0;
goto out_401;
}
/* --- We have User for this auth, now check ACL */
if (user->acl && !ast_apply_acl(user->acl, remote_address, "Manager User ACL:")) {
AST_RWLIST_UNLOCK(&users);
ast_log(LOG_NOTICE, "%s failed to pass IP ACL as '%s'\n", ast_sockaddr_stringify_addr(&session->addr), d.username);
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 403, "Permission denied", "Permission denied");
return 0;
}
/* --- We have auth, so check it */
/* compute the expected response to compare with what we received */
{
char *a2;
/* ast_md5_hash outputs 32 characters plus NULL terminator. */
char a2_hash[33];
char resp[256];
/* XXX Now request method are hardcoded in A2 */
if (ast_asprintf(&a2, "%s:%s", ast_get_http_method(method), d.uri) < 0) {
AST_RWLIST_UNLOCK(&users);
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (out of memory)");
return 0;
}
ast_md5_hash(a2_hash, a2);
ast_free(a2);
if (d.qop) {
/* RFC 2617 */
snprintf(resp, sizeof(resp), "%s:%08lx:%s:%s:auth:%s", user->a1_hash, nonce, d.nc, d.cnonce, a2_hash);
} else {
/* RFC 2069 */
snprintf(resp, sizeof(resp), "%s:%08lx:%s", user->a1_hash, nonce, a2_hash);
}
ast_md5_hash(resp_hash, resp);
}
if (strncasecmp(d.response, resp_hash, strlen(resp_hash))) {
/* Something was wrong, so give the client to try with a new challenge */
AST_RWLIST_UNLOCK(&users);
nonce = 0;
goto out_401;
}
/*
* User are pass Digest authentication.
* Now, cache the user data and unlock user list.
*/
ast_copy_string(u_username, user->username, sizeof(u_username));
u_readperm = user->readperm;
u_writeperm = user->writeperm;
u_displayconnects = user->displayconnects;
u_writetimeout = user->writetimeout;
AST_RWLIST_UNLOCK(&users);
if (!(session = find_session_by_nonce(d.username, nonce, &stale))) {
/*
* Create new session.
* While it is not in the list we don't need any locking
*/
if (!(session = build_mansession(remote_address))) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (out of memory)");
return 0;
}
ao2_lock(session);
ast_copy_string(session->username, u_username, sizeof(session->username));
session->managerid = nonce;
session->last_ev = grab_last();
AST_LIST_HEAD_INIT_NOLOCK(&session->datastores);
session->readperm = u_readperm;
session->writeperm = u_writeperm;
session->writetimeout = u_writetimeout;
if (u_displayconnects) {
ast_verb(2, "HTTP Manager '%s' logged in from %s\n", session->username, ast_sockaddr_stringify_addr(&session->addr));
}
session->noncetime = session->sessionstart = time_now;
session->authenticated = 1;
} else if (stale) {
/*
* Session found, but nonce is stale.
*
* This could be because an old request (w/old nonce) arrived.
*
* This may be as the result of http proxy usage (separate delay or
* multipath) or in a situation where a page was refreshed too quickly
* (seen in Firefox).
*
* In this situation, we repeat the 401 auth with the current nonce
* value.
*/
nonce = session->managerid;
ao2_unlock(session);
stale = 1;
goto out_401;
} else {
sscanf(d.nc, "%30lx", &nc);
if (session->nc >= nc || ((time_now - session->noncetime) > 62) ) {
/*
* Nonce time expired (> 2 minutes) or something wrong with nonce
* counter.
*
* Create new nonce key and resend Digest auth request. Old nonce
* is saved for stale checking...
*/
session->nc = 0; /* Reset nonce counter */
session->oldnonce = session->managerid;
nonce = session->managerid = ast_random();
session->noncetime = time_now;
ao2_unlock(session);
stale = 1;
goto out_401;
} else {
session->nc = nc; /* All OK, save nonce counter */
}
}
/* Reset session timeout. */
session->sessiontimeout = time(NULL) + (httptimeout > 5 ? httptimeout : 5);
ao2_unlock(session);
ast_mutex_init(&s.lock);
s.session = session;
fd = mkstemp(template); /* create a temporary file for command output */
unlink(template);
if (fd <= -1) {
ast_http_error(ser, 500, "Server Error", "Internal Server Error (mkstemp failed)");
goto auth_callback_out;
}
s.stream = ast_iostream_from_fd(&fd);
if (!s.stream) {
ast_log(LOG_WARNING, "HTTP Manager, fdopen failed: %s!\n", strerror(errno));
ast_http_error(ser, 500, "Server Error", "Internal Server Error (fdopen failed)");
close(fd);
goto auth_callback_out;
}
if (method == AST_HTTP_POST) {
params = ast_http_get_post_vars(ser, headers);
if (!params) {
switch (errno) {
case EFBIG:
ast_http_error(ser, 413, "Request Entity Too Large", "Body too large");
close_mansession_file(&s);
goto auth_callback_out;
case ENOMEM:
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Out of memory");
close_mansession_file(&s);
goto auth_callback_out;
case EIO:
ast_http_error(ser, 400, "Bad Request", "Error parsing request body");
close_mansession_file(&s);
goto auth_callback_out;
}
}
}
astman_append_headers(&m, params);
if (process_message(&s, &m)) {
if (u_displayconnects) {
ast_verb(2, "HTTP Manager '%s' logged off from %s\n", session->username, ast_sockaddr_stringify_addr(&session->addr));
}
session->needdestroy = 1;
}
astman_free_headers(&m);
result_size = lseek(ast_iostream_get_fd(s.stream), 0, SEEK_CUR); /* Calculate approx. size of result */
http_header = ast_str_create(80);
out = ast_str_create(result_size * 2 + 512);
if (http_header == NULL || out == NULL) {
ast_http_request_close_on_completion(ser);
ast_http_error(ser, 500, "Server Error", "Internal Server Error (ast_str_create() out of memory)");
close_mansession_file(&s);
goto auth_callback_out;
}
ast_str_append(&http_header, 0, "Content-type: text/%s\r\n", contenttype[format]);
if (format == FORMAT_XML) {
ast_str_append(&out, 0, "<ajax-response>\n");
} else if (format == FORMAT_HTML) {
ast_str_append(&out, 0,
"<!DOCTYPE HTML PUBLIC \"-//IETF//DTD HTML 2.0//EN\">\r\n"
"<html><head>\r\n"
"<title>Asterisk&trade; Manager Interface</title>\r\n"
"</head><body style=\"background-color: #ffffff;\">\r\n"
"<form method=\"POST\">\r\n"
"<table align=\"center\" style=\"background-color: #f1f1f1;\" width=\"500\">\r\n"
"<tr><th colspan=\"2\" style=\"background-color: #f1f1ff;\"><h1>Manager Tester</h1></th></tr>\r\n"
"<tr><th colspan=\"2\" style=\"background-color: #f1f1ff;\">Action: <input name=\"action\" /> Cmd: <input name=\"command\" /><br>"
"<input type=\"submit\" value=\"Send request\" /></th></tr>\r\n");
}
process_output(&s, &out, params, format);
if (format == FORMAT_XML) {
ast_str_append(&out, 0, "</ajax-response>\n");
} else if (format == FORMAT_HTML) {
ast_str_append(&out, 0, "</table></form></body></html>\r\n");
}
ast_http_send(ser, method, 200, NULL, http_header, out, 0, 0);
http_header = NULL;
out = NULL;
auth_callback_out:
ast_mutex_destroy(&s.lock);
/* Clear resources and unlock manager session */
if (method == AST_HTTP_POST && params) {
ast_variables_destroy(params);
}
ast_free(http_header);
ast_free(out);
ao2_lock(session);
if (session->stream) {
ast_iostream_close(session->stream);
session->stream = NULL;
}
ao2_unlock(session);
if (session->needdestroy) {
ast_debug(1, "Need destroy, doing it now!\n");
session_destroy(session);
}
ast_string_field_free_memory(&d);
return 0;
out_401:
if (!nonce) {
nonce = ast_random();
}
ast_http_auth(ser, global_realm, nonce, nonce, stale, NULL);
ast_string_field_free_memory(&d);
return 0;
}
static int manager_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = generic_http_callback(ser, method, FORMAT_HTML, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static int mxml_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = generic_http_callback(ser, method, FORMAT_XML, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static int rawman_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = generic_http_callback(ser, method, FORMAT_RAW, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static struct ast_http_uri rawmanuri = {
.description = "Raw HTTP Manager Event Interface",
.uri = "rawman",
.callback = rawman_http_callback,
.data = NULL,
.key = __FILE__,
};
static struct ast_http_uri manageruri = {
.description = "HTML Manager Event Interface",
.uri = "manager",
.callback = manager_http_callback,
.data = NULL,
.key = __FILE__,
};
static struct ast_http_uri managerxmluri = {
.description = "XML Manager Event Interface",
.uri = "mxml",
.callback = mxml_http_callback,
.data = NULL,
.key = __FILE__,
};
/* Callback with Digest authentication */
static int auth_manager_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = auth_http_callback(ser, method, FORMAT_HTML, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static int auth_mxml_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = auth_http_callback(ser, method, FORMAT_XML, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static int auth_rawman_http_callback(struct ast_tcptls_session_instance *ser, const struct ast_http_uri *urih, const char *uri, enum ast_http_method method, struct ast_variable *get_params, struct ast_variable *headers)
{
int retval;
struct ast_sockaddr ser_remote_address_tmp;
ast_sockaddr_copy(&ser_remote_address_tmp, &ser->remote_address);
retval = auth_http_callback(ser, method, FORMAT_RAW, &ser_remote_address_tmp, uri, get_params, headers);
ast_sockaddr_copy(&ser->remote_address, &ser_remote_address_tmp);
return retval;
}
static struct ast_http_uri arawmanuri = {
.description = "Raw HTTP Manager Event Interface w/Digest authentication",
.uri = "arawman",
.has_subtree = 0,
.callback = auth_rawman_http_callback,
.data = NULL,
.key = __FILE__,
};
static struct ast_http_uri amanageruri = {
.description = "HTML Manager Event Interface w/Digest authentication",
.uri = "amanager",
.has_subtree = 0,
.callback = auth_manager_http_callback,
.data = NULL,
.key = __FILE__,
};
static struct ast_http_uri amanagerxmluri = {
.description = "XML Manager Event Interface w/Digest authentication",
.uri = "amxml",
.has_subtree = 0,
.callback = auth_mxml_http_callback,
.data = NULL,
.key = __FILE__,
};
/*! \brief Get number of logged in sessions for a login name */
static int get_manager_sessions_cb(void *obj, void *arg, void *data, int flags)
{
struct mansession_session *session = obj;
const char *login = (char *)arg;
int *no_sessions = data;
if (strcasecmp(session->username, login) == 0) {
(*no_sessions)++;
}
return 0;
}
/*! \brief ${AMI_CLIENT()} Dialplan function - reads manager client data */
static int function_amiclient(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
struct ast_manager_user *user = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(name);
AST_APP_ARG(param);
);
if (ast_strlen_zero(data) ) {
ast_log(LOG_WARNING, "AMI_CLIENT() requires two arguments: AMI_CLIENT(<name>[,<arg>])\n");
return -1;
}
AST_STANDARD_APP_ARGS(args, data);
args.name = ast_strip(args.name);
args.param = ast_strip(args.param);
AST_RWLIST_RDLOCK(&users);
if (!(user = get_manager_by_name_locked(args.name))) {
AST_RWLIST_UNLOCK(&users);
ast_log(LOG_ERROR, "There's no manager user called : \"%s\"\n", args.name);
return -1;
}
AST_RWLIST_UNLOCK(&users);
if (!strcasecmp(args.param, "sessions")) {
int no_sessions = 0;
struct ao2_container *sessions;
sessions = ao2_global_obj_ref(mgr_sessions);
if (sessions) {
ao2_callback_data(sessions, 0, get_manager_sessions_cb, /*login name*/ data, &no_sessions);
ao2_ref(sessions, -1);
}
snprintf(buf, len, "%d", no_sessions);
} else {
ast_log(LOG_ERROR, "Invalid arguments provided to function AMI_CLIENT: %s\n", args.param);
return -1;
}
return 0;
}
/*! \brief description of AMI_CLIENT dialplan function */
static struct ast_custom_function managerclient_function = {
.name = "AMI_CLIENT",
.read = function_amiclient,
.read_max = 12,
};
static int webregged = 0;
/*! \brief cleanup code called at each iteration of server_root,
* guaranteed to happen every 5 seconds at most
*/
static void purge_old_stuff(void *data)
{
struct ast_tcptls_session_args *ser = data;
/* purge_sessions will return the number of sessions actually purged,
* up to a maximum of it's arguments, purge one at a time, keeping a
* purge interval of 1ms as long as we purged a session, otherwise
* revert to a purge check every 5s
*/
if (purge_sessions(1) == 1) {
ser->poll_timeout = 1;
} else {
ser->poll_timeout = 5000;
}
purge_events();
}
static struct ast_tls_config ami_tls_cfg;
static struct ast_tcptls_session_args ami_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
.tls_cfg = NULL,
.poll_timeout = 5000, /* wake up every 5 seconds */
.periodic_fn = purge_old_stuff,
.name = "AMI server",
.accept_fn = ast_tcptls_server_root, /* thread doing the accept() */
.worker_fn = session_do, /* thread handling the session */
};
static struct ast_tcptls_session_args amis_desc = {
.accept_fd = -1,
.master = AST_PTHREADT_NULL,
.tls_cfg = &ami_tls_cfg,
.poll_timeout = -1, /* the other does the periodic cleanup */
.name = "AMI TLS server",
.accept_fn = ast_tcptls_server_root, /* thread doing the accept() */
.worker_fn = session_do, /* thread handling the session */
};
/*! \brief CLI command manager show settings */
static char *handle_manager_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
switch (cmd) {
case CLI_INIT:
e->command = "manager show settings";
e->usage =
"Usage: manager show settings\n"
" Provides detailed list of the configuration of the Manager.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
#define FORMAT " %-25.25s %-15.55s\n"
#define FORMAT2 " %-25.25s %-15d\n"
#define FORMAT3 " %-25.25s %s\n"
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
ast_cli(a->fd, "\nGlobal Settings:\n");
ast_cli(a->fd, "----------------\n");
ast_cli(a->fd, FORMAT, "Manager (AMI):", AST_CLI_YESNO(manager_enabled));
ast_cli(a->fd, FORMAT, "Web Manager (AMI/HTTP):", AST_CLI_YESNO(webmanager_enabled));
ast_cli(a->fd, FORMAT, "TCP Bindaddress:", manager_enabled != 0 ? ast_sockaddr_stringify(&ami_desc.local_address) : "Disabled");
ast_cli(a->fd, FORMAT2, "HTTP Timeout (seconds):", httptimeout);
ast_cli(a->fd, FORMAT, "TLS Enable:", AST_CLI_YESNO(ami_tls_cfg.enabled));
ast_cli(a->fd, FORMAT, "TLS Bindaddress:", ami_tls_cfg.enabled != 0 ? ast_sockaddr_stringify(&amis_desc.local_address) : "Disabled");
ast_cli(a->fd, FORMAT, "TLS Certfile:", ami_tls_cfg.certfile);
ast_cli(a->fd, FORMAT, "TLS Privatekey:", ami_tls_cfg.pvtfile);
ast_cli(a->fd, FORMAT, "TLS Cipher:", ami_tls_cfg.cipher);
ast_cli(a->fd, FORMAT, "Allow multiple login:", AST_CLI_YESNO(allowmultiplelogin));
ast_cli(a->fd, FORMAT, "Display connects:", AST_CLI_YESNO(displayconnects));
ast_cli(a->fd, FORMAT, "Timestamp events:", AST_CLI_YESNO(timestampevents));
ast_cli(a->fd, FORMAT3, "Channel vars:", S_OR(manager_channelvars, ""));
ast_cli(a->fd, FORMAT3, "Disabled events:", S_OR(manager_disabledevents, ""));
ast_cli(a->fd, FORMAT, "Debug:", AST_CLI_YESNO(manager_debug));
#undef FORMAT
#undef FORMAT2
#undef FORMAT3
return CLI_SUCCESS;
}
#ifdef AST_XML_DOCS
static int ast_xml_doc_item_cmp_fn(const void *a, const void *b)
{
struct ast_xml_doc_item **item_a = (struct ast_xml_doc_item **)a;
struct ast_xml_doc_item **item_b = (struct ast_xml_doc_item **)b;
return strcmp((*item_a)->name, (*item_b)->name);
}
static char *handle_manager_show_events(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct ao2_container *events;
struct ao2_iterator *it_events;
struct ast_xml_doc_item *item;
struct ast_xml_doc_item **items;
struct ast_str *buffer;
int i = 0, totalitems = 0;
switch (cmd) {
case CLI_INIT:
e->command = "manager show events";
e->usage =
"Usage: manager show events\n"
" Prints a listing of the available Asterisk manager interface events.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
buffer = ast_str_create(128);
if (!buffer) {
return CLI_SUCCESS;
}
events = ao2_global_obj_ref(event_docs);
if (!events) {
ast_cli(a->fd, "No manager event documentation loaded\n");
ast_free(buffer);
return CLI_SUCCESS;
}
ao2_lock(events);
if (!(it_events = ao2_callback(events, OBJ_MULTIPLE | OBJ_NOLOCK, NULL, NULL))) {
ao2_unlock(events);
ast_log(AST_LOG_ERROR, "Unable to create iterator for events container\n");
ast_free(buffer);
ao2_ref(events, -1);
return CLI_SUCCESS;
}
if (!(items = ast_calloc(sizeof(struct ast_xml_doc_item *), ao2_container_count(events)))) {
ao2_unlock(events);
ast_log(AST_LOG_ERROR, "Unable to create temporary sorting array for events\n");
ao2_iterator_destroy(it_events);
ast_free(buffer);
ao2_ref(events, -1);
return CLI_SUCCESS;
}
ao2_unlock(events);
while ((item = ao2_iterator_next(it_events))) {
items[totalitems++] = item;
ao2_ref(item, -1);
}
qsort(items, totalitems, sizeof(struct ast_xml_doc_item *), ast_xml_doc_item_cmp_fn);
ast_cli(a->fd, "Events:\n");
ast_cli(a->fd, " -------------------- -------------------- -------------------- \n");
for (i = 0; i < totalitems; i++) {
ast_str_append(&buffer, 0, " %-20.20s", items[i]->name);
if ((i + 1) % 3 == 0) {
ast_cli(a->fd, "%s\n", ast_str_buffer(buffer));
ast_str_set(&buffer, 0, "%s", "");
}
}
if ((i + 1) % 3 != 0) {
ast_cli(a->fd, "%s\n", ast_str_buffer(buffer));
}
ao2_iterator_destroy(it_events);
ast_free(items);
ao2_ref(events, -1);
ast_free(buffer);
return CLI_SUCCESS;
}
static void print_event_instance(struct ast_cli_args *a, struct ast_xml_doc_item *instance)
{
char syntax_title[64], description_title[64], synopsis_title[64], seealso_title[64], arguments_title[64];
term_color(synopsis_title, "[Synopsis]\n", COLOR_MAGENTA, 0, 40);
term_color(description_title, "[Description]\n", COLOR_MAGENTA, 0, 40);
term_color(syntax_title, "[Syntax]\n", COLOR_MAGENTA, 0, 40);
term_color(seealso_title, "[See Also]\n", COLOR_MAGENTA, 0, 40);
term_color(arguments_title, "[Arguments]\n", COLOR_MAGENTA, 0, 40);
if (!ast_strlen_zero(ast_str_buffer(instance->synopsis))) {
char *synopsis = ast_xmldoc_printable(ast_str_buffer(instance->synopsis), 1);
ast_cli(a->fd, "%s%s\n\n", synopsis_title, synopsis);
ast_free(synopsis);
}
if (!ast_strlen_zero(ast_str_buffer(instance->syntax))) {
char *syntax = ast_xmldoc_printable(ast_str_buffer(instance->syntax), 1);
ast_cli(a->fd, "%s%s\n\n", syntax_title, syntax);
ast_free(syntax);
}
if (!ast_strlen_zero(ast_str_buffer(instance->description))) {
char *description = ast_xmldoc_printable(ast_str_buffer(instance->description), 1);
ast_cli(a->fd, "%s%s\n\n", description_title, description);
ast_free(description);
}
if (!ast_strlen_zero(ast_str_buffer(instance->arguments))) {
char *arguments = ast_xmldoc_printable(ast_str_buffer(instance->arguments), 1);
ast_cli(a->fd, "%s%s\n\n", arguments_title, arguments);
ast_free(arguments);
}
if (!ast_strlen_zero(ast_str_buffer(instance->seealso))) {
char *seealso = ast_xmldoc_printable(ast_str_buffer(instance->seealso), 1);
ast_cli(a->fd, "%s%s\n\n", seealso_title, seealso);
ast_free(seealso);
}
}
static char *handle_manager_show_event(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
RAII_VAR(struct ao2_container *, events, NULL, ao2_cleanup);
struct ao2_iterator it_events;
struct ast_xml_doc_item *item, *temp;
int length;
if (cmd == CLI_INIT) {
e->command = "manager show event";
e->usage =
"Usage: manager show event <eventname>\n"
" Provides a detailed description a Manager interface event.\n";
return NULL;
}
events = ao2_global_obj_ref(event_docs);
if (!events) {
ast_cli(a->fd, "No manager event documentation loaded\n");
return CLI_SUCCESS;
}
if (cmd == CLI_GENERATE) {
if (a->pos != 3) {
return NULL;
}
length = strlen(a->word);
it_events = ao2_iterator_init(events, 0);
while ((item = ao2_iterator_next(&it_events))) {
if (!strncasecmp(a->word, item->name, length)) {
if (ast_cli_completion_add(ast_strdup(item->name))) {
ao2_ref(item, -1);
break;
}
}
ao2_ref(item, -1);
}
ao2_iterator_destroy(&it_events);
return NULL;
}
if (a->argc != 4) {
return CLI_SHOWUSAGE;
}
if (!(item = ao2_find(events, a->argv[3], OBJ_KEY))) {
ast_cli(a->fd, "Could not find event '%s'\n", a->argv[3]);
return CLI_SUCCESS;
}
ast_cli(a->fd, "Event: %s\n", a->argv[3]);
for (temp = item; temp; temp = AST_LIST_NEXT(temp, next)) {
print_event_instance(a, temp);
}
ao2_ref(item, -1);
return CLI_SUCCESS;
}
#endif
static struct ast_cli_entry cli_manager[] = {
AST_CLI_DEFINE(handle_showmancmd, "Show a manager interface command"),
AST_CLI_DEFINE(handle_showmancmds, "List manager interface commands"),
AST_CLI_DEFINE(handle_showmanconn, "List connected manager interface users"),
AST_CLI_DEFINE(handle_showmaneventq, "List manager interface queued events"),
AST_CLI_DEFINE(handle_showmanagers, "List configured manager users"),
AST_CLI_DEFINE(handle_showmanager, "Display information on a specific manager user"),
AST_CLI_DEFINE(handle_mandebug, "Show, enable, disable debugging of the manager code"),
AST_CLI_DEFINE(handle_manager_reload, "Reload manager configurations"),
AST_CLI_DEFINE(handle_manager_show_settings, "Show manager global settings"),
#ifdef AST_XML_DOCS
AST_CLI_DEFINE(handle_manager_show_events, "List manager interface events"),
AST_CLI_DEFINE(handle_manager_show_event, "Show a manager interface event"),
#endif
};
/*!
* \internal
* \brief Load the config channelvars variable.
*
* \param var Config variable to load.
*/
static void load_channelvars(struct ast_variable *var)
{
char *parse = NULL;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(vars)[MAX_VARS];
);
ast_free(manager_channelvars);
manager_channelvars = ast_strdup(var->value);
/* parse the setting */
parse = ast_strdupa(manager_channelvars);
AST_STANDARD_APP_ARGS(args, parse);
ast_channel_set_manager_vars(args.argc, args.vars);
}
/*!
* \internal
* \brief Load the config disabledevents variable.
*
* \param var Config variable to load.
*/
static void load_disabledevents(struct ast_variable *var)
{
ast_free(manager_disabledevents);
manager_disabledevents = ast_strdup(var->value);
}
/*!
* \internal
* \brief Free a user record. Should already be removed from the list
*/
static void manager_free_user(struct ast_manager_user *user)
{
ast_free(user->a1_hash);
ast_free(user->secret);
if (user->whitefilters) {
ao2_t_ref(user->whitefilters, -1, "decrement ref for white container, should be last one");
}
if (user->blackfilters) {
ao2_t_ref(user->blackfilters, -1, "decrement ref for black container, should be last one");
}
user->acl = ast_free_acl_list(user->acl);
ast_variables_destroy(user->chanvars);
ast_free(user);
}
/*!
* \internal
* \brief Clean up resources on Asterisk shutdown
*/
static void manager_shutdown(void)
{
struct ast_manager_user *user;
/* This event is not actually transmitted, but causes all TCP sessions to be closed */
manager_event(EVENT_FLAG_SHUTDOWN, "CloseSession", "CloseSession: true\r\n");
ast_manager_unregister("Ping");
ast_manager_unregister("Events");
ast_manager_unregister("Logoff");
ast_manager_unregister("Login");
ast_manager_unregister("Challenge");
ast_manager_unregister("Hangup");
ast_manager_unregister("Status");
ast_manager_unregister("Setvar");
ast_manager_unregister("Getvar");
ast_manager_unregister("GetConfig");
ast_manager_unregister("GetConfigJSON");
ast_manager_unregister("UpdateConfig");
ast_manager_unregister("CreateConfig");
ast_manager_unregister("ListCategories");
ast_manager_unregister("Redirect");
ast_manager_unregister("Atxfer");
ast_manager_unregister("CancelAtxfer");
ast_manager_unregister("Originate");
ast_manager_unregister("Command");
ast_manager_unregister("ExtensionState");
ast_manager_unregister("PresenceState");
ast_manager_unregister("AbsoluteTimeout");
ast_manager_unregister("MailboxStatus");
ast_manager_unregister("MailboxCount");
ast_manager_unregister("ListCommands");
ast_manager_unregister("SendText");
ast_manager_unregister("UserEvent");
ast_manager_unregister("WaitEvent");
ast_manager_unregister("CoreSettings");
ast_manager_unregister("CoreStatus");
ast_manager_unregister("Reload");
ast_manager_unregister("LoggerRotate");
ast_manager_unregister("CoreShowChannels");
ast_manager_unregister("CoreShowChannelMap");
ast_manager_unregister("ModuleLoad");
ast_manager_unregister("ModuleCheck");
ast_manager_unregister("AOCMessage");
ast_manager_unregister("Filter");
ast_manager_unregister("BlindTransfer");
ast_custom_function_unregister(&managerclient_function);
ast_cli_unregister_multiple(cli_manager, ARRAY_LEN(cli_manager));
#ifdef AST_XML_DOCS
ao2_t_global_obj_release(event_docs, "Dispose of event_docs");
#endif
#ifdef TEST_FRAMEWORK
stasis_forward_cancel(test_suite_forwarder);
test_suite_forwarder = NULL;
#endif
if (stasis_router) {
stasis_message_router_unsubscribe_and_join(stasis_router);
stasis_router = NULL;
}
Multiple revisions 399887,400138,400178,400180-400181 ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30 18:55:27 +00:00
stasis_forward_cancel(rtp_topic_forwarder);
rtp_topic_forwarder = NULL;
stasis_forward_cancel(security_topic_forwarder);
security_topic_forwarder = NULL;
ao2_cleanup(manager_topic);
manager_topic = NULL;
STASIS_MESSAGE_TYPE_CLEANUP(ast_manager_get_generic_type);
ast_tcptls_server_stop(&ami_desc);
ast_tcptls_server_stop(&amis_desc);
ast_free(ami_tls_cfg.certfile);
ami_tls_cfg.certfile = NULL;
ast_free(ami_tls_cfg.pvtfile);
ami_tls_cfg.pvtfile = NULL;
ast_free(ami_tls_cfg.cipher);
ami_tls_cfg.cipher = NULL;
ast_free(ami_tls_cfg.cafile);
ami_tls_cfg.cafile = NULL;
ast_free(ami_tls_cfg.capath);
ami_tls_cfg.capath = NULL;
ao2_global_obj_release(mgr_sessions);
while ((user = AST_LIST_REMOVE_HEAD(&users, list))) {
manager_free_user(user);
}
acl_change_stasis_unsubscribe();
ast_free(manager_channelvars);
ast_free(manager_disabledevents);
}
/*! \brief Initialize all \ref stasis topics and routers used by the various
* sub-components of AMI
*/
static int manager_subscriptions_init(void)
{
int res = 0;
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
rtp_topic_forwarder = stasis_forward_all(ast_rtp_topic(), manager_topic);
if (!rtp_topic_forwarder) {
return -1;
}
security_topic_forwarder = stasis_forward_all(ast_security_topic(), manager_topic);
if (!security_topic_forwarder) {
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
return -1;
}
stasis_router = stasis_message_router_create(manager_topic);
if (!stasis_router) {
return -1;
}
stasis_message_router_set_congestion_limits(stasis_router, -1,
6 * AST_TASKPROCESSOR_HIGH_WATER_LEVEL);
stasis_message_router_set_formatters_default(stasis_router,
manager_default_msg_cb, NULL, STASIS_SUBSCRIPTION_FORMATTER_AMI);
res |= stasis_message_router_add(stasis_router,
ast_manager_get_generic_type(), manager_generic_msg_cb, NULL);
if (res != 0) {
return -1;
}
return 0;
}
static int subscribe_all(void)
{
if (manager_subscriptions_init()) {
ast_log(AST_LOG_ERROR, "Failed to initialize manager subscriptions\n");
return -1;
}
if (manager_system_init()) {
ast_log(AST_LOG_ERROR, "Failed to initialize manager system handling\n");
return -1;
}
if (manager_channels_init()) {
ast_log(AST_LOG_ERROR, "Failed to initialize manager channel handling\n");
return -1;
}
if (manager_mwi_init()) {
ast_log(AST_LOG_ERROR, "Failed to initialize manager MWI handling\n");
return -1;
}
if (manager_bridging_init()) {
return -1;
}
if (manager_endpoints_init()) {
ast_log(AST_LOG_ERROR, "Failed to initialize manager endpoints handling\n");
return -1;
}
subscribed = 1;
return 0;
}
static void manager_set_defaults(void)
{
manager_enabled = 0;
displayconnects = 1;
broken_events_action = 0;
authtimeout = 30;
authlimit = 50;
manager_debug = 0; /* Debug disabled by default */
/* default values */
ast_copy_string(global_realm, S_OR(ast_config_AST_SYSTEM_NAME, DEFAULT_REALM),
sizeof(global_realm));
ast_sockaddr_setnull(&ami_desc.local_address);
ast_sockaddr_setnull(&amis_desc.local_address);
ami_tls_cfg.enabled = 0;
ast_free(ami_tls_cfg.certfile);
ami_tls_cfg.certfile = ast_strdup(AST_CERTFILE);
ast_free(ami_tls_cfg.pvtfile);
ami_tls_cfg.pvtfile = ast_strdup("");
ast_free(ami_tls_cfg.cipher);
ami_tls_cfg.cipher = ast_strdup("");
ast_free(ami_tls_cfg.cafile);
ami_tls_cfg.cafile = ast_strdup("");
ast_free(ami_tls_cfg.capath);
ami_tls_cfg.capath = ast_strdup("");
}
static int __init_manager(int reload, int by_external_config)
{
struct ast_config *ucfg = NULL, *cfg = NULL;
const char *val;
char *cat = NULL;
int newhttptimeout = 60;
struct ast_manager_user *user = NULL;
struct ast_variable *var;
struct ast_flags config_flags = { (reload && !by_external_config) ? CONFIG_FLAG_FILEUNCHANGED : 0 };
char a1[256];
char a1_hash[256];
struct ast_sockaddr ami_desc_local_address_tmp;
struct ast_sockaddr amis_desc_local_address_tmp;
int tls_was_enabled = 0;
int acl_subscription_flag = 0;
if (!reload) {
struct ao2_container *sessions;
#ifdef AST_XML_DOCS
struct ao2_container *temp_event_docs;
#endif
int res;
res = STASIS_MESSAGE_TYPE_INIT(ast_manager_get_generic_type);
if (res != 0) {
return -1;
}
manager_topic = stasis_topic_create("manager:core");
if (!manager_topic) {
return -1;
}
/* Register default actions */
ast_manager_register_xml_core("Ping", 0, action_ping);
ast_manager_register_xml_core("Events", 0, action_events);
ast_manager_register_xml_core("Logoff", 0, action_logoff);
ast_manager_register_xml_core("Login", 0, action_login);
ast_manager_register_xml_core("Challenge", 0, action_challenge);
ast_manager_register_xml_core("Hangup", EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, action_hangup);
ast_manager_register_xml_core("Status", EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_status);
ast_manager_register_xml_core("Setvar", EVENT_FLAG_CALL, action_setvar);
ast_manager_register_xml_core("Getvar", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_getvar);
ast_manager_register_xml_core("GetConfig", EVENT_FLAG_SYSTEM | EVENT_FLAG_CONFIG, action_getconfig);
ast_manager_register_xml_core("GetConfigJSON", EVENT_FLAG_SYSTEM | EVENT_FLAG_CONFIG, action_getconfigjson);
ast_manager_register_xml_core("UpdateConfig", EVENT_FLAG_CONFIG, action_updateconfig);
ast_manager_register_xml_core("CreateConfig", EVENT_FLAG_CONFIG, action_createconfig);
ast_manager_register_xml_core("ListCategories", EVENT_FLAG_CONFIG, action_listcategories);
ast_manager_register_xml_core("Redirect", EVENT_FLAG_CALL, action_redirect);
ast_manager_register_xml_core("Atxfer", EVENT_FLAG_CALL, action_atxfer);
ast_manager_register_xml_core("CancelAtxfer", EVENT_FLAG_CALL, action_cancel_atxfer);
ast_manager_register_xml_core("Originate", EVENT_FLAG_ORIGINATE, action_originate);
ast_manager_register_xml_core("Command", EVENT_FLAG_COMMAND, action_command);
ast_manager_register_xml_core("ExtensionState", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_extensionstate);
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
ast_manager_register_xml_core("PresenceState", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_presencestate);
ast_manager_register_xml_core("AbsoluteTimeout", EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, action_timeout);
ast_manager_register_xml_core("MailboxStatus", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_mailboxstatus);
ast_manager_register_xml_core("MailboxCount", EVENT_FLAG_CALL | EVENT_FLAG_REPORTING, action_mailboxcount);
ast_manager_register_xml_core("ListCommands", 0, action_listcommands);
ast_manager_register_xml_core("SendText", EVENT_FLAG_CALL, action_sendtext);
ast_manager_register_xml_core("UserEvent", EVENT_FLAG_USER, action_userevent);
ast_manager_register_xml_core("WaitEvent", 0, action_waitevent);
ast_manager_register_xml_core("CoreSettings", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, action_coresettings);
ast_manager_register_xml_core("CoreStatus", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, action_corestatus);
ast_manager_register_xml_core("Reload", EVENT_FLAG_CONFIG | EVENT_FLAG_SYSTEM, action_reload);
ast_manager_register_xml_core("LoggerRotate", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, action_loggerrotate);
ast_manager_register_xml_core("CoreShowChannels", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, action_coreshowchannels);
ast_manager_register_xml_core("CoreShowChannelMap", EVENT_FLAG_SYSTEM | EVENT_FLAG_REPORTING, action_coreshowchannelmap);
ast_manager_register_xml_core("ModuleLoad", EVENT_FLAG_SYSTEM, manager_moduleload);
ast_manager_register_xml_core("ModuleCheck", EVENT_FLAG_SYSTEM, manager_modulecheck);
ast_manager_register_xml_core("AOCMessage", EVENT_FLAG_AOC, action_aocmessage);
ast_manager_register_xml_core("Filter", EVENT_FLAG_SYSTEM, action_filter);
ast_manager_register_xml_core("BlindTransfer", EVENT_FLAG_CALL, action_blind_transfer);
#ifdef TEST_FRAMEWORK
test_suite_forwarder = stasis_forward_all(ast_test_suite_topic(), manager_topic);
#endif
ast_cli_register_multiple(cli_manager, ARRAY_LEN(cli_manager));
__ast_custom_function_register(&managerclient_function, NULL);
ast_extension_state_add(NULL, NULL, manager_state_cb, NULL);
/* Append placeholder event so master_eventq never runs dry */
if (append_event("Event: Placeholder\r\n\r\n", 0)) {
return -1;
}
#ifdef AST_XML_DOCS
temp_event_docs = ast_xmldoc_build_documentation("managerEvent");
if (temp_event_docs) {
ao2_t_global_obj_replace_unref(event_docs, temp_event_docs, "Toss old event docs");
ao2_t_ref(temp_event_docs, -1, "Remove creation ref - container holds only ref now");
}
#endif
/* If you have a NULL hash fn, you only need a single bucket */
sessions = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, mansession_cmp_fn);
if (!sessions) {
return -1;
}
ao2_global_obj_replace_unref(mgr_sessions, sessions);
ao2_ref(sessions, -1);
/* Initialize all settings before first configuration load. */
manager_set_defaults();
}
cfg = ast_config_load2("manager.conf", "manager", config_flags);
if (cfg == CONFIG_STATUS_FILEUNCHANGED) {
return 0;
} else if (!cfg || cfg == CONFIG_STATUS_FILEINVALID) {
ast_log(LOG_NOTICE, "Unable to open AMI configuration manager.conf, or configuration is invalid.\n");
return 0;
}
/* If this wasn't performed due to a forced reload (because those can be created by ACL change events, we need to unsubscribe to ACL change events. */
if (!by_external_config) {
acl_change_stasis_unsubscribe();
}
if (reload) {
/* Reset all settings before reloading configuration */
tls_was_enabled = ami_tls_cfg.enabled;
manager_set_defaults();
}
ast_sockaddr_parse(&ami_desc_local_address_tmp, "[::]", 0);
ast_sockaddr_set_port(&ami_desc_local_address_tmp, DEFAULT_MANAGER_PORT);
for (var = ast_variable_browse(cfg, "general"); var; var = var->next) {
val = var->value;
/* read tls config options while preventing unsupported options from being set */
if (strcasecmp(var->name, "tlscafile")
&& strcasecmp(var->name, "tlscapath")
&& strcasecmp(var->name, "tlscadir")
&& strcasecmp(var->name, "tlsverifyclient")
&& strcasecmp(var->name, "tlsdontverifyserver")
&& strcasecmp(var->name, "tlsclientmethod")
&& strcasecmp(var->name, "sslclientmethod")
&& !ast_tls_read_conf(&ami_tls_cfg, &amis_desc, var->name, val)) {
continue;
}
if (!strcasecmp(var->name, "enabled")) {
manager_enabled = ast_true(val);
} else if (!strcasecmp(var->name, "webenabled")) {
webmanager_enabled = ast_true(val);
} else if (!strcasecmp(var->name, "port")) {
int bindport;
if (ast_parse_arg(val, PARSE_UINT32|PARSE_IN_RANGE, &bindport, 1024, 65535)) {
ast_log(LOG_WARNING, "Invalid port number '%s'\n", val);
}
ast_sockaddr_set_port(&ami_desc_local_address_tmp, bindport);
} else if (!strcasecmp(var->name, "bindaddr")) {
/* remember port if it has already been set */
int setport = ast_sockaddr_port(&ami_desc_local_address_tmp);
if (ast_parse_arg(val, PARSE_ADDR|PARSE_PORT_IGNORE, NULL)) {
ast_log(LOG_WARNING, "Invalid address '%s' specified, default '%s' will be used\n", val,
ast_sockaddr_stringify_addr(&ami_desc_local_address_tmp));
} else {
ast_sockaddr_parse(&ami_desc_local_address_tmp, val, PARSE_PORT_IGNORE);
}
if (setport) {
ast_sockaddr_set_port(&ami_desc_local_address_tmp, setport);
}
} else if (!strcasecmp(var->name, "brokeneventsaction")) {
broken_events_action = ast_true(val);
} else if (!strcasecmp(var->name, "allowmultiplelogin")) {
allowmultiplelogin = ast_true(val);
} else if (!strcasecmp(var->name, "displayconnects")) {
displayconnects = ast_true(val);
} else if (!strcasecmp(var->name, "timestampevents")) {
timestampevents = ast_true(val);
} else if (!strcasecmp(var->name, "debug")) {
manager_debug = ast_true(val);
} else if (!strcasecmp(var->name, "httptimeout")) {
newhttptimeout = atoi(val);
} else if (!strcasecmp(var->name, "authtimeout")) {
int timeout = atoi(var->value);
if (timeout < 1) {
ast_log(LOG_WARNING, "Invalid authtimeout value '%s', using default value\n", var->value);
} else {
authtimeout = timeout;
}
} else if (!strcasecmp(var->name, "authlimit")) {
int limit = atoi(var->value);
if (limit < 1) {
ast_log(LOG_WARNING, "Invalid authlimit value '%s', using default value\n", var->value);
} else {
authlimit = limit;
}
} else if (!strcasecmp(var->name, "channelvars")) {
load_channelvars(var);
} else if (!strcasecmp(var->name, "disabledevents")) {
load_disabledevents(var);
} else {
ast_log(LOG_NOTICE, "Invalid keyword <%s> = <%s> in manager.conf [general]\n",
var->name, val);
}
}
if (manager_enabled && !subscribed) {
if (subscribe_all() != 0) {
ast_log(LOG_ERROR, "Manager subscription error\n");
return -1;
}
}
ast_sockaddr_copy(&amis_desc_local_address_tmp, &amis_desc.local_address);
/* if the amis address has not been set, default is the same as non secure ami */
if (ast_sockaddr_isnull(&amis_desc_local_address_tmp)) {
ast_sockaddr_copy(&amis_desc_local_address_tmp, &ami_desc_local_address_tmp);
}
/* if the amis address was not set, it will have non-secure ami port set; if
amis address was set, we need to check that a port was set or not, if not
use the default tls port */
if (ast_sockaddr_port(&amis_desc_local_address_tmp) == 0 ||
(ast_sockaddr_port(&ami_desc_local_address_tmp) == ast_sockaddr_port(&amis_desc_local_address_tmp))) {
ast_sockaddr_set_port(&amis_desc_local_address_tmp, DEFAULT_MANAGER_TLS_PORT);
}
if (manager_enabled) {
ast_sockaddr_copy(&ami_desc.local_address, &ami_desc_local_address_tmp);
ast_sockaddr_copy(&amis_desc.local_address, &amis_desc_local_address_tmp);
}
AST_RWLIST_WRLOCK(&users);
/* First, get users from users.conf */
ucfg = ast_config_load2("users.conf", "manager", config_flags);
if (ucfg && (ucfg != CONFIG_STATUS_FILEUNCHANGED) && ucfg != CONFIG_STATUS_FILEINVALID) {
const char *hasmanager;
int genhasmanager = ast_true(ast_variable_retrieve(ucfg, "general", "hasmanager"));
while ((cat = ast_category_browse(ucfg, cat))) {
if (!strcasecmp(cat, "general")) {
continue;
}
hasmanager = ast_variable_retrieve(ucfg, cat, "hasmanager");
if ((!hasmanager && genhasmanager) || ast_true(hasmanager)) {
const char *user_secret = ast_variable_retrieve(ucfg, cat, "secret");
const char *user_read = ast_variable_retrieve(ucfg, cat, "read");
const char *user_write = ast_variable_retrieve(ucfg, cat, "write");
const char *user_displayconnects = ast_variable_retrieve(ucfg, cat, "displayconnects");
const char *user_allowmultiplelogin = ast_variable_retrieve(ucfg, cat, "allowmultiplelogin");
const char *user_writetimeout = ast_variable_retrieve(ucfg, cat, "writetimeout");
/* Look for an existing entry,
* if none found - create one and add it to the list
*/
if (!(user = get_manager_by_name_locked(cat))) {
if (!(user = ast_calloc(1, sizeof(*user)))) {
break;
}
/* Copy name over */
ast_copy_string(user->username, cat, sizeof(user->username));
/* Insert into list */
AST_LIST_INSERT_TAIL(&users, user, list);
user->acl = NULL;
user->keep = 1;
user->readperm = -1;
user->writeperm = -1;
/* Default displayconnect from [general] */
user->displayconnects = displayconnects;
/* Default allowmultiplelogin from [general] */
user->allowmultiplelogin = allowmultiplelogin;
user->writetimeout = 100;
}
if (!user_secret) {
user_secret = ast_variable_retrieve(ucfg, "general", "secret");
}
if (!user_read) {
user_read = ast_variable_retrieve(ucfg, "general", "read");
}
if (!user_write) {
user_write = ast_variable_retrieve(ucfg, "general", "write");
}
if (!user_displayconnects) {
user_displayconnects = ast_variable_retrieve(ucfg, "general", "displayconnects");
}
if (!user_allowmultiplelogin) {
user_allowmultiplelogin = ast_variable_retrieve(ucfg, "general", "allowmultiplelogin");
}
if (!user_writetimeout) {
user_writetimeout = ast_variable_retrieve(ucfg, "general", "writetimeout");
}
if (!ast_strlen_zero(user_secret)) {
ast_free(user->secret);
user->secret = ast_strdup(user_secret);
}
if (user_read) {
user->readperm = get_perm(user_read);
}
if (user_write) {
user->writeperm = get_perm(user_write);
}
if (user_displayconnects) {
user->displayconnects = ast_true(user_displayconnects);
}
if (user_allowmultiplelogin) {
user->allowmultiplelogin = ast_true(user_allowmultiplelogin);
}
if (user_writetimeout) {
int value = atoi(user_writetimeout);
if (value < 100) {
ast_log(LOG_WARNING, "Invalid writetimeout value '%d' in users.conf\n", value);
} else {
user->writetimeout = value;
}
}
}
}
ast_config_destroy(ucfg);
}
/* cat is NULL here in any case */
while ((cat = ast_category_browse(cfg, cat))) {
struct ast_acl_list *oldacl;
if (!strcasecmp(cat, "general")) {
continue;
}
/* Look for an existing entry, if none found - create one and add it to the list */
if (!(user = get_manager_by_name_locked(cat))) {
if (!(user = ast_calloc(1, sizeof(*user)))) {
break;
}
/* Copy name over */
ast_copy_string(user->username, cat, sizeof(user->username));
user->acl = NULL;
user->readperm = 0;
user->writeperm = 0;
/* Default displayconnect from [general] */
user->displayconnects = displayconnects;
/* Default allowmultiplelogin from [general] */
user->allowmultiplelogin = allowmultiplelogin;
user->writetimeout = 100;
user->whitefilters = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, NULL);
user->blackfilters = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, NULL);
if (!user->whitefilters || !user->blackfilters) {
manager_free_user(user);
break;
}
/* Insert into list */
AST_RWLIST_INSERT_TAIL(&users, user, list);
} else {
ao2_t_callback(user->whitefilters, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL, "unlink all white filters");
ao2_t_callback(user->blackfilters, OBJ_UNLINK | OBJ_NODATA | OBJ_MULTIPLE, NULL, NULL, "unlink all black filters");
}
/* Make sure we keep this user and don't destroy it during cleanup */
user->keep = 1;
oldacl = user->acl;
user->acl = NULL;
ast_variables_destroy(user->chanvars);
var = ast_variable_browse(cfg, cat);
for (; var; var = var->next) {
if (!strcasecmp(var->name, "secret")) {
ast_free(user->secret);
user->secret = ast_strdup(var->value);
} else if (!strcasecmp(var->name, "deny") ||
!strcasecmp(var->name, "permit") ||
!strcasecmp(var->name, "acl")) {
int acl_error = 0;
ast_append_acl(var->name, var->value, &user->acl, &acl_error, &acl_subscription_flag);
if (acl_error) {
ast_log(LOG_ERROR, "Invalid ACL '%s' for manager user '%s' on line %d. Deleting user\n",
var->value, user->username, var->lineno);
user->keep = 0;
}
} else if (!strcasecmp(var->name, "read") ) {
user->readperm = get_perm(var->value);
} else if (!strcasecmp(var->name, "write") ) {
user->writeperm = get_perm(var->value);
} else if (!strcasecmp(var->name, "displayconnects") ) {
user->displayconnects = ast_true(var->value);
} else if (!strcasecmp(var->name, "allowmultiplelogin") ) {
user->allowmultiplelogin = ast_true(var->value);
} else if (!strcasecmp(var->name, "writetimeout")) {
int value = atoi(var->value);
if (value < 100) {
ast_log(LOG_WARNING, "Invalid writetimeout value '%s' at line %d\n", var->value, var->lineno);
} else {
user->writetimeout = value;
}
} else if (!strcasecmp(var->name, "setvar")) {
struct ast_variable *tmpvar;
char varbuf[256];
char *varval;
char *varname;
ast_copy_string(varbuf, var->value, sizeof(varbuf));
varname = varbuf;
if ((varval = strchr(varname,'='))) {
*varval++ = '\0';
if ((tmpvar = ast_variable_new(varname, varval, ""))) {
tmpvar->next = user->chanvars;
user->chanvars = tmpvar;
}
}
} else if (!strcasecmp(var->name, "eventfilter")) {
const char *value = var->value;
manager_add_filter(value, user->whitefilters, user->blackfilters);
} else {
ast_debug(1, "%s is an unknown option.\n", var->name);
}
}
oldacl = ast_free_acl_list(oldacl);
}
ast_config_destroy(cfg);
/* Check the flag for named ACL event subscription and if we need to, register a subscription. */
if (acl_subscription_flag && !by_external_config) {
acl_change_stasis_subscribe();
}
/* Perform cleanup - essentially prune out old users that no longer exist */
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&users, user, list) {
if (user->keep) { /* valid record. clear flag for the next round */
user->keep = 0;
/* Calculate A1 for Digest auth */
snprintf(a1, sizeof(a1), "%s:%s:%s", user->username, global_realm, user->secret);
ast_md5_hash(a1_hash,a1);
ast_free(user->a1_hash);
user->a1_hash = ast_strdup(a1_hash);
continue;
}
/* We do not need to keep this user so take them out of the list */
AST_RWLIST_REMOVE_CURRENT(list);
ast_debug(4, "Pruning user '%s'\n", user->username);
manager_free_user(user);
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&users);
if (webmanager_enabled && manager_enabled) {
if (!webregged) {
ast_http_uri_link(&rawmanuri);
ast_http_uri_link(&manageruri);
ast_http_uri_link(&managerxmluri);
ast_http_uri_link(&arawmanuri);
ast_http_uri_link(&amanageruri);
ast_http_uri_link(&amanagerxmluri);
webregged = 1;
}
} else {
if (webregged) {
ast_http_uri_unlink(&rawmanuri);
ast_http_uri_unlink(&manageruri);
ast_http_uri_unlink(&managerxmluri);
ast_http_uri_unlink(&arawmanuri);
ast_http_uri_unlink(&amanageruri);
ast_http_uri_unlink(&amanagerxmluri);
webregged = 0;
}
}
if (newhttptimeout > 0) {
httptimeout = newhttptimeout;
}
ast_tcptls_server_start(&ami_desc);
if (tls_was_enabled && !ami_tls_cfg.enabled) {
ast_tcptls_server_stop(&amis_desc);
} else if (ast_ssl_setup(amis_desc.tls_cfg)) {
ast_tcptls_server_start(&amis_desc);
}
return 0;
}
static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub,
Multiple revisions 399887,400138,400178,400180-400181 ........ r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line Minor performance bump by not allocate manager variable struct if we don't need it ........ r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines Stasis performance improvements This patch addresses several performance problems that were found in the initial performance testing of Asterisk 12. The Stasis dispatch object was allocated as an AO2 object, even though it has a very confined lifecycle. This was replaced with a straight ast_malloc(). The Stasis message router was spending an inordinate amount of time searching hash tables. In this case, most of our routers had 6 or fewer routes in them to begin with. This was replaced with an array that's searched linearly for the route. We more heavily rely on AO2 objects in Asterisk 12, and the memset() in ao2_ref() actually became noticeable on the profile. This was #ifdef'ed to only run when AO2_DEBUG was enabled. After being misled by an erroneous comment in taskprocessor.c during profiling, the wrong comment was removed. Review: https://reviewboard.asterisk.org/r/2873/ ........ r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines Taskprocessor optimization; switch Stasis to use taskprocessors This patch optimizes taskprocessor to use a semaphore for signaling, which the OS can do a better job at managing contention and waiting that we can with a mutex and condition. The taskprocessor execution was also slightly optimized to reduce the number of locks taken. The only observable difference in the taskprocessor implementation is that when the final reference to the taskprocessor goes away, it will execute all tasks to completion instead of discarding the unexecuted tasks. For systems where unnamed semaphores are not supported, a really simple semaphore implementation is provided. (Which gives identical performance as the original taskprocessor implementation). The way we ended up implementing Stasis caused the threadpool to be a burden instead of a boost to performance. This was switched to just use taskprocessors directly for subscriptions. Review: https://reviewboard.asterisk.org/r/2881/ ........ r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines Optimize how Stasis forwards are dispatched This patch optimizes how forwards are dispatched in Stasis. Originally, forwards were dispatched as subscriptions that are invoked on the publishing thread. This did not account for the vast number of forwards we would end up having in the system, and the amount of work it would take to walk though the forward subscriptions. This patch modifies Stasis so that rather than walking the tree of forwards on every dispatch, when forwards and subscriptions are changed, the subscriber list for every topic in the tree is changed. This has a couple of benefits. First, this reduces the workload of dispatching messages. It also reduces contention when dispatching to different topics that happen to forward to the same aggregation topic (as happens with all of the channel, bridge and endpoint topics). Since forwards are no longer subscriptions, the bulk of this patch is simply changing stasis_subscription objects to stasis_forward objects (which, admittedly, I should have done in the first place.) Since this required me to yet again put in a growing array, I finally abstracted that out into a set of ast_vector macros in asterisk/vector.h. Review: https://reviewboard.asterisk.org/r/2883/ ........ r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines Remove dispatch object allocation from Stasis publishing While looking for areas for performance improvement, I realized that an unused feature in Stasis was negatively impacting performance. When a message is sent to a subscriber, a dispatch object is allocated for the dispatch, containing the topic the message was published to, the subscriber the message is being sent to, and the message itself. The topic is actually unused by any subscriber in Asterisk today. And the subscriber is associated with the taskprocessor the message is being dispatched to. First, this patch removes the unused topic parameter from Stasis subscription callbacks. Second, this patch introduces the concept of taskprocessor local data, data that may be set on a taskprocessor and provided along with the data pointer when a task is pushed using the ast_taskprocessor_push_local() call. This allows the task to have both data specific to that taskprocessor, in addition to data specific to that invocation. With those two changes, the dispatch object can be removed completely, and the message is simply refcounted and sent directly to the taskprocessor. Review: https://reviewboard.asterisk.org/r/2884/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30 18:55:27 +00:00
struct stasis_message *message)
{
if (stasis_message_type(message) != ast_named_acl_change_type()) {
return;
}
/* For now, this is going to be performed simply and just execute a forced reload. */
ast_log(LOG_NOTICE, "Reloading manager in response to ACL change event.\n");
__init_manager(1, 1);
}
static int unload_module(void)
{
return 0;
}
static int load_module(void)
{
ast_register_cleanup(manager_shutdown);
return __init_manager(0, 0) ? AST_MODULE_LOAD_FAILURE : AST_MODULE_LOAD_SUCCESS;
}
static int reload_module(void)
{
return __init_manager(1, 0);
}
int astman_datastore_add(struct mansession *s, struct ast_datastore *datastore)
{
AST_LIST_INSERT_HEAD(&s->session->datastores, datastore, entry);
return 0;
}
int astman_datastore_remove(struct mansession *s, struct ast_datastore *datastore)
{
return AST_LIST_REMOVE(&s->session->datastores, datastore, entry) ? 0 : -1;
}
struct ast_datastore *astman_datastore_find(struct mansession *s, const struct ast_datastore_info *info, const char *uid)
{
struct ast_datastore *datastore = NULL;
if (info == NULL)
return NULL;
AST_LIST_TRAVERSE_SAFE_BEGIN(&s->session->datastores, datastore, entry) {
if (datastore->info != info) {
continue;
}
if (uid == NULL) {
/* matched by type only */
break;
}
if ((datastore->uid != NULL) && !strcasecmp(uid, datastore->uid)) {
/* Matched by type AND uid */
break;
}
}
AST_LIST_TRAVERSE_SAFE_END;
return datastore;
}
int ast_str_append_event_header(struct ast_str **fields_string,
const char *header, const char *value)
{
if (!*fields_string) {
*fields_string = ast_str_create(128);
if (!*fields_string) {
return -1;
}
}
return (ast_str_append(fields_string, 0, "%s: %s\r\n", header, value) < 0) ? -1 : 0;
}
static void manager_event_blob_dtor(void *obj)
{
struct ast_manager_event_blob *ev = obj;
ast_string_field_free_memory(ev);
}
struct ast_manager_event_blob *
__attribute__((format(printf, 3, 4)))
ast_manager_event_blob_create(
int event_flags,
const char *manager_event,
const char *extra_fields_fmt,
...)
{
struct ast_manager_event_blob *ev;
va_list argp;
ast_assert(extra_fields_fmt != NULL);
ast_assert(manager_event != NULL);
ev = ao2_alloc_options(sizeof(*ev), manager_event_blob_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
if (!ev) {
return NULL;
}
if (ast_string_field_init(ev, 20)) {
ao2_ref(ev, -1);
return NULL;
}
ev->manager_event = manager_event;
ev->event_flags = event_flags;
va_start(argp, extra_fields_fmt);
ast_string_field_ptr_build_va(ev, &ev->extra_fields, extra_fields_fmt, argp);
va_end(argp);
return ev;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Asterisk Manager Interface",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.reload = reload_module,
.load_pri = AST_MODPRI_CORE,
.requires = "extconfig,acl,http",
);