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Author	SHA1	Message	Date
Matt Jordan	25332911fe	Merge "res/res_endpoint_stats: Add module to emit endpoint StatsD statistics" into 13	10 years ago
Matt Jordan	6a7cb60a47	Merge "res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts" into 13	10 years ago
Matt Jordan	16437667eb	Merge "res/res_pjsip_outbound_registration: Add registration statistics for StatsD" into 13	10 years ago
Matt Jordan	001b7f482b	Merge "res_statsd: Add functions that support variable arguments" into 13	10 years ago
Matt Jordan	0f88f909ec	Merge "StatsD: Add res_statsd compatibility" into 13	10 years ago
Matt Jordan	4875e5ac32	chan_pjsip: Handle T.38 faxes with direct media bridges ... When a channel is in a direct media bridge, a re-INVITE may arrive that forces Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge must change its technology to a simple bridge, and re-INVITE the media back to Asterisk. Generally, this logic mostly already exists in Asterisk. However, prior to this patch, there were a few bugs: (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from ever entering into a direct media bridge. This applies even when the only media being passed over the channel is audio. This patch fixes this bug by having the framehook specify that it defers caring about any frame type. This allows the channels to enter into a direct media bridge, which will be broken when a re-INVITE is received. (2) When a re-INVITE is received, nothing instructed the bridging layer to re-inspect the allowed bridging technology. This now occurs when either a re-INVITE is received from a peer, or when a response is received from the far end (that is, when the T.38 state changes to either T38_PEER_REINVITE or T38_LOCAL_REINVITE). (3) chan_pjsip needs to do a small amount of work to prevent a direct media bridge from being chosen when a T.38 session is in progress. When a T.38 session supplement has a t38 datastore - which is added when we detect we should start thinking about T.38 on a channel - we now refuse a native RTP bridge. (4) When a BYE request is received, we don't terminate the T.38 session. If the other side of a T.38 fax survives the hangup (due to the 'g' flag in Dial, for example), we don't currently re-INVITE the media on the other channel back to audio. This patch now has res_pjsip_t38 intercept BYE requests and inform the far side that the T.38 session is terminated. This naturally causes the correct re-INVITEs to be sent. ASTERISK-25582 Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb	10 years ago
Joshua Colp	fa969196b3	Merge "main/cli: Use proper string methods to check existence of context/exten/app" into 13	10 years ago
Joshua Colp	6dd8b67216	Merge "res/res_pjsip_t38: Add debug statements" into 13	10 years ago
Matt Jordan	aa8f1b04b6	Merge "res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts." into 13	10 years ago
Matt Jordan	2b94d9a10d	res/res_pjsip_t38: Add debug statements ... This patch adds some debug statements to res_pjsip_t38. These statements help to determine which SDP negotiation callbacks are being executed, and, when a particular callback exits, why a callback may not have applied its logic to the local or remote SDP. Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77	10 years ago
Matt Jordan	af288b2d96	main/cli: Use proper string methods to check existence of context/exten/app ... Because the context, extension, and application are stored in stringfields, checking for them being NULL doesn't work so well. This patch uses the appropriate string library call, ast_strlen_zero, to see if there is a value in the context/exten/app values. Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23	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	d27aac0a9d	res/res_endpoint_stats: Add module to emit endpoint StatsD statistics ... This patch adds a module that emits StatsD statistics about Asterisk endpoints. This includes: * A GUAGE statistic for endpoint states, tracking how many endpoints are in a particular state. * A GUAGE statistic for each endpoint, counting the number of channels currently associated with an endpoint. ASTERISK-25572 Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305	10 years ago
Matt Jordan	90d9a70789	res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts ... This patch adds the ability to send StatsD statistics related to the state of PJSIP contacts. This includes: * A GUAGE statistic measuring the count of contacts in a particular state. This measures how many contacts are reachable, unreachable, etc. * The RTT time for each contact, if those contacts are qualified. This provides StatsD engines useful time-based data about each contact. ASTERISK-25571 Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c	10 years ago
Matt Jordan	75097a0955	res/res_pjsip_outbound_registration: Add registration statistics for StatsD ... This patch adds outbound registration statistics for StatsD. This includes the following: * A GUAGE metric for the overall count of outbound registrations. * A GUAGE metric for each state an outbound registration can be in. As the outbound registrations change state, the overall count of how many outbound registrations are in the particular state is changed. These statistics are particularly useful for systems with a large number of SIP trunks, and where measuring the change in state of the trunks is useful for monitoring. ASTERISK-25571 Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37	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	0b508789ab	translate: Provide translation modules the result of SDP negotiation. ... Previously, a trancoding module did not have access to the joint but cached format. Therefore, the module did not have access to the attributes negotiated via SDP (line fmtp). Now, a translation module receives the joint format. ASTERISK-25545 #close Change-Id: Id6878a989b50573298dab115d3371ea369e1a718	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
Matt Jordan	3354b325c6	res_statsd: Add functions that support variable arguments ... Often, the metric names of statistics we are generating for StatsD have some dynamic component to them. This can be the name of a particular resource, or some internal status label in Asterisk. With the current set of functions, callers of the statsd API must first build the metric name themselves, then pass this to the API functions. This results in a large amount of boilerplate code and usage of either fixed length static buffers or dynamic memory allocation, neither of which is desireable. This patch adds two new functions to the StatsD API that support a printf style format specifier for constructing the metric name. A dynamic string, allocated in threadstorage, is used to build the metric name. This eases the burden on users of the StatsD API. Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea	10 years ago
Richard Mudgett	d4a522d587	res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. ... Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d	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
tcambron	1e0040b88f	StatsD: Add res_statsd compatibility ... Added a new api to res_statsd.c to allow it to receive a character pointer for the value argument. This allows for a '+' and a '-' to easily be sent with the value. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611	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