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Author	SHA1	Message	Date
George Joseph	1b29d0a7ed	Merge "res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6" into 13	6 years ago
Torrey Searle	6fd195ca91	channel/chan_pjsip: add dialplan function for music on hold ... Add a new dialplan function PJSIP_MOH_PASSTHROUGH that allows the on-hold behavior to be controlled on a per-call basis ASTERISK-28542 #close Change-Id: Iebe905b2ad6dbaa87ab330267147180b05a3c3a8	6 years ago
Alexei Gradinari	c1d82f73a3	res_pjsip_pubsub: add endpoint to some warning ... There are some warning messages which are not informative without endpoint: "No registered subscribe handler for event presence.winfo" "No registered publish handler for event presence" This patch adds an endpoint name to these messages. Change-Id: Ia2811ec226d8a12659b4f9d4d224b48289650827	6 years ago
Sean Bright	438085f10e	res_pjsip_transport_websocket: Don't put brackets around local_name if IPv6 ... ASTERISK-28544 #close Change-Id: I8e62c444d107674c298f472e3545661de8a80dce	6 years ago
George Joseph	c6f81a7ba4	Merge "taskprocessor.c: Added "like" support to 'core show taskprocessors'" into 13	6 years ago
Friendly Automation	88dea7a111	Merge "pbx: Prevent Realtime switch crash on invalid priority" into 13	6 years ago
Friendly Automation	ac64659b33	Merge "res_pjsip_registrar: Validate Contact URI before adding to responses" into 13	6 years ago
Joshua Colp	b60bf27881	Merge "chan_pjsip: Add support for passing hold and unhold requests through." into 13	6 years ago
Ben Ford	67bd02d08d	taskprocessor.c: Added "like" support to 'core show taskprocessors' ... Added "like" support for 'core show taskprocessors'. Now you can specify a specific set of taskprocessors (or just one) by adding the keyword "like" to the above command, followed by your search criteria. Change-Id: I021e740201e9ba487204b5451e46feb0e3222464	6 years ago
Asterisk Development Team	0dc1ba17b8	Update CHANGES and UPGRADE.txt for 13.29.0-rc2	6 years ago
Sean Bright	7280e2ec51	pbx: Prevent Realtime switch crash on invalid priority ... pbx_extension_helper takes two 'context' arguments. One (con) is a pointer directly to a 'struct ast_context' and the other (context) is the name of the context. In all cases, one of these arguments is NULL and the other is non-NULL. Functions that are ultimately called by pbx_extension_helper expect that 'context' will be non-NULL, so we set it unconditionally on entry into this function. ASTERISK-28534 #close Change-Id: Ifbbc5e71440afd80efd441f7a9d72e8b10b6f47d	6 years ago
Friendly Automation	fde6ed7d91	Merge "core: Fix ABI mismatch of ao2_global_obj." into 13	6 years ago
George Joseph	85f47022e0	Merge "res_pjsip_pubsub: change warning to debug" into 13	6 years ago
George Joseph	c5a9c039ba	Merge "taskprocessor.c: Add CLI commands to reset taskprocessor stats." into 13	6 years ago
Sean Bright	22fb0f286d	res_pjsip_registrar: Validate Contact URI before adding to responses ... If a permanent contact URI associated with an AOR is invalid, we add a Contact header to REGISTER responses with a NULL URI, causing a crash. ASTERISK-28463 #close Change-Id: Id2b643e58b975bc560aab1c111e6669d54db9102	6 years ago
Joshua Colp	d775bc5add	chan_pjsip: Add support for passing hold and unhold requests through. ... This change adds an option, moh_passthrough, that when enabled will pass hold and unhold requests through using a SIP re-invite. When placing on hold a re-invite with sendonly will be sent and when taking off hold a re-invite with sendrecv will be sent. This allows remote servers to handle the musiconhold instead of the local Asterisk instance being responsible. Review: https://reviewboard.asterisk.org/r/4103/ Change-Id: Ib6294e906e577e1a4245cb1f058d3976ff484c52	6 years ago
George Joseph	6991a944c0	Merge "pjproject_bundled: Revert pjproject 2.9 commits causing leaks" into 13	6 years ago
Kevin Harwell	9384639953	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	c881d36319	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	3069153bdf	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	e8a9cc21ab	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	6320b86d89	Merge "res_sorcery_memory_cache: stale item update leak" into 13	6 years ago
Friendly Automation	d6c52479a4	Merge "astmm.c: Display backtrace with memory show allocations" into 13	6 years ago
Kevin Harwell	f5dda8d796	Merge "res_pjsip_mwi: Remove inappropriate topic unreference." into 13	6 years ago
Kevin Harwell	6517690168	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	afd6b98757	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	48fd773743	res_pjsip_mwi: Remove inappropriate topic unreference. ... ast_mwi_topic() returns a borrowed reference which should not be unreferenced, doing so leads to a FRACK. This was hidden by the fact that stasis_cache.c leaked the result of cache_remove in caching_topic_exec. Change-Id: I51101bf7d07b8dc8ce8fc46b6cb31fbbd213fbc7	6 years ago
Corey Farrell	25918f2837	app_voicemail: Fix module unload leak. ... Change-Id: Ib9a06565b9a178822d3bbb67eccf51432e12d84a	6 years ago
Joshua Colp	16790d6029	Merge "chan_pjsip: Relock correct channel during "fax" redirect." into 13	6 years ago
Joshua Colp	a97bd89f84	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	ae70c2a428	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
Friendly Automation	a1b371603f	Merge "channels: Allow updating variable value" into 13	6 years ago
Sean Bright	622e487cd4	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
Asterisk Development Team	18fbd8d8c3	Update CHANGES and UPGRADE.txt for 13.29.0	6 years ago
Asterisk Development Team	2a7cec9051	Update CHANGES and UPGRADE.txt for 13.29.0	6 years ago
Friendly Automation	5bb71c4c1d	Merge "ChanIsAvail() generates a CDR when unanswered=yes in cdr.conf." into 13	6 years ago
George Joseph	a0f13c1e22	Merge "res_musiconhold: Added unregister realtime moh class" into 13	6 years ago
Joshua Colp	0bf277e8f6	Merge "chan_dahdi: set CHANNEL(hangupsource) when a PRI channel hangs up" into 13	6 years ago
Joshua Colp	228682501a	Merge "codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary" into 13	6 years ago
sungtae kim	2705652152	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
Frederic LE FOLL	9462722b17	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	2b563368df	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	e6f74e94f6	Merge "chan_unistim: Fix clang warning: variable sized type not at end of a struct" into 13	6 years ago
Sean Bright	060495ed86	codec_resample: Ensure OUTSIDE_SPEEX is defined when necessary ... ASTERISK-28511 Change-Id: If0d58598ce14aad3c786a1c0127b5f7b200b737d	6 years ago
Joshua Colp	c711546559	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
Igor Goncharovsky	8e07fcc30d	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	92876e8f0b	Merge "res_pjsip_mwi: add better handling of solicited vs unsolicited subscriptions" into 13	6 years ago
Friendly Automation	8b3cee9068	Merge "chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk" into 13	6 years ago
George Joseph	5632bcc126	Merge "codec_resample: Upgrade speex_resample to fix up-sampling bug" into 13	6 years ago
Kevin Harwell	af12d36f7e	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