3910 Commits (0747ac893b277dbcab384607e533732b8e6cc0ef)

Author	SHA1	Message	Date
Jenkins2	fabfe701bb	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge"	8 years ago
Joshua Colp	d6e1acd25e	Merge "app_confbridge / bridge_softmix: Add ability to configure REMB interval."	8 years ago
Jenkins2	070428415a	Merge "res_rtp_asterisk: Queue video update on picture loss indication."	8 years ago
Richard Mudgett	0c03eab962	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	8 years ago
Joshua Colp	8a602f18db	res_rtp_asterisk: Queue video update on picture loss indication. ... The previous payload specific feedback handling was very single minded in that it just assumed everything should trigger a video update. This was changed but the handling of picture loss indication was not added. The result was that video may not flow. This change adds it explicitly in. Change-Id: I1894be02e39ee10a0af841b5a1dca5f0ec7d60b6	8 years ago
Richard Mudgett	71a67a98c4	res_pjsip: Update authenticate_qualify documentation. ... Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4	8 years ago
Joshua Colp	0f6431e8e4	app_confbridge / bridge_softmix: Add ability to configure REMB interval. ... This change adds a configuration option to app_confbridge which can be used to set the interval at which we will send a combined REMB (remote estimated maximum bitrate) frame to sources of video. The bridging API has also been extended slightly to allow setting this so bridge_softmix can use it. ASTERISK-27786 Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82	8 years ago
Jenkins2	c66fde8247	Merge "BuildSystem: With external editline, do not require libs for internal editline."	8 years ago
Jenkins2	0718964e32	Merge "core: Create main/options.c."	8 years ago
Kevin Harwell	48ef239a01	Merge "res_rtp_asterisk: Add support for raising additional RTCP messages."	8 years ago
Ben Ford	138e0eff4e	Add data buffer API to store packets. ... Adds a data buffer with a configurable size that can store different kinds of packets (like RTP packets for retransmission). Given a number it will store a data packet at that position relative to the others. Given a number it will retrieve the given data packet if it is present. This is purposely a storage of arbitrary things so it can be used not just for RTP packets but also Asterisk frames in the future if needed. The API does not internally use a lock, so it will be up to the user of the API to properly protect the data buffer. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements Change-Id: Iff13c5d4795d52356959fe2a57360cd57dfade07	8 years ago
Joshua Colp	e14b0e960d	res_rtp_asterisk: Add support for raising additional RTCP messages. ... This change extends the existing AST_FRAME_RTCP frame type to be able to contain additional RTCP message types, such as feedback messages. The payload type is contained in the subclass which allows knowing what is in the frame itself. The RTCP feedback message type is now handled and REMB[1] messages are raised with their containing information. This also fixes a bug where all feedback messages were triggering video updates instead of just FIR and FUR. Finally RTCP frames are now passed up through the Asterisk core to what is handling the channel, mapped appropriately in the case of bridging, and written to an outgoing stream. Since RTCP frames are on a per-stream basis this is only done on multistream capable channels. [1] https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 ASTERISK-27758 ASTERISK-26366 Change-Id: I680da0ad8d5059d5e9655d896fb9d92e9da8491e	8 years ago
Jenkins2	a4a5b8d562	Merge "loader: Reserve space for additional pointers in ast_module_info."	8 years ago
Alexander Traud	d6fda173a4	BuildSystem: With external editline, do not require libs for internal editline. ... ASTERISK-27761 Change-Id: Ib17a7415297a210cfcdbf149e4df9b6edadbfab6	8 years ago
Corey Farrell	a6d58c518a	core: Create main/options.c. ... This creates a separate source to 'own' symbols related to options.h and paths.h. This significantly reduces the number of exports created by main/asterisk.o. This change is required to eventually be able to link unmodified Asterisk sources to utilities and/or stand-alone tests. ASTERISK~26245 Change-Id: I5cf184f4757f9363b80c9e678bdc35c477122380	8 years ago
Jenkins2	fa892d8dfd	Merge "core: Stop using AST_INLINE_API for allocator functions."	8 years ago
Jenkins2	39ba6aacf8	Merge "rtp: Add REMB RTP property and set it on PJSIP video RTP."	8 years ago
Corey Farrell	411915af28	loader: Reserve space for additional pointers in ast_module_info. ... This creates 4 reserved pointers in case we need additional dependency management fields. Change-Id: If991ec99b779df1b2dfbd38ce1a0cd79f9e01821	8 years ago
Joshua Colp	f03b984724	Merge "core: Remove additional symbols."	8 years ago
Jenkins2	60a06fbd1d	Merge "core: Remove dead symbols from asterisk.exports.in."	8 years ago
Jenkins2	f863f1f91b	Merge "channel.c: Allow generic plc then channel formats are equal"	8 years ago
Jenkins2	ffc20301fa	Merge "stringfields: Remove MALLOC_DEBUG fields from struct ast_string_field_mgr."	8 years ago
Joshua Colp	0bfc83ee4f	Merge "BuildSystem: Remove unused dependency on libltdl."	8 years ago
Corey Farrell	040bb21771	core: Remove additional symbols. ... Remove symbols that are depreacated and replaced: * ast_channel_datastore_alloc * ast_channel_datastore_free * ast_channel_cmpwhentohangup * ast_channel_setwhentohangup * config_text_file_save * devstate2str * ast_device_state_changed * ast_device_state_changed_literal * ast_verbose_get_by_module Remove unused symbols: * channelreloadreason2txt (last used in Asterisk 12). Remove unused ast_options flags: * AST_OPT_FLAG_END_CDR_BEFORE_H_EXTEN / ast_opt_end_cdr_before_h_exten * AST_OPT_FLAG_VERBOSE_MODULE / ast_opt_verb_module * AST_OPT_FLAG_INITIATED_SECONDS Change-Id: I841255995d195f8efc1ed47af9c7a2f131c08645	8 years ago
Corey Farrell	de77cf8698	core: Remove dead symbols from asterisk.exports.in. ... * dahdi_chan_name * dahdi_chan_name_len * dahdi_chan_mode * __manager_event * dialed_interface_info Added comment about __progname and environ being needed for FreeBSD to prevent accidental removal in the future. Change-Id: I3ae026bc541cd9cb572be2ffa95fc359547642b5	8 years ago
George Joseph	5d097f8236	channel.c: Allow generic plc then channel formats are equal ... If the two formats on a channel are equal, we don't transcode and since the generic plc needs slin to work, it doesn't get invoked. * A new configuration option "genericplc_on_equal_codecs" was added to the "plc" section of codecs.conf to allow generic packet loss concealment even if no transcoding was originally needed. Transcoding via SLIN is forced in this case. ASTERISK-27743 Change-Id: I0577026a179dea34232e63123254b4e0508378f4	8 years ago
Jenkins2	5843a19797	Merge "loader: Convert reload_classes to built-in modules."	8 years ago
Joshua Colp	115939caeb	rtp: Add REMB RTP property and set it on PJSIP video RTP. ... This change adds a property to RTP instances to indicate that REMB support is enabled and that sending/receiving should be passed through. This also enables it on video RTP instances in PJSIP if WebRTC support is enabled. Finally the goog-remb extension is added to the SDP using the rtcp-fb attribute to indicate our support for it. Details about REMB can be found on the draft document for it: https://tools.ietf.org/html/draft-alvestrand-rmcat-remb-03 Change-Id: I1902dda1c0882bd1a0d71b2f120684b44b97e789	8 years ago
Alexander Traud	10a978829e	BuildSystem: Remove unused dependency on libltdl. ... Asterisk does not need the development package of libltdl, because it does not use any symbol of -lltdl directly. Instead, it uses the runtime package via the shared library -lodbc. On the supported platforms, that shared library declares its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have failed. ASTERISK-27745 Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba	8 years ago
Corey Farrell	4d1c9d8711	core: Stop using AST_INLINE_API for allocator functions. ... This replaces AST_INLINE_API allocators in utils.h with real functions implemented in astmm.c. Associated macro's are also moved from utils.h to astmm.h. Remove menuselect conflicts between MALLOC_DEBUG and DEBUG_CHAOS as they can now be combined. This has multiple benefits: * Simplifies asterisk/utils.h by removing inline functions and use of the logger. * Removal of these inline functions decreases size of Asterisk and module binaries by 1% or more. * Puts memory management functions together with and without MALLOC_DEBUG enabled, simplifying management of the code. * Enables DEBUG_CHAOS for ASTMM_REDIRECT and bundled pjproject. Change-Id: If9df4377f74bdbb627461b27a473123e05525887	8 years ago
Jenkins2	1a9ed413a1	Merge "core: Remove incorrect usage of attribute_malloc."	8 years ago
Jenkins2	15bf51af73	Merge "core: Remove non-critical cleanup from startup aborts."	8 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	8 years ago
Jenkins2	200f7c520c	Merge "BuildSystem: Enable PortAudio in NetBSD."	8 years ago
Corey Farrell	9e488dd482	core: Remove incorrect usage of attribute_malloc. ... GCC documentation states that when __attribute__((malloc)) is used it should not return storage which contains any valid pointers. It specifically mentions that realloc functions should not have the malloc attribute, but this also means that complex initializers which could contain initialized pointers should not use this attribute. Change-Id: If507f33ffb3ca3b83b702196eb0e8215d27fc7d2	8 years ago
Corey Farrell	ea9768ff07	stringfields: Remove MALLOC_DEBUG fields from struct ast_string_field_mgr. ... This causes MALLOC_DEBUG reporting to be slightly different, calls which cause additional memory pools to be allocated now report the callers location rather than the location which originally allocated the string field structure. This reduces storage needed by string fields and allows MALLOC_DEBUG to identify the source of additional allocations rather than obscuring it by reporting the original allocation caller. Change-Id: Idd18e6639a87ab862079b580c114d90361412289	8 years ago
Jenkins2	4b7872c9db	Merge "core: Remove ABI effects of MALLOC_DEBUG."	8 years ago
Corey Farrell	fee929c8ac	core: Remove non-critical cleanup from startup aborts. ... When built-in components of Asterisk fail to start they cause the Asterisk startup to abort. In these cases only the most critical cleanup should be performed - closing databases and terminating proceses. These cleanups are registered using ast_register_atexit, all other cleanups should not be run during startup abort. The main reason for this change is that these cleanup procedures are untestable from the partially initialized states, if they fail it could prevent us from ever running the critical cleanup with ast_run_atexits. Create separate initialization for dns_core.c to be run unconditionally during startup instead of being initialized by the first dns resolver to be registered. This ensures that 'sched' is initialized before it can be potentially used. Replace ast_register_atexit with ast_register_cleanup in media_cache.c. There is no reason for this cleanup to happen unconditionally. Change-Id: Iecc2df98008b21509925ff16740bd5fa29527db3	8 years ago
Alexander Traud	16f6e94033	BuildSystem: Enable PortAudio in NetBSD. ... In NetBSD, PortAudio 1 is still the default version. PortAudio 2 can be installed side by side but gets placed in a 'portaudio2' subdirectory. To find PortAudio 2 even in a subdirectory, the tool pkg-config is queried via AST_PKG_CONFIG_CHECK. For those platforms, which do not list PowerAudio 2 via pkg-config, the previous check remains and is executed thereafter. ASTERISK-27721 Change-Id: I4175500126909ad1b181fff8e11bb4a3a6ae4fa9	8 years ago
Corey Farrell	add03e207c	app_osplookup: Move header defines into the app. ... astosp.h is leftover from when logic was split between app_osplookup and res_osp. All logic was moved into app_osplookup by 109737eb1c in 2006, but astosp.h remained. This moves the remaining defines into app_osplookup and deletes astosp.h. Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd	8 years ago
Jenkins2	e21f168477	Merge "BuildSystem: Detect whether uselocale(.) is available."	8 years ago
Alexander Traud	9749524520	BuildSystem: Detect whether uselocale(.) is available. ... ASTERISK-27712 Reported by: Joerg Sonnenberger, D'Arcy Cain Change-Id: Idf1c9d43617a3e13028b95b313415903d80ef807	8 years ago
Richard Mudgett	c711e4076a	core: Remove ABI effects of MALLOC_DEBUG. ... This allows asterisk to be compiled with MALLOC_DEBUG to load modules built without MALLOC_DEBUG. Now pre-compiled third-party modules will still work regardless of MALLOC_DEBUG being enabled or not. Change-Id: Ic07ad80b2c2df894db984cf27b16a69383ce0e10	8 years ago
Richard Mudgett	1a36a452bd	pjproject: Add cache_pools debugging option. ... The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	8 years ago
Ben Ford	0be1c388e4	Add extended properties to rtp_engine for RTP retransmission support. ... A couple of additional properties are needed in rtp_engine to enable support for packet retransmission: AST_RTP_PROPERTY_RETRANS_RECV and AST_RTP_PROPERTY_RETRANS_SEND. These will both be enabled automatically if an endpoint has the webrtc option enabled. While this adds no functionality currently, it will serve as a building block for future changes for RTP retransmission support. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements Change-Id: Ic598acd042a045f9d10e5bdccb66f4efc9e587cc	8 years ago
Alexander Traud	8ac198aff3	BuildSystem: Remove chan_h323 leftovers. ... ASTERISK-27670 Change-Id: I07a8ef8bbd6001e25711fa1bff152eb6c9efa729	8 years ago
Jenkins2	f041bc7863	Merge "app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs."	8 years ago
George Joseph	b148453708	Merge "res_pjsip_pubsub: Prune subs with reliable transports at startup"	8 years ago
Jenkins2	7a6db221ab	Merge "core: Create ast_atomic macro's."	8 years ago
Richard Mudgett	b9024197ab	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. ... The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	8 years ago