32288 Commits (0cfe9186b07bdd35bb82c74c93f9fbccac5f96a3)

 

All Branches   

Search

Author	SHA1	Message	Date
Friendly Automation	5b8e0fef34	Merge "core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option." into 17	6 years ago
George Joseph	ad4a358595	Merge "pjproject_bundled: Revert pjproject 2.9 commits causing leaks" into 17	6 years ago
Kevin Harwell	175a7ccac7	res_pjsip_pubsub: change warning to debug ... The following message: "Subscription request from endpoint <blah> rejected. Expiration of 0 is invalid" Would sometimes spam the log with warnings if Asterisk restarted and a bunch of clients sent unsubscribes. This patch changes it from a warning to a debug message. Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467	6 years ago
Corey Farrell	5bedd4a9b4	core: Fix ABI mismatch of ao2_global_obj. ... astobj2.c declares DEBUG_THREADS_LOOSE_ABI to avoid overhead of debug threads tracking information in the internal structures of astobj2. Unfortunately this means that ao2_global_obj contains the statically allocated debug threads tracking fields which are used by initialization and cleanup but main/astobj2.c believed those fields and associated space did not exist. Change-Id: Icef41ad97d88a8c1d1515e034ec8133cab3b1527	6 years ago
Ben Ford	f181dfc065	taskprocessor.c: Add CLI commands to reset taskprocessor stats. ... Added two new CLI commands to reset stats for taskprocessors. You can reset stats for a single, specific taskprocessor ('core reset taskprocessor <taskprocessor>'), or you can reset all taskprocessors ('core reset taskprocessors'). These commands will reset the counter for the number of tasks processed as well as the max queue size. Change-Id: Iaf17fc4ae29396ab0c6ac92408fc7bdc2f12362d	6 years ago
George Joseph	d95a76edd3	Merge "astmm.c: Display backtrace with memory show allocations" into 17	6 years ago
George Joseph	7c0435f854	pjproject_bundled: Revert pjproject 2.9 commits causing leaks ... We've found a connection re-use regression in pjproject 2.9 introduced by commit "Close #1019: Support for multiple listeners." https://trac.pjsip.org/repos/changeset/6002 https://trac.pjsip.org/repos/ticket/1019 Normally, multiple SSL requests should reuse the same connection if one already exists to the remote server. When a transport error occurs, the next request should establish a new connection and any following requests should use that same one. With this patch, when a transport error occurs, every new request creates a new connection so you can wind up with thousands of open tcp sockets, possibly exhausting file handles, and increasing memory usage. Reverting pjproject commit 6002 (and related 6021) restores the expected behavior. We also found a memory leak in SSL processing that was introduced by commit "Fixed #2204: Add OpenSSL remote certificate chain info" https://trac.pjsip.org/repos/changeset/6014 https://trac.pjsip.org/repos/ticket/2204 Apparently the remote certificate chain is continually recreated causing the leak. Reverting pjproject commit 6014 (and related 6022) restores the expected behavior. Both of these issues have been acknowledged by Teluu. ASTERISK-28521 Change-Id: I8ae7233c3ac4ec29a3b991f738e655dabcaba9f1	6 years ago
Friendly Automation	3b485b3e5a	Merge "res_sorcery_memory_cache: stale item update leak" into 17	6 years ago
Kevin Harwell	1d322dc071	Merge "stasis: refcounter.py can incorrectly report skewed objects." into 17	6 years ago
Friendly Automation	09d211b611	Merge "app_voicemail: Fix module unload leak." into 17	6 years ago
Corey Farrell	ec712cfab2	core: Add AO2_ALLOC_OPT_NO_REF_DEBUG option. ... Previous to this patch passing a NULL tag to ao2_alloc or ao2_ref based functions would result in the reference not being logged under REF_DEBUG. This could sometimes cause inaccurate logging if NULL was accidentally passed to a reference action. Now reference logging is only disabled by option passed to the allocation method. Change-Id: I3c17d867d901d53f9fcd512bef4d52e342637b54	6 years ago
Kevin Harwell	f821e81071	res_sorcery_memory_cache: stale item update leak ... When a stale item was being updated the object was being retrieved, but its reference was not being decremented after the update. This patch makes it so the object is now appropriately de-referenced. ASTERISK-28523 Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7	6 years ago
George Joseph	b1c5753b78	astmm.c: Display backtrace with memory show allocations ... You can currently capture backtraces of memory allocations but they only get displayed when you stop asterisk and the atexit hooks are enabled. Now, if memory backtrace is on and you issue a "memory show allocations" CLI command for a specific file, then a backtrace will show for each allocation that occurred after you turned "memory backtrace on". The backtrace display is shown only when a specific file's allocations are displayed to prevent a massive CLI dump of every file's allocations. Change-Id: Ic657afc1fc6ec7205e16eb36a97a611d235a2b4f	6 years ago
Corey Farrell	af0ccbc45e	stasis: refcounter.py can incorrectly report skewed objects. ... It is possible for topic->name to be NULL, this causes the allocation reference to not be logged. Use the name variable instead which has been verified to be a non-empty. Change-Id: I3d0031d03c8356e4808f00cdf2d5428712575883	6 years ago
Corey Farrell	fd7fc3d7a5	stasis: Fix leaks ... * Release reference returned by cache_remove * state_alloc unconditionally bumped state_topic even when it was locally allocated. Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7	6 years ago
Corey Farrell	78f329dad9	app_voicemail: Fix module unload leak. ... Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a	6 years ago
Joshua Colp	6e75e2565d	Merge "func_jitterbuffer: Add audio/video sync support." into 17	6 years ago
Joshua Colp	d2087607b6	Merge "core: Add H.265/HEVC passthrough support" into 17	6 years ago
Joshua Colp	926053d7bd	func_jitterbuffer: Add audio/video sync support. ... This change adds support to the JITTERBUFFER dialplan function for audio and video synchronization. When enabled the RTCP SR report is used to produce an NTP timestamp for both the audio and video streams. Using this information the video frames are queued until their NTP timestamp is equal to or behind the NTP timestamp of the audio. The audio jitterbuffer acts as the leader deciding when to shrink/grow the jitterbuffer when adaptive is in use. For both adaptive and fixed the video buffer follows the size of the audio jitterbuffer. ASTERISK-28533 Change-Id: I3fd75160426465e6d46bb2e198c07b9d314a4492	6 years ago
Joshua Colp	69e8a44039	Merge "chan_pjsip: Relock correct channel during "fax" redirect." into 17	6 years ago
George Joseph	b7e9c8437e	Merge "chan_dahdi: Fix build with clang/llvm" into 17	6 years ago
Florian Floimair	d7a3e4f5cf	core: Add H.265/HEVC passthrough support ... This change adds H.265/HEVC as a known codec and creates a cached "h265" media format for use. Note that RFC 7798 section 7.2 also describes additional SDP parameters. Handling of these is not yet supported. ASTERISK-28512 Change-Id: I26d262cc4110b4f7e99348a3ddc53bad0d2cd1f2	6 years ago
Joshua Colp	4d1baa3ae8	chan_pjsip: Relock correct channel during "fax" redirect. ... When fax detection occurs on an outbound PJSIP channel the redirect operation will result in a masquerade occurring and the underlying channel on the session changing. The code incorrectly relocked the new channel instead of the old channel when returning. This resulted in the new channel being locked indefinitely. The code now always acts on the expected channel. ASTERISK-28538 Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3	6 years ago
Guido Falsi	5ff2f7a016	chan_dahdi: Fix build with clang/llvm ... On FreeBSD using the clang/llvm compiler build fails to build due to the switch statement argument being a non integer type expression. Switch to an if/else if/else construct to sidestep the issue. ASTERISK-28536 #close Change-Id: Idf4a82cc1e94580a2d017fe9e351c226f23e20c8	6 years ago
Ben Ford	1d960195c2	res_rtp_asterisk.c: Send RTCP as compound packets. ... According to RFC3550, ALL RTCP packets must be sent in a compond packet of at least two individual packets, including SR/RR and SDES. REMB, FIR, and NACK were not following this format, and as a result, would fail the packet check in ast_rtcp_interpret. This was found from writing unit tests for RTCP. The browser would accept the way we were constructing these RTCP packets, but when sending directly from one Asterisk instance to another, the above mentioned problem would occur. Change-Id: Ieb140e9c22568a251a564cd953dd22cd33244605	6 years ago
Sean Bright	2fa296e7d4	channels: Allow updating variable value ... When modifying an already defined variable in some channel drivers they add a new variable with the same name to the list, but that value is never used, only the first one found. Introduce ast_variable_list_replace() and use it where appropriate. ASTERISK-23756 #close Patches: setvar-multiplie.patch submitted by Michael Goryainov Change-Id: Ie1897a96c82b8945e752733612ee963686f32839	6 years ago
Friendly Automation	cb9fdc21d1	Merge "res_rtp: Add unit tests for RTCP stats." into 17	6 years ago
George Joseph	492da93771	Merge "ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf." into 17	6 years ago
Friendly Automation	db1da38649	Merge "res_musiconhold: Added unregister realtime moh class" into 17	6 years ago
Joshua Colp	377f56bdc0	Merge "chan_sip: Update links referenced in deprecation notice" into 17	6 years ago
Joshua Colp	9f5a027aa3	Merge "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up" into 17	6 years ago
Joshua Colp	e82a765169	Merge "codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary" into 17	6 years ago
sungtae kim	9e26136ee6	res_musiconhold: Added unregister realtime moh class ... This fix allows a realtime moh class to be unregistered from the command line. This is useful when the contents of a directory referenced by a realtime moh class have changed. The realtime moh class is then reloaded on the next request and uses the new directory contents. ASTERISK-17808 Change-Id: Ibc4c6834592257c4bb90601ee299682d15befbce	6 years ago
Ben Ford	5dad4f4ea1	res_rtp: Add unit tests for RTCP stats. ... Added unit tests for RTCP video stats. These tests include NACK, REMB, FIR/FUR/PLI, SR/RR/SDES, and packet loss statistics. The REMB and FIR tests are currently disabled due to a bug. We expect to receive a compound packet, but the code sends this out as a single packet, which the browser accepts, but makes Asterisk upset. While writing these tests, I noticed an issue with NACK as well. Where it is handling a received NACK request, it was reading in only the first 8 bits of following packets that were also lost. This has been changed to the correct value of 16 bits. Also made a minor fix to the data buffer unit test. Change-Id: I56107c7411003a247589bbb6086d25c54719901b	6 years ago
Frederic LE FOLL	50997de887	ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf. ... ChanIsAvail() creates a temporary channel with ast_request() to test resource availability. It should not generate a CDR when it hangs up this temporary channel. This patch disables CDR generation for the temporary channel with ast_cdr_set_property(). ASTERISK-28527 Change-Id: I7b0555c6909c7d322e452dde97c9ea5b111552d1	6 years ago
Frederic LE FOLL	9e67c92502	chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up ... When the remote ISDN party ends an ISDN call on a PRI link (DISCONNECT), CHANNEL(hangupsource) information is not available. chan_dahdi already contains an ast_set_hangupsource() in __dahdi_exception() function but it seems that ISDN message processing does not use this part of code. Two other channel modules associate ast_queue_hangup() and ast_set_hangupsource() functions calls: - chan_pjsip in chan_pjsip_session_end() function, - chan_sip in sip_queue_hangup_cause() function. chan_iax2 separates them, in iax2_queue_hangup()/iax2_destroy() and set_hangup_source_and_cause(). Thus, I propose to add ast_set_hangupsource() beside ast_queue_hangup() in sig_pri_queue_hangup(), like chan_pjsip and chan_sip already do. ASTERISK-28525 Change-Id: I0f588a4bcf15ccd0648fd69830d1b801c3f21b7c	6 years ago
George Joseph	44068cc6bf	ARI: External Media ... The Channel resource has a new sub-resource "externalMedia". This allows an application to create a channel for the sole purpose of exchanging media with an external server. Once created, this channel could be placed into a bridge with existing channels to allow the external server to inject audio into the bridge or receive audio from the bridge. See https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI for more information. Change-Id: I9618899198880b4c650354581b50c0401b58bc46	6 years ago
Friendly Automation	03625b4bc6	Merge "test_utils.c: Skip test adsi_loaded_test if module not loaded." into 17	6 years ago
Friendly Automation	70ebcf26c0	Merge "chan_unistim: Fix clang warning: variable sized type not at end of a struct" into 17	6 years ago
George Joseph	c060ef7dee	chan_sip: Update links referenced in deprecation notice ... The links in the deprecation notice were the shortened variety but it makes better sense to show the unshortened links as they're more descriptive. I.E. wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip rather than wiki.asterisk.org/wiki/x/tAHOAQ Change-Id: If2da5d5243e2d4a6f193b15691d23e7e5a7c57a9	6 years ago
Sean Bright	75d6418d8e	codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary ... ASTERISK-28511 Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d	6 years ago
George Joseph	e35918235e	Merge "AST-2019-005 - translate: Don't assume all frames will have a src." into 17	6 years ago
Joshua Colp	61c01df560	AST-2019-005 - translate: Don't assume all frames will have a src. ... This change removes the assumption that a frame will always have a src set on it. This assumption is incorrect. Given a scenario where an RTP packet is received with no payload the resulting audio frame will have no samples. If this frame goes through a signed linear translation path an interpolated frame can be created (if generic packet loss concealment is enabled) that has minimal data on it, including no src. If this frame is given to a translation path a crash will occur due to the lack of src. ASTERISK-28499 Change-Id: I024d10dd98207eb8a6b35b59880bcdf1090538f8	6 years ago
Kevin Harwell	9d4f1e8ebe	AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media ... After receiving a 200 OK with a declined stream in response to a T.38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before attempting to use it. ASTERISK-28495 patches: ast-2019-004.patch submitted by Alexei Gradinari (license 5691) Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572	6 years ago
Chris-Savinovich	7d99710fa6	test_utils.c: Skip test adsi_loaded_test if module not loaded. ... Module res_adsi.so is deprecated, therefore it does not load by default. Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c. This fix checks if the corresponding module is loaded at the start of the test, and if not, it passes the test and exits with a message. This fix is applied to all versions where the module is marked deprecated. Change-Id: I52be64c8f6af222e15148a856d1f10cb113e1e94	6 years ago
Igor Goncharovsky	056ddf76ce	chan_unistim: Fix clang warning: variable sized type not at end of a struct ... On reading information about initial client packet unistim use dirty implementation of destination ip address retrieval. This fix uses CMSG_*(..) to get ip address and make clang compile without warning. ASTERISK-25592 #close Reported-by: Alexander Traud Change-Id: Ic1fd34c2c2bcc951da65bf62e3f7a8adff8351b1	6 years ago
George Joseph	16ae4cd2db	Merge "res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions" into 17	6 years ago
George Joseph	2cd5b84af8	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk" into 17	6 years ago
George Joseph	0aae0499da	Merge "codec_resample: Upgrade speex_resample to fix up-sampling bug" into 17	6 years ago
Kevin Harwell	a36fb473fe	res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions ... res_pjsip_mwi allows both solicited and unsolicited MWI subscription types. While both can be set in the configuration for a given endpoint/aor, only one is allowed. Precedence is given to unsolicited. Meaning if an endpoint/aor is configured to allow both types then the solicited subscription is rejected when it comes in. However, there is a configuration option to override that behavior: mwi_subscribe_replaces_unsolicited When set to "yes" then when a solicited subscription comes in instead of rejecting it Asterisk is suppose to replace the unsolicited one if it exists. Prior to this patch there was a bug in Asterisk that allowed the solicted one to be added, but did not remove the unsolicited. As a matter of fact a new unsolicited subscription got added everytime a SIP register was received. Over time this eventually could "flood" a phone with SIP notifies. This patch fixes that behavior to now make it work as expected. If configured to do so a solicited subscription now properly replaces the unsolicited one. As well when an unsubscribe is received the unsolicited subscription is restored. Logic was also put in to handle reloads, and any configuration changes that might result from that. For instance, if a solicited subscription had previously replaced an unsolicited one, but after reload it was configured to not allow that then the solicited one needs to be shutdown, and the unsolicited one added. ASTERISK-28488 Change-Id: Iec2ec12d9431097e97ed5f37119963aee41af7b1	6 years ago