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Author	SHA1	Message	Date
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
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