asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	49aaabbd7e	Merge "rtp: Add REMB RTP property and set it on PJSIP video RTP." into 15	8 years ago
Joshua Colp	86ac447646	Merge "core: Remove dead symbols from asterisk.exports.in." into 15	8 years ago
Joshua Colp	f062657287	Merge "channel.c: Allow generic plc then channel formats are equal" into 15	8 years ago
Corey Farrell	3abb3bb834	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	5de4dae6dd	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
Joshua Colp	dfb2ebe72e	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	57ed69dcd4	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
Jenkins2	8a2abfccc4	Merge "core: Remove incorrect usage of attribute_malloc." into 15	8 years ago
Jenkins2	1e9a0d9285	Merge "core: Remove non-critical cleanup from startup aborts." into 15	8 years ago
Corey Farrell	472b2835c9	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	06e84e1c42	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	2231e1a04d	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
Jenkins2	b8b6fca9c1	Merge "BuildSystem: Detect whether uselocale(.) is available." into 15	8 years ago
Alexander Traud	e30f075f48	BuildSystem: Detect whether uselocale(.) is available. ASTERISK-27712 Reported by: Joerg Sonnenberger, D'Arcy Cain Change-Id: Idf1c9d43617a3e13028b95b313415903d80ef807	8 years ago
Richard Mudgett	f6d5bf99f2	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
Joshua Colp	cc976d1e76	Merge "Add extended properties to rtp_engine for RTP retransmission support." into 15	8 years ago
Ben Ford	8d707d33ab	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
Kevin Harwell	483c5f7add	AMI: Bumping AMI non-breaking number for Asterisk 15.3.0 release A few changes were made to AMI: * Fixed "(null):" header in AMI AsyncAGIEnd event * A mute header was added to the ConfbridgeJoin AMI event * ConfbridgeList action's ConfbridgeList events now output all the standard channel snapshot headers Change-Id: I583817e2f3d782aeb4300665250362ef7b59a375	8 years ago
Alexander Traud	226b70c711	BuildSystem: Remove chan_h323 leftovers. ASTERISK-27670 Change-Id: I07a8ef8bbd6001e25711fa1bff152eb6c9efa729	8 years ago
Jenkins2	da3bcd4feb	Merge "app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs." into 15	8 years ago
George Joseph	90e5bae6ca	Merge "res_pjsip_pubsub: Prune subs with reliable transports at startup" into 15	8 years ago
Richard Mudgett	447fc8cc0a	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
Jenkins2	dbbd1211f2	Merge "loader: Use ast_cli_completion_add for 'module load' completion." into 15	8 years ago
Jenkins2	642b339cd8	Merge "pbx_variables.c: Misc fixes in variable substitution." into 15	8 years ago
George Joseph	af2fedb6ba	res_pjsip_pubsub: Prune subs with reliable transports at startup In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	8 years ago
Corey Farrell	f1975769a7	loader: Use ast_cli_completion_add for 'module load' completion. This addresses all performance issues with 'module load' completion. In addition to using ast_cli_completion_add we stop using libedit's filename_completion_function, instead using ast_file_read_dir. This ensures all results are produced from a single call to opendir. Change-Id: I8bf51ffaa7ef1606f3bd1b5bb13f1905d72c6134	8 years ago
Alexander Traud	83dfe4d160	headers: Consistent use of typeof and/or __typeof__. Because of a copy-and-paste error, the Asterisk project was using __typeof instead of typeof. It works because typeof, __typeof, and __typeof__ are supported by GCC, but here the escaped variant was not intended. Therefore, for consistence, we change this to typeof. Change-Id: I2a962c3e596e882f691a19345445b14571a5f07c	8 years ago
Richard Mudgett	7db4606235	pbx_variables.c: Misc fixes in variable substitution. * Copy more than one character at a time when there is nothing to substitute. * Fix off by one error if a '}' or ']' is missing. * Eliminated the requirement that the "used" parameter had to point to a variable. The current callers were always declaring a variable to meet the requirement and discarding the value put into that variable. Now it can be NULL. * In ast_str_substitute_variables_full() fixed using the bogus channel to evaluate a function. We were not using the bogus channel we just created to help evaluate a subexpression. Change-Id: Ia83d99f4f16abe47f329eb39b6ff2013ae7c9854	8 years ago
Richard Mudgett	821b718a96	res_pjsip: Split type=identify to IP address and SIP header matching priorities The type=identify endpoint identification method can match by IP address and by SIP header. However, the SIP header matching has limited usefulness because you cannot specify the SIP header matching priority relative to the IP address matching. All the matching happens at the same priority and the order of evaluating the identify sections is indeterminate. e.g., If you had two type=identify sections where one matches by IP address for endpoint alice and the other matches by SIP header for endpoint bob then you couldn't predict which endpoint is matched when a request comes in that matches both. * Extract the SIP header matching criteria into its own "header" endpoint identification method so the user can specify the relative priority of the SIP header and the IP address matching criteria in the global endpoint_identifier_order option. The "ip" endpoint identification method now only matches by IP address. ASTERISK-27491 Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095	8 years ago
Corey Farrell	94b4bf1d2a	astobj2: Create case-insensitive variants of container function macros. * AO2_STRING_FIELD_CASE_HASH_FN * AO2_STRING_FIELD_CASE_CMP_FN * AO2_STRING_FIELD_CASE_SORT_FN Change-Id: I11af8c6a0c43380a42732553f519c667abb842cf	8 years ago
Richard Mudgett	b53d1ce03e	core: Remove unused/incomplete SDP modules from released branches. Change-Id: Icc28fbdc46f58e54a21554e6fe8b078f841b1f86	8 years ago
Sean Bright	b64c02e414	Remove as much trailing whitespace as possible. Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Corey Farrell	70cd4430a4	Fix Common Typo's. Fix instances of: * Retreive * Recieve * other then * different then * Repeated words ("the the", "an an", "and and", etc). * othterwise, teh ASTERISK-24198 #close Change-Id: I3809a9c113b92fd9d0d9f9bac98e9c66dc8b2d31	8 years ago
Richard Mudgett	5c6a217d61	manager.h: Bump AMI version Change-Id: I62e6ddeb261ef012687e1fb6734c554e2499b6bf	8 years ago
Joshua Colp	bac31e55bb	Merge changes from topic 'faster-aco' into 15 * changes: aco: Minimize use of regex. aco: Create ways to minimize use of regex.	8 years ago
George Joseph	fb6aad5301	Merge "loader: Replace priority heap with vector." into 15	8 years ago
Corey Farrell	77af356e30	loader: Replace priority heap with vector. This is needed for future changes which will require being able to process the load priority out of order. Change-Id: Ia23421197f09789940510b03ebbbf3bf24d51bea	8 years ago
Corey Farrell	5d49e2e329	aco: Create ways to minimize use of regex. ACO uses regex in many situations where it is completely unneeded. In some cases this doubles the total processing performed by aco_process_config. * Create ACO_IGNORE category type for use in place of skip_category regex source string. * Create additional aco_category_op values to allow specifying category filter using either a single plain string or a NULL terminated array of plain strings. * Create ACO_PREFIX to allow matching option names to case insensitive prefixes. Change-Id: I66a920dcd8e2b0301f73f968016440a985e72821	8 years ago
Richard Mudgett	39c8d566ad	res_rtp_asterisk.c: Disable packet flood detection for video streams. We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
Richard Mudgett	c0822c32ac	chan_pjsip/res_pjsip: Add CHANNEL(pjsip,request_uri) This patch does three things associated with the initial incoming INVITE request URI. 1) Add access to the full initial incoming INVITE request URI. 2) We were not setting DNID on incoming PJSIP channels. The DNID is the user portion of the initial incoming INVITE Request-URI. The value is accessed by reading CALLERID(dnid). 3) Fix CHANNEL(pjsip,target_uri) documentation. * The initial incoming INVITE request URI is now available using CHANNEL(pjsip,request_uri). * Set the DNID on PJSIP channel creation so CALLERID(dnid) can return the initial incoming INVITE request URI user portion. * CHANNEL(pjsip,target_uri) now correctly documents that the target URI is the contact URI. * Refactored print_escaped_uri() out of channel_read_pjsip() to handle pjsip_uri_print() error condition when the buffer is too small. ASTERISK-27478 Change-Id: I512e60d1f162395c946451becb37af3333337b33	8 years ago
Jenkins2	5fb14aab28	Merge "astdb: Improve prefix searches in astdb" into 15	8 years ago
Sean Bright	d9843c84ef	astdb: Improve prefix searches in astdb Using the LIKE operator requires a full table scan of 'astdb', whereas a comparison operation is able to use the primary key index. This patch adds a new function to the AstDB API for quick prefix matches and updates res_sorcery_astdb to utilize it. This showed substantial performance improvement in my test environment. Related to ASTERISK~26806, but does not completely resolve it. Change-Id: I7d37f9ba2aea139dabf2ca72d31fbe34bd9b2fa1	8 years ago
Sean Bright	f9c6f69205	utils: Add convenience function for setting fd flags There are many places in the code base where we ignore the return value of fcntl() when getting/setting file descriptior flags. This patch introduces a convenience function that allows setting or clearing file descriptor flags and will also log an error on failure for later analysis. Change-Id: I8b81901e1b1bd537ca632567cdb408931c6eded7	8 years ago
Richard Mudgett	04f7a58f67	security-events: Fix SuccessfulAuth using_password declaration. The SuccessfulAuth using_password field was declared as a pointer to a uint32_t when the field was later read as a uint32_t value. This resulted in unnecessary casts and a non-portable field value reinterpret in main/security_events.c:add_json_object(). i.e., It would work on a 32 bit architecture but not on a 64 bit big endian architecture. Change-Id: Ia08bc797613a62f07e5473425f9ccd8d77c80935	8 years ago
George Joseph	d5eb530bdf	Merge "CLI: Create ast_cli_completion_add function." into 15	8 years ago
George Joseph	c34223c958	Merge "CLI: Remove calls to ast_cli_generator." into 15	8 years ago
Joshua Colp	69fcd96e72	Merge "BuildSystem: pjsip_evsub_set_uas_timeout was not used." into 15	8 years ago
Corey Farrell	6748f690f5	CLI: Create ast_cli_completion_add function. Some completion generators are very inefficent due to the way CLI requests matches one at a time. ast_cli_completion_add can be called multiple times during one invokation of a CLI generator to add all results without having to reinitialize the search state for each match. Change-Id: I73d26d270bbbe1e3e6390799cfc1b639e39cceec	8 years ago
Corey Farrell	369e77d68b	CLI: Remove calls to ast_cli_generator. The ability to add to localized storage cannot be supported by ast_cli_generator. The only calls to ast_cli_generator should be by functions that need to proxy the CLI generator, for example 'cli check permissions' or 'core show help'. * ast_cli_generatornummatches now retrieves the vector of matches and reports the number of elements (not including 'best' match). * test_substitution retrieves and iterates the vector. Change-Id: I8cd6b93905363cf7a33a2d2b0e2a8f8446d9f248	8 years ago
Joshua Colp	5de5f10a5e	Merge "README: Send people to secure websites where available." into 15	8 years ago

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3849 Commits (49aaabbd7eb05bdf56873a70852b03a108da1875)