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Author	SHA1	Message	Date
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
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
George Joseph	0e747da168	Merge "chan_unistim: Fix RTP port byte order for big-endian arch" into 13	6 years ago
Igor Goncharovsky	7f1a574ce7	chan_unistim: Fix code, causing all incoming DTMF sent back to asterisk ... Current implementation of ast_channel_tech send_digit_begin hook uses same function for tone playback as key press handler. This cause every incoming dtmf send back to asterisk. In case of two unistim phones connected to each other, it'll cause indefinite DTMF loop. Fix add separate function for dtmf tone phone play. Change-Id: I5795db468df552f0c89c7576b6b3858b26c4eab4	6 years ago
Igor Goncharovsky	b003c9b3ae	chan_unistim: Fix RTP port byte order for big-endian arch ... This patch fixes one-way oudio that users expirienced on big-endian architechtires. RTP port number bytes was stored in improper order and phone sent RTP to wrong RTP port. Reported-by: Andrey Ionov Change-Id: I9a9ca7f26e31a67bbbceff12923baa10dfb8a3be	6 years ago
Sean Bright	5154e7f359	codec_resample: Upgrade speex_resample to fix up-sampling bug ... ASTERISK-28511 #close Change-Id: Idd07bf341e89ac999c7f5701d9b72b8a9cb11e82	6 years ago
Joshua Colp	4d4a430128	Merge "Fix misname 'res_external_mwi' to 'res_mwi_external' in comments." into 13	6 years ago
Joshua Colp	f0bc74a291	Merge "pjproject: Configurable setting for cnonce to include hyphens or not" into 13	6 years ago
Alexei Gradinari	e6a4cca4aa	Fix misname 'res_external_mwi' to 'res_mwi_external' in comments. ... Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb	6 years ago
George Joseph	b02f53463c	Merge "res_pjsip: Channel variable SIPFROMDOMAIN" into 13	6 years ago
Dan Cropp	1a86b2499a	pjproject: Configurable setting for cnonce to include hyphens or not ... NEC SIP Station interface with authenticated registration only supports cnonce up to 32 characters. In Linux, PJSIP would generate 36 character cnonce which included hyphens. Teluu developed this patch adding a compile time setting to default to not include the hyphens. They felt it best to still generate the UUID and strip the hyphens. They have indicated it will be part of PJSIP 2.10. ASTERISK-28509 Reported-by: Dan Cropp Change-Id: Ibdfcf845d4f8c0a14df09fd983b11f2d72c5f470	6 years ago
Friendly Automation	8bc51928c1	Merge "res_ari.c: Prefer exact handler match over wildcard" into 13	6 years ago
Stas Kobzar	06e5c11589	res_pjsip: Channel variable SIPFROMDOMAIN ... In chan_sip, there was variable SIPFROMDOMAIN that allows to set From header URI domain per channel. This patch introduces res_pjsip variable SIPFROMDOMAIN for backward compatibility with chan_sip. ASTERISK-28489 Change-Id: I715133e43172ce2a1e82093538dc39f9e99e5f2e	6 years ago
George Joseph	22372d5d2b	res_ari.c: Prefer exact handler match over wildcard ... Given the following request path and 2 handler paths... Request: /channels/externalMedia Handler: /channels/{channelId} "wildcard" Handler: /channels/externalmedia "non-wildcard" ...if /channels/externalMedia was registered as a handler after /channels/{channelId} as shown above, the request would automatically match the wildcard handler and attempt to parse "externalMedia" into the channelId variable which isn't what was intended. It'd work if the non-wildard entry was defined in rest-api/api-docs/channels.json before the wildcard entry but that makes the json files order-dependent which isn't a good thing. To combat this issue, the search loop saves any wildcard match but continues looking for exact matches at the same level. If it finds one, it's used. If it hasn't found an exact match at the end of the current level, the wildcard is used. Regardless, after searching the current level, the wildcard is cleared so it won't accidentally match for a different object or a higher level. BTW, it's currently not possible for more than 1 wildcard entry to be defined for a level. For instance, there couldn't be: Handler: /channels/{channelId} Handler: /channels/{channelName} We wouldn't know which one to match. Change-Id: I574aa3cbe4249c92c30f74b9b40e750e9002f925	6 years ago
Sean Bright	ddc64ca059	audiohook.c: Substitute silence for unavailable audio frames ... There are 4 scenarios to consider when capturing audio from a channel with an audiohook: 1. There is no rx and no tx audio, so return nothing. 2. There is rx but no tx audio, so return rx. 3. There is tx but no rx audio, so return tx. 4. There is rx and tx audio, so mix them and return. The file passed as the primary argument to MixMonitor will be written to in scenarios 2, 3, and 4. However, if you pass the r() and t() options to MixMonitor, a frame will only be written to the r() file if there was rx audio and a frame will only be written to the t() file if there was tx audio. If you subsequently take the r() and t() files and try to mix them, the sides of the conversation will 'drift' and be non-representative of the user experience. This patch adds a new 'S' option to MixMonitor that injects a frame of silence on either the r() side or the t() side of the channel so that when later mixed, there is no such drift. Change-Id: Ibf5ed73a811087727bd561a89a59f4447b4ee20e	6 years ago
Alexei Gradinari	16bc4ed93c	app_voicemail/IMAP: check mailstream not NULL in leave_voicemail ... The function leave_voicemail checks if expungeonhangup is set, but does not check if IMAP stream is closed, so it could call imap function with NULL stream. This leads to segfault. ASTERISK-28505 #close Change-Id: Ib66c57c1f1ba97774e447b36349198e2626a8d7c	6 years ago
Sean Bright	8a96746994	menuselect: Fix curses build on Gentoo Linux ... Because keypad() is exported by libtinfo, it needs to be explicitly added to the linker options. ASTERISK-28487 #close Change-Id: I6c2ad5b95f422c263d078b5c0e84c111807dffc6	6 years ago
George Joseph	7251b76c62	Merge "srtp: Fix possible race condition, and add NULL checks" into 13	6 years ago
Friendly Automation	5598dc045b	Merge "cdr / cel: Use event time at event creation instead of processing." into 13	6 years ago
George Joseph	c6558e09af	CI: Escape backslashes in printenv/sort/tr ... Change-Id: I52be64c8f6af2bbe15148a856d1f10cb113e1e94	6 years ago
Kevin Harwell	feaadbd250	srtp: Fix possible race condition, and add NULL checks ... Somehow it's possible for the srtp session object to be NULL even though the Asterisk srtp object itself is valid. When this happened it would cause a crash down in the srtp code when attempting to protect or unprotect data. After looking at the code there is at least one spot that makes this situation possible. If Asterisk fails to unprotect the data, and after several retries it still can't then the srtp->session gets freed, and set to NULL while still leaving the Asterisk srtp object around. However, according to the original issue reporter this does not appear to be their situation since they found no errors logged stating the above happened (which Asterisk does for that situation). An issue was found however, where a possible race condition could occur between the pjsip incoming negotiation, and the receiving of RTP packets. Both places could attempt to create/setup srtp for the same rtp instance at the same time. This potentially could be the cause of the problem as well. Given the above this patch adds locking around srtp setup for a given rtp, or rtcp instance. NULL checks for the session have also been added within the protect and unprotect functions as a precaution. These checks should at least stop Asterisk from crashing if it gets in this situation again. This patch also fixes one other issue noticed during investigation. When doing a replace the old object was freed before creating the replacement. If the new replacement object failed to create then the rtp/rtcp instance would now point to freed srtp data which could potentially cause a crash as well when the next attempt to reference it was made. This is now fixed so the old srtp object is kept upon replacement failure. Lastly, more logging has been added to help diagnose future issues. ASTERISK-28472 Change-Id: I240e11cbb1e9ea8083d59d50db069891228fe5cc	6 years ago
George Joseph	17908cb76e	CI: Add "throttle" label and "skip_gate" capability ... To make throttling by label fully active, the "throttle" option has to be specified with a specific label. You can now specify "skip_gate" in the Gerrit comments when you do a +2 code review to tell Jenkins not to actually run the gate. You'd do this if you plan to manually merge the change. Also updated the "printenv" debug output to better sort multi-line comments. Change-Id: I4c0b1085acec4805f2ca207eebac50aad81f27e2	6 years ago