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Author	SHA1	Message	Date
Igor Goncharovsky	95b3c9839d	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	e7f1f7eefd	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	34ab9964f5	codec_resample: Upgrade speex_resample to fix up-sampling bug ... ASTERISK-28511 #close Change-Id: Idd07bf341e89ac999c7f5701d9b72b8a9cb11e82	6 years ago
Friendly Automation	7f64921a37	Merge "Fix misname 'res_external_mwi' to 'res_mwi_external' in comments." into 17	6 years ago
Friendly Automation	e33ab894bf	Merge "pjproject: Configurable setting for cnonce to include hyphens or not" into 17	6 years ago
Alexei Gradinari	a0b0c7540c	Fix misname 'res_external_mwi' to 'res_mwi_external' in comments. ... Change-Id: Ic784be8500e5cb75dcb34bae9f03cfd93b6b34fb	6 years ago
George Joseph	581ffdf6a9	chan_rtp: Accept hostname as well as ip address as destination ... The UnicastRTP channel driver provided by chan_rtp now accepts "<hostname>:<port>" as an alternative to "<ip_address>:<port>" in the destination. The first AAAA (preferred) or A record resolved will be used as the destination. The lookup is synchronous so beware of possible dialplan delays if you specify a hostname. Change-Id: Ie6f95b983a8792bf0dacc64c7953a41032dba677	6 years ago
George Joseph	b3d9f6ef6c	dns_core: Create new API ast_dns_resolve_ipv6_and_ipv4 ... The new function takes in a pointer to an ast_sockaddr structure, a hostname and an optional port and then dispatches parallel "AAAA" and "A" record queries. If an "AAAA" record is returned, it's parsed into the ast_sockaddr structure along with the port if it was supplied. If no "AAAA" record was returned, the first "A" record returned (if any) is parsed instead. This is a synchronous call. If you need asynchronous lookups, use ast_dns_query_set_resolve_async and roll your own. Change-Id: I194b0b0e73da94b35cc35263a868ffac3a8d0a95	6 years ago
Dan Cropp	a1d38e19a2	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	61edef60a1	Merge "res_pjsip: Channel variable SIPFROMDOMAIN" into 17	6 years ago
Friendly Automation	8f7052f8dc	Merge "res_ari.c: Prefer exact handler match over wildcard" into 17	6 years ago
Stas Kobzar	3a246c2a69	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	f6ab42df26	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	e123d1ccc5	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	052ab9d966	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	a5f05eed70	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	5c9ae25090	Merge "srtp: Fix possible race condition, and add NULL checks" into 17	6 years ago
George Joseph	71f89ce175	Merge "cdr / cel: Use event time at event creation instead of processing." into 17	6 years ago
George Joseph	5489ad5574	CI: Escape backslashes in printenv/sort/tr ... Change-Id: I52be64c8f6af2bbe15148a856d1f10cb113e1e94 (cherry picked from commit c6558e09af)	6 years ago
Kevin Harwell	5daa9bbaee	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	aa44e723c5	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
Friendly Automation	90a14a0a90	Merge "app_voicemail: Remove extra menuselect build options" into 17	6 years ago
Friendly Automation	b4ab3dab90	Merge "CI: Make node labels job-specific" into 17	6 years ago
Joshua Colp	108b1abbd9	cdr / cel: Use event time at event creation instead of processing. ... When updating times on CDR or CEL records using the time at which it is done can result in times being incorrect if the system is heavily loaded and stasis message processing is delayed. This change instead makes it so CDR and CEL use the time at which the stasis messages that drive the systems are created. This allows them to be backed up while still producing correct records. ASTERISK-28498 Change-Id: I6829227e67aefa318efe5e183a94d4a1b4e8500a	6 years ago
George Joseph	ec7cd3b0a5	Merge "res_musiconhold: Use a vector instead of custom array allocation" into 17	6 years ago
George Joseph	8652140794	CI: Make node labels job-specific ... Originally, the eligible nodes for a job were labelled only by "swdev-docker". So basically any node could run any job. We had found that allowing a node to run more than 1 gate at a time was problematic so we limited the nodes to processing 1 job at a time. With the creation of the Asterisk 17 branches however, we now have so many active branches that getting checks and gates through in a timely manner is problematic when a node can run only 1 job at a time. Now the nodes are also labelled by the job type they can run. For instance: "asterisk-check", "asterisk-gate", etc. With the "Throttle Concurrent Builds" plugin, we can now allow a node to run more than 1 job BUT throttle by job type. For instance: Allow 2 jobs but only 1 asterisk-gate at a time. Now a node can run 2 checks or 1 check and 1 gate or 1 gate but not 2 gates at a time. Change-Id: I2032bf6afbcec5c341d9b852214c0c812d3d6db5	6 years ago
Friendly Automation	261cb42ccb	Merge "various modules: json integer overflow" into 17	6 years ago
Friendly Automation	a2a4e9c717	Merge "main/udptl.c: correctly handle udptl sequence wrap around" into 17	6 years ago
Sean Bright	87f8ca2812	app_voicemail: Remove extra menuselect build options ... You now select voicemail backends like normal dialplan applications, so there is no longer a need for their own menuselect category. Reported by snuff-work in #asterisk-dev Change-Id: Idfa4c9c8349726074318a9e6b68d24c374521005	6 years ago
Kevin Harwell	a305f2fdcb	various modules: json integer overflow ... There were still a few places in the code that could overflow when "packing" a json object with a value outside the base type integer's range. For instance: unsigned int value = INT_MAX + 1 ast_json_pack("{s: i}", value); would result in a negative number being "packed". In those situations this patch alters those values to a ast_json_int_t, which widens the value up to a long or long long. ASTERISK-28480 Change-Id: Ied530780d83e6f1772adba0e28d8938ef30c49a1	6 years ago
Sean Bright	1976db6ffd	res_musiconhold: Use a vector instead of custom array allocation ... Change-Id: Ic476a56608b1820ca93dcf68d10cd76fc0b94141	6 years ago
Joshua Colp	20459d4cac	res_pjsip: Fix multiple of the same contact in "pjsip show contacts". ... The code for gathering contacts could result in the same contact being retrieved and added to the list multiple times. The container which stores the contacts to display will now only allow a contact to be added to it once instead of multiple times. ASTERISK-28228 Change-Id: I805185cfcec03340f57d2b9e6cc43c49401812df	6 years ago
George Joseph	7300bdd382	Merge "loader.c: Fix possible SEGV when a module fails to register" into 17	6 years ago
George Joseph	cccad75517	Merge "res_musiconhold: Use ast_pipe_nonblock() wrapper" into 17	6 years ago
Torrey Searle	44af3e9018	main/udptl.c: correctly handle udptl sequence wrap around ... incorrect handling of UDPTL squence number wrap arounds causes loss of packets every time the wrap around occurs ASTERISK-28483 #close Change-Id: I33caeb2bf13c574a1ebb81714b58907091d64234	6 years ago
Sean Bright	c748187bbf	manager: Send fewer packets ... The functions that build manager message headers do so in a way that results in a single messages being split across multiple packets. While this doesn't matter to the remote end, it makes network captures noisier and harder to follow, and also means additional system calls. With this patch, we build up more of the message content into the TLS buffer before flushing to the network. This change is completely internal to the manager code and does not affect any of the existing API's consumers. Change-Id: I50128b0769060ca5272dbbb5e60242d131eaddf9	6 years ago
Asterisk Development Team	f1e0c9d1b8	Update CHANGES and UPGRADE.txt for 17.0.0	6 years ago
George Joseph	4f2ad5a55d	doc: Add "master-only" flag back to the CHANGES and UPGRADE files ... In order to run the documentation scripts the flags needs to be added back to the staging files. Change-Id: Ia10a153c50c970cfa1e85815208dfaddb3f2ccd4	6 years ago
Sean Bright	a66ccb90ac	res_musiconhold: Use ast_pipe_nonblock() wrapper ... Change-Id: Ib0a4b41e5ececbe633079e2d8c2b66c031d2d1f2	6 years ago
George Joseph	e821a637a8	loader.c: Fix possible SEGV when a module fails to register ... When a module fails to register itself (usually a coding error in the module), dlerror() can return NULL. We weren't checking for that in load_dlopen() before trying to strdup the error message so a SEGV was thrown. dlerror() is now surrounded with an S_OR so we don't SEGV. Change-Id: Ie0fb9316f08a321434f3f85aecf3c7d2ede8b956	6 years ago
George Joseph	80d8dce6af	Prepare Asterisk 17 Branch ... Change-Id: Idb79a69646d2511e7bf1573b9b0322cc22ea54e8	6 years ago
George Joseph	32642b83ea	Merge "contrib/scripts: Make spandspflow2pcap.py Python 2.7+/3.3+ compatible"	6 years ago
George Joseph	dd80db10a9	Merge "CI: Don't enable non-core modules in Certified branches"	6 years ago
George Joseph	03813e51f0	CI: Don't enable non-core modules in Certified branches ... We don't support non-core modules for Certified releases but we were enabling them for CI builds which was causing lots of test failures. Now we don't. Change-Id: I0b3254c08a2479f3d39151690350cce5ce5ad766	6 years ago
Sean Bright	2424ecaf66	res_config_sqlite3: Only join threads that we started ... ASTERISK-28477 #close Reported by: Dennis ASTERISK-28478 #close Reported by: Dennis Change-Id: I77347ad46a86dc5b35ed68270cee56acefb4f475	6 years ago
Friendly Automation	7646f4196b	Merge "openr2(6/6): Set hangup cause"	6 years ago
George Joseph	96a2ce1a0d	Merge "openr2(5/6): added cli command -- mfcr2 destroy link <index>"	6 years ago
George Joseph	64b6d0fc28	Merge "openr2(4/6): added new cli command -- mfcr2 show links"	6 years ago
Friendly Automation	d24d94beba	Merge "openr2(3/6): Convert r2links to standard Asterisk AST_LIST*"	6 years ago
George Joseph	8c3ed46829	Merge "openr2(2/6): Stop polling channels when DAHDI returns -ENODEV (e.g: plug-out)"	6 years ago