404 Commits (4da293459297ed66fbb85aa53e328bd560e73d17)

Author	SHA1	Message	Date
Rodrigo Ramírez Norambuena	94532b2c22	main/asterisk.c: Update Asterisk copyright year ... Change-Id: I5e75d7f7e2c096d74edd9e8735268a894f4b93ab	11 years ago
Corey Farrell	8f3cee1258	Astobj2: Fix initialization order of refdebug and AO2_DEBUG. ... This ensures that refdebug is initialized before AO2_DEBUG if both are enabled, since AO2_DEBUG allocates a container. This change also makes AO2_DEBUG initialization critical, a failure will abort Asterisk startup. This is needed since the failure would be caused by reg_containers allocation failure, and that would result in a segmentation fault by ao2_container_register later in startup. ASTERISK-25048 #close Reported by: Corey Farrell Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244	11 years ago
Corey Farrell	5c1d07baf0	Astobj2: Allow reference debugging to be enabled/disabled by config. ... * The REF_DEBUG compiler flag no longer has any effect on code that uses Astobj2. It is used to determine if reference debugging is enabled by default. Reference debugging can be enabled or disabled in asterisk.conf. * Caller information is provided in logger errors for ao2 bad magic numbers. * Optimizes AO2 by merging internal functions with the public counterpart. This was possible now that we no longer require a dual ABI. ASTERISK-24974 #close Reported by: Corey Farrell Change-Id: Icf3552721fe999365ba8a8cf00a965aa6b897cc1	11 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	11 years ago
George Joseph	9a63ada03a	loader/main: Don't set ast_fully_booted until deferred reloads are processed ... Until we have a true module management facility it's sometimes necessary for one module to force a reload on another before its own load is complete. If Asterisk isn't fully booted yet, these reloads are deferred. The problem is that asterisk reports fully booted before processing the deferred reloads which means Asterisk really isn't quite ready when it says it is. This patch moves the report of fully booted after the processing of the deferred reloads is complete. Since the pjsip stack has the most number of related modules, I ran the channels/pjsip testsuite to make sure there aren't any issues. All tests passed. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4604/ ........ Merged revisions 434544 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	3ddd92902a	Replace most uses of ast_register_atexit with ast_register_cleanup. ... Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433497 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	66670f02e6	Fix compilation issues for OpenBSD ... This patch addresses compilation issues for OpenBSD. Specifically, it addresses: * It allows including <sys/vmmeter.h> in asterisk.c * Provides a needed (size_t) cast in xmldoc.c In 13+, it also addresses a conditional inclusion in loader.c. Review: https://reviewboard.asterisk.org/r/4506 ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy patches: misc-openbsd.diff uploaded by snuffy (License 5024) ........ Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 433247 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433248 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	8c65c9167e	Various: bugfixes found via chaos ... Using DEBUG_CHAOS several instances of a null pointer crash, and one uninitialized variable were uncovered and fixed. Also added details on why Asterisk failed to initialize. Review: https://reviewboard.asterisk.org/r/4468/ ........ Merged revisions 433064 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433065 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	bb71672a47	main/asterisk.c: Reverse #if statement in listener() to fix code folding. ... listener() opens the same code block in two places (#if and #else). This confuses some folding editors causing it to think that an extra code block was opened. Folding in 'geany' causes all code after listener() to be folded as if it were part of that procedure. ASTERISK-24813 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4435/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e2d3215b83	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
David M. Lee	965777ccfc	Various fixes for OS X ... This patch addresses compilation errors on OS X. It's been a while, so there's quite a few things. * Fixed __attribute__ decls in route.h to be portable. * Fixed htonll and ntohll to work when they are defined as macros. * Replaced sem_t usage with our ast_sem wrapper. * Added ast_sem_timedwait to our ast_sem wrapper. * Fixed some GCC 4.9 warnings using sig*set() functions. * Fixed some format strings for portability. * Fixed compilation issues with res_timing_kqueue (although tests still fail on OS X). * Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue on OS X). ASTERISK-24539 #close Reported by: George Joseph ASTERISK-24544 #close Reported by: George Joseph Review: https://reviewboard.asterisk.org/r/4327/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	bbd9ff122e	queue_log: Post QUEUESTART entry when Asterisk fully boots. ... The QUEUESTART log entry has historically acted like a fully booted event for the queue_log file. When the QUEUESTART entry was posted to the log was broken by the change made by ASTERISK-15863. * Made post the QUEUESTART queue_log entry when Asterisk fully boots. This restores the intent of that log entry and happens after realtime has had a chance to load. AST-1444 #close Reported by: Denis Martinez Review: https://reviewboard.asterisk.org/r/4282/ ........ Merged revisions 430009 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 430010 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	8fe45f0f0a	core: avoid possible asterisk -r crash from long id ... When connecting to the remote console, an id string is first provided that consts of the hostname, pid, and version. This is parsed by the remote instance using a buffer that may be too short, and can allow a buffer overrun because it is not terminated. This patch adds termination and a larger buffer. Review: https://reviewboard.asterisk.org/r/4182/ ........ Merged revisions 429223 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@429224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	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	11 years ago
Sean Bright	a1eec851c6	Update Asterisk copyright year in main/asterisk.c ... It's been 2014 for like... 6 months. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	97834718c2	Remove many deprecated modules ... Billing records are fair, To get paid is quite bright, You should really use ODBC; Good-bye cdr_sqlite. Microsoft did once push H.323, Hell, we all remember NetMeeting. But try to compile chan_h323 now And you will take quite a beating. The XMPP and SIP war was fierce, And in the distant fray Was birthed res_jabber/chan_jingle; But neither to stay. For everyone did care and chase what Google professed. "Free Internet Calling" was what devotees cried, But Google did change the specs so often That the developers were happy the day chan_gtalk died. And then there was that odd application Dedicated to the Polish tongue. app_saycountpl was subsumed by Say; One could say its bell was rung. To read and parse a file from the dialplan You could (I guess) use an application. app_readfile did fill that purpose, but I think A function is perhaps better in its creation. Barging is rude, I'm not sure why we do it. Inwardly, the caller will probably sigh. But if you really must do it, Don't use app_dahdibarge, use ChanSpy. We all despise the sound of tinny robots It makes our queues so cold. To control such an abomination It's better to not use Wait/SetMusicOnHold. It's often nice to know properties of a channel It makes our calls right We have a nice function called CHANNEL And so SIPCHANINFO is sent off into the night. And now things get odd; Apparently one could delimit with a colon Properties from the SIPPEER function! Commas are in; all others are done. Finally, a word on pipes and commas. We're sorry. We can't say it enough. But those compatibility options in asterisk.conf; To maintain them forever was just too tough. This patch removes: * cdr_sqlite * chan_gtalk * chan_jingle * chan_h323 * res_jabber * app_saycountpl * app_readfile * app_dahdibarge It removes the following applications/functions: * WaitMusicOnHold * SetMusicOnHold * SIPCHANINFO It removes the colon delimiter from the SIPPEER function. Finally, it also removes all compatibility options that were configurable from asterisk.conf, as these all applied to compatibility with Asterisk 1.4 systems. Review: https://reviewboard.asterisk.org/r/3698/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fb5690ce4b	Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages ... This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9653c6d357	main/asterisk: Fix startup sequence for realtime features ... When ASTERISK-23265/ASTERISK-23320 was fixed, it inadvertently led to realtime features breaking. This was due to features loading prior to realtime. This patch fixes this by loading features after loading dynamic modules. ASTERISK-23487 #close Reported by: Denis Tested by: Denis ........ Merged revisions 412698 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412699 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4f30c7e91f	main/astobj2: Make REF_DEBUG a menuselect item; improve REF_DEBUG output ... This patch does the following: (1) It makes REF_DEBUG a meneselect item. Enabling REF_DEBUG now enables REF_DEBUG globally throughout Asterisk. (2) The ref debug log file is now created in the AST_LOG_DIR directory. Every run will now blow away the previous run (as large ref files sometimes caused issues). We now also no longer open/close the file on each write, instead relying on fflush to make sure data gets written to the file (in case the ao2 call being performed is about to cause a crash) (3) It goes with a comma delineated format for the ref debug file. This makes parsing much easier. This also now includes the thread ID of the thread that caused ref change. (4) A new python script instead for refcounting has been added in the contrib/scripts folder. (5) The old refcounter implementation in utils/ has been removed. Review: https://reviewboard.asterisk.org/r/3377/ ........ Merged revisions 412114 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412115 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412153 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5ca5d42646	Internal timing: Add notice that the -I and internal_timing option are no longer needed. ... Add notice messages during execution that the -I command line option and the astersik.conf internal_timing option are no longer needed. The internal timing functionality is now always enabled if there is a timing module loaded. NOTE: Since the command line options and the asterisk.conf config file are processed before the logging system is initialized, the messages are output to stderr. Change requested as a result of asterisk-dev list comments about the commit for ASTERISK-22846 that removed the -I and internal_timing options. Review: https://reviewboard.asterisk.org/r/3423/ ........ Merged revisions 411964 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411974 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	03beadb6e9	internal_timing: Remove the option and always make it enabled if a timing module is loaded. ... The masquerade supertest frequently fails because either the local channel chain doesn't completely optimize out or the DTMF handshake doesn't completely get accross. Local channel optimization requires frames flowing to trigger when optimization can happen. When optimization happens the media frame that triggered the optimization is dropped. Sending DTMF requires frames to flow in the other direction for timing purposes while sending nothing. If internal timing is not enabled when MOH is playing, Asterisk switches to received timing when an audio frame is received. With optimization dropping media frames and MOH not sending frames unless it receives frames, occasionaly there are no more frames being passed and the test fails. * The asterisk command line -I option and the asterisk.conf internal_timing option are removed. Asterisk now always uses internal timing when needed if any timing module is loaded. The issue ASTERISK-14861 did this quite awhile ago in v1.4 but effectively is broken if other internal timing modules besides DAHDI are used. The ast_read_generator_actions() now only does received timing if it has no choice for frame generators like MOH, silence, and playback streaming. * Cleaned up some code dealing with frame generators in ast_deactivate_generator(), generator_write_format_change(), ast_activate_generator(), and ast_channel_stop_silence_generator(). * Removed ast_internal_timing_enabled(), AST_OPT_FLAG_INTERNAL_TIMING, and ast_opt_internal_timing. ASTERISK-22846 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3414/ ........ Merged revisions 411715 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411716 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411717 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411724 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	43858c24ab	doxygen: Tweak the link back to ye olde Digium website ... ........ Merged revisions 409361 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409362 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409363 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	31707b1d69	main: Initialize dialplan providing core components prior to module pre-load ... It is possible to pre-load pbx_config. As a result, pbx_config - which will load and parse the dialplan - will attempt to use various dialplan components, such as device state providers and presence state providers, prior to them being initialized by the core. This would lead to a crash, as the components had not created their Stasis cache entries. This patch moves a number of core component initializations before the module pre-load. This guarantees that if someone does pre-load pbx_config - or other pbx modules - that the Stasis caches for the various core components are created. (closes issue ASTERISK-23320) Reported by: xrobau (closes issue ASTERISK-23265) Reported by: Andrew Nagy Tested by: Andrew Nagy, Rusty Newton ........ Merged revisions 408855 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	828f339a9c	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	12 years ago
Tzafrir Cohen	3eee3f21ad	asterisk.c: suppress live_dangerously warning on rasterisk ... Even since the fixes of AST-2013-007, Asterisk prints the following warning on startup if the user decided to live dangerously: Privilege escalation protection disabled! See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details. This message is intended for the logs and interactive startup. No need for it to appear on a remote console. This commit removes it from there. (closes issue ASTERISK-23084) Review: https://reviewboard.asterisk.org/r/3101/ ........ Merged revisions 404861 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 404888 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 404911 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	27f37f6e3d	Changed the default for live_dangerously to no ... ........ Merged revisions 404006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	744556c01d	security: Inhibit execution of privilege escalating functions ... This patch allows individual dialplan functions to be marked as 'dangerous', to inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. Also, the ABI was changed to something more reasonable, since Asterisk 12 does not yet have a public release. (closes issue ASTERISK-22905) Review: http://reviewboard.digium.internal/r/432/ ........ Merged revisions 403913 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 403917 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 403959 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403960 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	2564ed26f7	app_dahdiras: Use waitpid instead of wait4. ... Several places in the code were using wait4 while other places were using waitpid. This change makes all places use waitpid in order to make things more consistent and since the 'rusage' object passed in/out of wait4 was never used. (closes issue ASTERISK-22557) Reported by: YvesGael Patches: asterisk-11.5.1-wait4.patch uploaded by hurdman (license 6537) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402546 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4ca0f222e8	memory leaks: Memory leak cleanup patch by Corey Farrell (second set) ... Also covers ast_app_parse_timelen-fail-zero-length.patch, but the patch was replaced with one of my own. (issue ASTERISK-22467) Reported by: Corey Farrell Patches: chan_dahdi-cleanup_push.patch uploaded by coreyfarrell (license 5909) clicompat-r2.patch uploaded by coreyfarrell (license 5909) codecs-ilbc-doCPLC.patch uploaded by coreyfarrell (license 5909) data-cleanup-test-registration.patch uploaded by coreyfarrell (license 5909) main-asterisk-kill-listener.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 401704 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 401705 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 401706 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401707 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8d7873b836	ARI: Add subscription support ... This patch adds an /applications API to ARI, allowing explicit management of Stasis applications. * GET /applications - list current applications * GET /applications/{applicationName} - get details of a specific application * POST /applications/{applicationName}/subscription - explicitly subscribe to a channel, bridge or endpoint * DELETE /applications/{applicationName}/subscription - explicitly unsubscribe from a channel, bridge or endpoint Subscriptions work by a reference counting mechanism: if you subscript to an event source X number of times, you must unsubscribe X number of times to stop receiveing events for that event source. Review: https://reviewboard.asterisk.org/r/2862 (issue ASTERISK-22451) Reported by: Matt Jordan ........ Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	667fa56b1b	Remote console: more output discrepancies ... The remote console continued to have issues with its output. In this case CLI command output would either not show up (if verbose level = 0) or would contain verbose prefixes (if verbose level > 0) once log messages were sent to the remote console. The fix now now adds verbose prefix data to all new lines contained in a verbose log string. (closes issue ASTERISK-22450) Reported by: David Brillert (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2825/ ........ Merged revisions 399267 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399268 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	83bf017db9	Fix incorrect usages of ast_realloc(). ... There are several locations in the code base where this is done: buf = ast_realloc(buf, new_size); This is going to leak the original buf contents if the realloc fails. Review: https://reviewboard.asterisk.org/r/2832/ ........ Merged revisions 398757 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 398758 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398759 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	2d1d5a98d5	Fix graceful shutdown crash. ... The cleanup code for optional_api needs to happen after all of the optional API users and providers have unused/unprovided. Unfortunately, regsitering the atexit() handler at the beginning of main() isn't soon enough, since module destructors run after that. ........ Merged revisions 398149 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1d3d6e0661	Check return value on fwrite ... ........ Merged revisions 398000 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 398002 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@398003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	9bed50db41	optional_api: Fix linking problems between modules that export global symbols ... With the new work in Asterisk 12, there are some uses of the optional_api that are prone to failure. The details are rather involved, and captured on [the wiki][1]. This patch addresses the issue by removing almost all of the magic from the optional API implementation. Instead of relying on weak symbol resolution, a new optional_api.c module was added to Asterisk core. For modules providing an optional API, the pointer to the implementation function is registered with the core. For modules that use an optional API, a pointer to a stub function, along with a optional_ref function pointer are registered with the core. The optional_ref function pointers is set to the implementation function when it's provided, or the stub function when it's now. Since the implementation no longer relies on magic, it is now supported on all platforms. In the spirit of choice, an OPTIONAL_API flag was added, so we can disable the optional_api if needed (maybe it's buggy on some bizarre platform I haven't tested on) The AST_OPTIONAL_API*() macros themselves remained unchanged, so existing code could remain unchanged. But to help with debugging the optional_api, the patch limits the #include of optional API's to just the modules using the API. This also reduces resource waste maintaining optional_ref pointers that aren't used. Other changes made as a part of this patch: * The stubs for http_websocket that wrap system calls set errno to ENOSYS. * res_http_websocket now properly increments module use count. * In loader.c, the while() wrappers around dlclose() were removed. The while(!dlclose()) is actually an anti-pattern, which can lead to infinite loops if the module you're attempting to unload exports a symbol that was directly linked to. * The special handling of nonoptreq on systems without weak symbol support was removed, since we no longer rely on weak symbols for optional_api. [1]: https://wiki.asterisk.org/wiki/x/wACUAQ (closes issue ASTERISK-22296) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2797/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	d7b9a702d8	Verbose logging discrepancies ... Refactored cases where a combination of ast_verbose/options_verbose were present. Also in general tried to eliminate, in as many places as possible, where the options_verbose global variable was being used. Refactored the way local and remote consoles handle verbose message logging in an attempt to solve the various discrepancies that sometimes would show between the two. (closes issue AST-1193) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/2798/ ........ Merged revisions 397948 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 397958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397959 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	e1cfc18a78	Memory leaks fix ... (closes ASTERISK-22376) Reported by: John Hardin Patches: memleak.patch uploaded by jhardin (license 6512) memleak2.patch uploaded by jhardin (license 6512) ........ Merged revisions 397946 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	dd33217762	Add the bucket API. ... Bucket is a URI based API for the creation, retrieval, updating, and deletion of "buckets" and files contained within them. Review: https://reviewboard.asterisk.org/r/2715/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	28e9d3afc9	Don't store repeated commands in the editline history buffer. ... The equivalent of bash HISTCONTROL=ignoredups. Review: https://reviewboard.asterisk.org/r/2775/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d7f1f31270	Refactor CEL to avoid using the event system core ... This removes usage of the event system for CEL backend data distribution and strips unused pieces out of the event system. Review: https://reviewboard.asterisk.org/r/2732/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	bd352e0827	Remove leading spaces from the CLI command before parsing ... If you've mistakenly put a space before typing in a command, the leading space will be included as part of the command, and the command parser will not find the corresponding command. This patch rectifies that situation by stripping the leading spaces on commands. Review: https://reviewboard.asterisk.org/r/2709/ Patch-by: Tilghman Lesher ........ Merged revisions 396745 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396746 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	357b275239	Fix res_ari_asterisk load issue ... The new res_ari_asterisk.so module presents several config options from asterisk main. Unfortunately, they aren't exported, so the module won't load on Linux. This patch renames the variables, adding the ast_ prefix so they will be exported. Review: https://reviewboard.asterisk.org/r/2737 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	38236e54a8	Remove dead code from features.c; refactor pickup code into pickup.c ... This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	715d894d48	Update copyright year to 2013 in asterisk.c; some whitespace fixes ... (closes issue ASTERISK-22179) Reported by: Malcolm Davenport ........ Merged revisions 395032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 395033 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c3b8939be8	Add CEL local optimization record type ... This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent local channel optimizations. Local channel optimizations were one of several things conveyed by the now defunct BRIDGE_UPDATE event type. This also adds a unit test to test generation of this new CEL event. Review: https://reviewboard.asterisk.org/r/2676/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	7422581b6d	Move channel driver Registry manager events to core. ... This also shuffles the stasis system topic and related handling. (closes issue ASTERISK-21488) Review: https://reviewboard.asterisk.org/r/2631/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d0a55fa52d	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	12 years ago
David M. Lee	a75fd32212	ARI - channel recording support ... This patch is the first step in adding recording support to the Asterisk REST Interface. Recordings are stored in /var/spool/recording. Since recordings may be destructive (overwriting existing files), the API rejects attempts to escape the recording directory (avoiding issues if someone attempts to record to ../../lib/sounds/greeting, for example). (closes issue ASTERISK-21594) (closes issue ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	c4adaf9106	Configuration for Stasis threadpool ... The appropriate settings for the Stasis threadpool is very system specific, depending upon both workload and system configuration. This patch adds a stasis.conf file which can be used to configure the key attributes of the threadpool for the Stasis message bus. (closes issue ASTERISK-21280) Review: https://reviewboard.asterisk.org/r/2651/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago