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Author	SHA1	Message	Date
Matt Jordan	65102d19bc	Merge "app_bridgeaddchan: ability to barge into existing call"	10 years ago
Alec Davis	8c14b91651	app_bridgeaddchan: ability to barge into existing call ... To be able to barge into a call by dialling a prefix+extension that maps to the extensions device. Senario is that DECT headset users may be away from their desks and need to transfer the call, the goal is that from any phone they dial a prefix then their extension and are added to the bridge that they are in, from there they can drop the headset call, as it's also on the handset, and transfer the caller. The dialplan would look like, where prefix=73, extension = 8512; exten => _738512,1,BridgeAdd(SIP/cisco0001) ASTERISK-25551 #close Reported By: Alec Davis Change-Id: I8eb5096a02168dcc8d7aeea416ef36ba4ed10540	10 years ago
tcambron	05addf3d8f	StatsD: Add sample rate compatibility ... Implemented support for the StatsD sample rate parameter, which is a parameter for determining when to send computed statistics to a client. Valid sample rate values are: Less than or equal to 0.0 will never be sent. Between 0.0 and 1.0 will randomly be sent. Greater than or equal to 1.0 will always be sent. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I11d315d0a5034fffeae1178e650aa8264485ed52	10 years ago
Richard Mudgett	3dbaf696e9	res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. ... Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d	10 years ago
Richard Mudgett	eaf898ac88	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	e90bb44b76	Merge "res_pjsip_rfc3326.c: Fix crash when channel goes away."	10 years ago
Alec Davis	4013f9d577	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
Matt Jordan	f57da2ae90	Merge "format: Register format-attribute module with cached formats."	10 years ago
Matt Jordan	e6ac74f0f7	Merge "res/res_pjsip: Fix off nominal crash with requests that fail and have a timer"	10 years ago
Joshua Colp	e18b07da9d	Merge "Confbridge: Add a user timeout option"	10 years ago
George Joseph	6919daab61	dns: Fix pointer increment in dns_parse_answer_ex ... When dns_parse_answer_ex was iterating over the answers it wasn't incrementing the answer pointer correctly after the first answer. The result was that no answers after the first were being returned. For results where multiple records should have been sorted by priority, weight, etc., there was nothing to sort so the only the first record was returned even if it wouldn't have been the correct record based on the sort. ASTERISK-25565 #close Reported-by: Daniel Tryba Tested-by George Joseph Change-Id: I8622604fefdcd3c11e2c5609a6382e53b1467b0b	10 years ago
Mark Michelson	ed13732188	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
Matt Jordan	a83e426e91	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
Joshua Colp	a1fcf6f7b2	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	436023a322	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	e8881e1770	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	fd23d423d8	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	40b58a5d2b	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	6879ce7fba	Merge "res_pjsip: Deny requests when threadpool queue is backed up."	10 years ago
Matt Jordan	1d4f813a07	Merge "format_cap: Don't append the 'none' format when appending all."	10 years ago
Mark Michelson	264c74aa22	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	469ad02404	Merge "Further fixes to improper usage of scheduler"	10 years ago
Joshua Colp	a159747660	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	d982b99e71	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	2954354404	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	c1abe8906d	Merge "rtp_engine: Init a format-attribute module to its RFC defaults."	10 years ago
Matt Jordan	21ee2f029c	Merge "Increase account code maximum length to 80."	10 years ago
Matt Jordan	9c9ebbd482	Merge "dns: Use ntohl for ans->ttl in dns_parse_answer_ex"	10 years ago
Matt Jordan	d77bba4326	Merge "res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP."	10 years ago
Matt Jordan	7987bf54c6	Merge "ast_format_cap: Avoid format creation on module load, use cache instead."	10 years ago
Matt Jordan	8b4b3e9b88	Merge "xmldoc: Improve xmldoc wrapping of 'core show ...' output."	10 years ago
Alexander Traud	525c7ab780	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	d8dadf4894	Merge "taskprocessor: Add high water mark warnings"	10 years ago
Joshua Colp	e4af14784e	Merge "Remove ABI compatibility stub functions."	10 years ago
Joshua Colp	0be147f713	Merge "ast_format_cap_get_names: To display all formats, the buffer was increased."	10 years ago
Corey Farrell	be93036a4e	Remove ABI compatibility stub functions. ... ABI compatibility stubs existed for ast_app_separate_args and ast_verbose, this is not needed in master. Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453	10 years ago
Corey Farrell	02a124eda5	Remove execute permission from dns_system_resolver.c ... Change-Id: I3185735db42064bab00d3e073aed703385a00bf4	10 years ago
Joshua Colp	131b750fc6	Merge "func_callerid: Document that CALLERID(pres) is available."	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Alexander Traud	e85f0c81af	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	7dd8f89a50	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	39daf9f066	docs: Fix a few typo's in app docs (more then, resourse). ... Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7	10 years ago
George Joseph	d82a4b098f	dns: Use ntohl for ans->ttl in dns_parse_answer_ex ... dns_parse_answer_ex was not converting ans->ttl from network by order to host byte order which was causing certain ttls it to go negative. In turn this was causing answer edit checks to fail. ASTERISK-25528 #close Reported-by: Daniel Tryba Tested-by: George Joseph Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037	10 years ago
Walter Doekes	74e7333317	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	9d6e917349	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	a2c2a8e1bb	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
Corey Farrell	cd5ae02812	Increase account code maximum length to 80. ... This increases the maximum length of account code's to match extensions. This ensures it is always possible to set an accountcode to ${EXTEN} without truncation. ASTERISK-23904 Reported by: Ben Merrills Change-Id: If122602304ce03362722eb213a3111b32da5eeb9	10 years ago
Joshua Colp	f12ebe3584	Merge "StatsD: Add res_statsd compatibility"	10 years ago
tcambron	379c041038	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	9c293b5104	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