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Author	SHA1	Message	Date
Alexander Traud	3e2178c05e	res_format_attr_opus: Update to latest RFC 7587. ... Beside that, the format-attribute module sends only non-default values in the line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, previously the parameter stereo was not parsed when being the first parameter. ASTERISK-25583 #close Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73	10 years ago
Mark Michelson	6fcd361540	Merge "res_pjsip_outbound_registration.c: Fix 423 response handling." into 13	10 years ago
Joshua Colp	bdc7845a43	Merge "res_format_attr_h264: Do not reset string buffer." into 13	10 years ago
Matt Jordan	8f71263e72	res/res_pjsip_outbound_registration: Apply configuration on object type load ... When Asterisk is configured to use a dynamic sorcery backend (such as res_sorcery_astdb) with 'registration' objects, it will fail to create the internal state objects associated with the registration objects on module load. This is due to nothing actually querying for the specific objects and calling their sorcery apply handler during module load. This patch fixes that by calling get_registrations in the sorcery observer's object_type_loaded handler. Doing this causes the sorcery backends to be asked for the current state of all registration objects, which causes the apply handler to be called and the internal run-time state to be created. ASTERISK-25575 #close Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23	10 years ago
Alexander Traud	1aa552b2a2	res_format_attr_h264: Do not reset string buffer. ... When no parameter is present, Asterisk does not generate the line fmtp, as expected. However, because a buffer was reset, even rtpmap and fmtp of previous media codecs got removed. Now, Asterisk does not reset other codecs in case of no parameter for H.264. ASTERISK-25573 #close Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286	10 years ago
Richard Mudgett	e44ab3816c	res_pjsip_outbound_registration.c: Fix 423 response handling. ... Receiving a 423 Interval Too Brief response after authentication for an outbound registration attempt results in assuming that the registrar has rejected the registration permanently. If there are no configured retries for fatal responses then the outbound registration is stopped for that endpoint. For registrations, PJSIP/PJPROJECT intercepts the handling of 423 responses and does not include any authentication in the updated registration request. When the updated request is challenged then the Asterisk code assumes that we were challenged again because the peer rejected the authentication we sent earlier. * Made registration challenges keep track of the CSeq number to determine if the received challenge response was for the request we thought we sent. If the response's CSeq number differs from the CSeq number we last sent with authentication then authenticate again because it is a challenge to a different request. Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09	10 years ago
Matt Jordan	ccf80f95a2	Merge "res_pjsip_rfc3326.c: Fix crash when channel goes away." into 13	10 years ago
Matt Jordan	e3cb27d341	Merge "format: Register format-attribute module with cached formats." into 13	10 years ago
Matt Jordan	6c10d30d0e	Merge "res/res_pjsip: Fix off nominal crash with requests that fail and have a timer" into 13	10 years ago
Joshua Colp	0843e6043e	Merge "Confbridge: Add a user timeout option" into 13	10 years ago
Matt Jordan	f62b642fe3	res/res_pjsip: Fix off nominal crash with requests that fail and have a timer ... When a request is sent using pjsip_endpt_send_request and fails, a condition exists where the request wrapper, which is an AO2 object, may be de-ref'd more times than it should. This occurs when the request's callback is called, and, in the callback, the timer on the PJSIP heap is cancelled. When that occurs, the request wrapper's lifetime is decremented. When pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of the request wrapper again, even though we've already cancelled the reference associated with the timer. This patch checks the return result of pj_timer_heap_cancel_if_active before removing the reference associated with the timer. We now only decrement it in this case if a timer is cancelled as a result of the function call. Change-Id: I21332343a1a019c1117076f9bf2df27be2850102	10 years ago
Mark Michelson	fdd2afcd16	Confbridge: Add a user timeout option ... This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98	10 years ago
Alec Davis	7debb986a5	app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! ... commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a	10 years ago
Joshua Colp	afd9a89e5a	hashtab: Add NULL check when destroying iterator. ... The hashtab API is pretty NULL tolerant which has resulted in remaining callers not doing much checks themselves. Unfortunately the function to destroy an iterator does not do a NULL check and will result in a crash if passed NULL. This change fixes that. ASTERISK-25552 #close Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619	10 years ago
Richard Mudgett	c0f2f8de45	res_pjsip_rfc3326.c: Fix crash when channel goes away. ... If an authenticated incoming caller does not respond to our 200 OK INVITE response with an ACK then PJSIP will hangup the call. Unfortunately, there is a chance that the session's channel will go away between one use of the channel pointer and another when building the BYE request because the BYE is being built by the monitor thread and not the call's serializer thread. * Added a check to ensure that the thread trying to add the Reason header is the call's serializer thread. This ensures that the channel will not go away on us. Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89	10 years ago
Mark Michelson	4f43b85c92	Taskprocessors: Increase high-water mark ... In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce	10 years ago
Alexander Traud	d8d3991390	format: Register format-attribute module with cached formats. ... In Asterisk 13, cached formats are created before their corresponding format- attribute module is registered. Cached formats are involved when a local extension is called. Therefore, ast_format_generate_sdp_fmtp did not work on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, and format-attribute modules provided externally. ASTERISK-25160 #close Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354	10 years ago
Mark Michelson	367972e42d	res_pjsip distributor: Don't send 503 response to responses. ... When the SIP threadpool is backed up with tasks, we send 503 responses to ensure that we don't try to overload ourselves. The problem is that we were not insuring that we were not trying to send a 503 to an incoming SIP response. This change makes it so that we only send the 503 on incoming requests. Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404	10 years ago
Joshua Colp	cd51b0aeac	Merge "res_pjsip: Deny requests when threadpool queue is backed up." into 13	10 years ago
Matt Jordan	db93c357ce	Merge "format_cap: Don't append the 'none' format when appending all." into 13	10 years ago
Mark Michelson	2f9cb7d62b	res_pjsip: Deny requests when threadpool queue is backed up. ... We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816	10 years ago
Joshua Colp	7ae22c2690	Merge "Further fixes to improper usage of scheduler" into 13	10 years ago
Joshua Colp	4e5bf12b33	format_cap: Don't append the 'none' format when appending all. ... When appending all formats of a type all the codecs are iterated and added. This operation was incorrectly adding the ast_format_none format which is special in that it is supposed to be used when no format is present. It shouldn't be appended. ASTERISK-25535 Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c	10 years ago
Steve Davies	07583c2888	Further fixes to improper usage of scheduler ... When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b	10 years ago
Joshua Colp	b818d70533	threadpool: Handle worker thread transitioning to dead when going active. ... This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143	10 years ago
Matt Jordan	dac0bf063c	Merge "rtp_engine: Init a format-attribute module to its RFC defaults." into 13	10 years ago
Matt Jordan	e07f5a6133	Merge "res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP." into 13	10 years ago
Matt Jordan	e098fb1813	Merge "ast_format_cap: Avoid format creation on module load, use cache instead." into 13	10 years ago
Matt Jordan	bd157b9ca8	Merge "xmldoc: Improve xmldoc wrapping of 'core show ...' output." into 13	10 years ago
Alexander Traud	4bf84459c7	rtp_engine: Init a format-attribute module to its RFC defaults. ... Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52	10 years ago
Joshua Colp	18e61a6442	Merge "taskprocessor: Add high water mark warnings" into 13	10 years ago
Joshua Colp	3e0f161761	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased." into 13	10 years ago
Joshua Colp	07189ee2c9	Merge "func_callerid: Document that CALLERID(pres) is available." into 13	10 years ago
Alexander Traud	1bff400df7	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Alexander Traud	f3ac4d8090	ast_format_cap: Avoid format creation on module load, use cache instead. ... Since Asterisk 13, formats are immutable and cached. However while loading a module like chan_sip, some formats were created instead using cached ones. ASTERISK-25535 #close Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b	10 years ago
Walter Doekes	6d1bdb9d3b	func_callerid: Document that CALLERID(pres) is available. ... CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a	10 years ago
Walter Doekes	8410336681	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
Walter Doekes	0d425f2eb4	xmldoc: Improve xmldoc wrapping of 'core show ...' output. ... Previously, the wrapping did both lookahead and lookback, which, together with color escape sequences, caused some lines to be wrapped way earlier than other lines. This led to inconsistent output. This simplifies the wrapping code and makes it more sane: if maxcolumns is hit, we simply jump back to the last space and wrap there. ASTERISK-25527 #close Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957	10 years ago
Alexander Traud	33752e0837	res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP. ... In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual amount of channels is negotiated in-band. Therefore now, the Opus codec and its attribute rtpmap are registered with two channels. ASTERISK-24779 #close Reported by: PowerPBX Tested by: Alexander Traud patches: asterisk-24779.patch submitted by Sean Bright (license #5060) Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b	10 years ago
Jonathan Rose	6ff48319d9	taskprocessor: Add high water mark warnings ... If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c	10 years ago
Matt Jordan	506aea26e6	main/dial: Protect access to the format_cap structure of the requesting channel ... When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f	10 years ago
Corey Farrell	d098d00424	Fix cli display of build options. ... A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325	10 years ago
Matt Jordan	afec1b1b64	res_pjsip/location: Destroy contact_status objects on contact deletion ... The contact_status Sorcery objects are currently not destroyed when a contact is deleted. This causes the contact's last known RTT/status to be 'sticky' when the contact itself may no longer exist. This patch causes the contact_status objects associated with both dynamic and static contacts to be destroyed if the AoR holding those contacts is also destroyed (or via other paths where a contact may be deleted.) Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e	10 years ago
Matt Jordan	1cf699c848	Merge "pjsip_configuration: On delete, remove the persistent version of an endpoint" into 13	10 years ago
Matt Jordan	562556c79f	Merge "main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field" into 13	10 years ago
Matt Jordan	715f770c9f	pjsip_configuration: On delete, remove the persistent version of an endpoint ... When an endpoint is deleted (such as through an API), the persistent endpoint currently continues to lurk around. While this isn't harmful from a memory consumption perspective - as all persistent endpoints are reclaimed on shutdown - it does cause Stasis endpoint related operations to continue to believe that the endpoint may or may not exist. This patch causes the persistent endpoint related to a PJSIP endpoint to be destroyed if the PJSIP endpoint is deleted. Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb	10 years ago
Matt Jordan	f0f190af08	main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field ... The JSON packing for the ContactStatusChange event forgot to include the roundtrip_usec field. As a result, the field never showed up in any event, even when the data was available. This patch corrects that error by properly packing the JSON blob with the data. Change-Id: I8df80da659a44010afbd48f645967518ff5daa17	10 years ago
Corey Farrell	0393bd6bed	chan_sip: Allow websockets to be disabled. ... This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7	10 years ago
Mark Michelson	6fbffe42e1	res_pjsip: Set threadpool max size default to 50. ... During a stress test of subscriptions, a huge blast of subscription-related traffic resulted in the threadpool expanding to a ridiculous number of threads. The balooning of threads resulted in an increase of memory, which led to a crash due to being out of memory. An easy fix for the particular test was to limit the size of the threadpool, thus reining in the amount of memory that would be used. It was decided that there really is no downside to having a non-infinite default value for the maximum size of the threadpool, so this change introduces 50 threads as the maximum threadpool size for the SIP threadpool. ASTERISK-25513 #close Reported by John Bigelow Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be	10 years ago
Joshua Colp	0071a993f0	Merge "pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction" into 13	10 years ago