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Author	SHA1	Message	Date
Richard Mudgett	a433bb38b5	heap.c: No need to calloc heap pointer array. ... Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68	8 years ago
Jenkins2	b6d5e9223c	Merge "pjsip_message_filter: Fix regression causing bad contact address" into 13	8 years ago
Joshua Colp	f4daa91dcb	Merge "pjproject: Patch to correct STUN FINGERPRINT usage" into 13	8 years ago
George Joseph	d70d7b2f5d	pjsip_message_filter: Fix regression causing bad contact address ... The "res_pjsip: Filter out non SIP(S) requests" commit moved the filtering of messages to pjproject's PJSIP_MOD_PRIORITY_TRANSPORT_LAYER in order to filter out incoming bad uri schemes as early as possible. Since the change affected outgoing messages as well and the TRANSPORT layer is the last to be run on outgoing messages, we were overwriting the setting of external_signaling_address (which is set earlier by res_pjsip_nat) with an internal address. * pjsip_message_filter now registers itself as a pjproject module twice. Once in the TSX layer for the outgoing messages (as it was originally), then a second time in the TRANSPORT layer for the incoming messages to catch the invalid uri schemes. ASTERISK-27295 Reported by: Sean Bright Change-Id: I2c90190c43370f8a9d1c4693a19fd65840689c8c	8 years ago
Richard Mudgett	221d8a5c24	res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential. ... The bridge_p2p_rtp_write() has potential reentrancy problems. * Accessing the bridged RTP members must be done with the instance1 lock held. The DTMF and asymmetric codec checks must be split to be done with the correct RTP instance struct locked. i.e., They must be done when working on the appropriate side of the point to point bridge. * Forcing the RTP mark bit was referencing the wrong side of the point to point bridge. The set mark bit is used everywhere else to set the mark bit when sending not receiving. The patches for ASTERISK_26745 and ASTERISK_27158 did not take into account that not everything carried by RTP uses a codec. The telephony DTMF events are not exchanged with a codec. As a result when RFC2833/RFC4733 sent digits you would crash if "core set debug 1" is enabled, the DTMF digits would always get passed to the core even though the local native RTP bridge is active, and the DTMF digits would go out using the wrong SSRC id. * Add protection for non-format payload types like DTMF when updating the lastrxformat and lasttxformat. Also protect against non-format payload types when checking for asymmetric codecs. ASTERISK-27292 Change-Id: I6344ab7de21e26f84503c4d1fca1a41579364186	8 years ago
Jenkins2	a40978f865	Merge "channel.c: Fix invalid reference in conditionaled out code." into 13	8 years ago
Jenkins2	df17e01274	Merge "app_queue: Only do announcement logic between ringing cycles" into 13	8 years ago
Sean Bright	f3b1b64d21	pjproject: Patch to correct STUN FINGERPRINT usage ... Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6	8 years ago
Joshua Colp	7977daa0cc	Merge "build: A few gcc 7 error fixes" into 13	8 years ago
Richard Mudgett	8d2c3effc2	channel.c: Fix invalid reference in conditionaled out code. ... ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c	8 years ago
George Joseph	690f7f7c76	build: A few gcc 7 error fixes ... Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec	8 years ago
Sean Bright	f39af4d36d	res_pjsip: Use ast_sip_is_content_type() where appropriate ... Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d	8 years ago
Jenkins2	9576ae0e7e	Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1" into 13	8 years ago
Jenkins2	fef8b6efec	Merge "res_srtp: lower log level of auth failures" into 13	8 years ago
Rodrigo Ramírez Norambuena	c98e980fff	res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1 ... In PostgreSQL 9.1 the backslash are string literals and not the escape of characters. In previous issue ASTERISK_26057 was fixed the use of escape LIKE but the support for old version of Postgresql than 9.1 was dropped. The sentence before make was "ESCAPE '\'" but in version before than 9.1 need it to be as follow "ESCAPE '\\'". ASTERISK-27283 Change-Id: I96d9ee1ed7693ab17503cb36a9cd72847165f949	8 years ago
Joshua Colp	2f8e7be0d1	Merge "chan_sip: Expose read-only access to the full SIP INVITE Request-URI" into 13	8 years ago
StefanEng86	0adf6f3bd9	app_queue: Only do announcement logic between ringing cycles ... This patch reverts the change by patch 2263 from old reviewboard. Note that reverting that 2263-patch still preserves the behaviour that the commit log of the 2263-patch claimed to add. The reason for this is: The function wait_for_answer is only called from try_calling which in turn is only called from the main for loop in queue_exec, and earlier in that loop we already check the things that's removed by this patch. There's no need to check those things twice each loop iteration, and I think the proper place to check it is before each ringing cycle. By checking it in wait_for_answer, you allow the issue explained in the jira - that the head caller hears announcements while the agents' sip phones are actively ringing. Reported-by: Stefan Engström Tested-by: Stefan Engström ASTERISK-27216 #close Change-Id: Ic4290dc75256f9743900c6762ee1bb915f672db0	8 years ago
Jean Aunis	da40976987	bridge : Fix one-way direct-media when early bridging with native_rtp ... When two channels were early bridged in a native_rtp bridge, the RTP description on one side was not updated when the other side answered. This patch forbids non-answered channels to enter a native_rtp bridge, and triggers a bridge reconfiguration when an ANSWER frame is received. ASTERISK-27257 Change-Id: If1aaee1b4ed9658a1aa91ab715ee0a6413b878df	8 years ago
Jenkins2	4bde3d8634	Merge "res_pjsip_pubsub: Check for Content-Type header in rx_notify_request" into 13	8 years ago
George Joseph	828a0611bc	res_pjsip_pubsub: Check for Content-Type header in rx_notify_request ... pubsub_on_rx_notify_request wasn't checking for a null Content-Type header before checking that it was application/simple-message-summary. ASTERISK-27279 Reported by: Ross Beer Change-Id: Iec2a6c4d2e74af37ff779ecc9fd35644c5c4ea52	8 years ago
Jenkins2	2f11ea59db	Merge "AST-2017-008: Improve RTP and RTCP packet processing." into 13	8 years ago
David J. Pryke	94f616e5e2	chan_sip: Expose read-only access to the full SIP INVITE Request-URI ... Provide a way to get the contents of the the Request URI from the initial SIP INVITE in dial plan function call. (In this case "${CHANNEL(ruri)}") ASTERISK-27278 Reported by: David J. Pryke Tested by: David J. Pryke Change-Id: I1dd4d6988eed1b6c98a9701e0e833a15ef0dac3e	8 years ago
Joshua Colp	839c35adab	Merge "res_calendar: On reload, update all configuration" into 13	8 years ago
Jenkins2	1a35e2b8c1	Merge "cdr_mysql.c: Apply cdrzone to start and answer" into 13	8 years ago
Alexander Traud	cfc0ca1fb5	tcptls: Fixed a white space error. ... ASTERISK-26606 Change-Id: I81a7268ef7ba012d4d80d44c70b6276d48e397fa	8 years ago
Alexander Traud	99a08eb7ab	res_srtp: lower log level of auth failures ... Previously, sRTP authentication failures were reported on log level WARNING. When such failures happen, each RT(C)P packet is affected, spamming the log. Now, those failures are reported at log level VERBOSE 2. Furthermore, the amount is further reduced (previously all two seconds, now all three seconds). Additionally, the new log entry informs whether media (RTP) or statistics (RTCP) are affected. ASTERISK-16898 #close Change-Id: I6c98d46b711f56e08655abeb01c951ab8e8d7fa0	8 years ago
alex	f1eb36ea51	cdr_mysql.c: Apply cdrzone to start and answer ... Change-Id: I7de0a5adc89824a5f2b696fc22c80fc22dff36b0	8 years ago
Richard Mudgett	6d4b801c83	AST-2017-008: Improve RTP and RTCP packet processing. ... Validate RTCP packets before processing them. * Validate that the received packet is of a minimum length and apply the RFC3550 RTCP packet validation checks. * Fixed potentially reading garbage beyond the received RTCP record data. * Fixed rtp->themssrc only being set once when the remote could change the SSRC. We would effectively stop handling the RTCP statistic records. * Fixed rtp->themssrc to not treat a zero value as special by adding rtp->themssrc_valid to indicate if rtp->themssrc is available. ASTERISK-27274 Make strict RTP learning more flexible. Direct media can cause strict RTP to attempt to learn a remote address again before it has had a chance to learn the remote address the first time. Because of the rapid relearn requests, strict RTP could latch onto the first remote address and fail to latch onto the direct media remote address. As a result, you have one way audio until the call is placed on and off hold. The new algorithm learns remote addresses for a set time (1.5 seconds) before locking the remote address. In addition, we must see a configured number of remote packets from the same address in a row before switching. * Fixed strict RTP learning from always accepting the first new address packet as the new stream. * Fixed strict RTP to initialize the expected sequence number with the last received sequence number instead of the last transmitted sequence number. * Fixed the predicted next sequence number calculation in rtp_learning_rtp_seq_update() to handle overflow. ASTERISK-27252 Change-Id: Ia2d3aa6e0f22906c25971e74f10027d96525f31c	8 years ago
Jenkins2	b6e1b13de4	Merge "res_pjsip: Filter out non SIP(S) requests" into 13	8 years ago
Sean Bright	5075cc8eed	res_calendar: On reload, update all configuration ... This changes the behavior of res_calendar to drop all existing calendars and re-create them whenever a reload is done. The Calendar API provides no way for configuration information to be pushed down to calendar 'techs' so updated settings would not take affect until a module unload/load was done or Asterisk was restarted. Asterisk 15+ already has a configuration option 'fetch_again_at_reload' that performs a similar function. Also fix a tiny memory leak in res_calendar_caldav while we're at it. ASTERISK-25524 #close Reported by: Jesper Change-Id: Ib0f8057642e9d471960f1a79fd42e5a3ce587d3b	8 years ago
Jenkins2	14109355f3	Merge "res_calendar: Various fixes" into 13	8 years ago
George Joseph	63900374fa	res_pjsip: Filter out non SIP(S) requests ... Incoming requests with non sip(s) URIs in the Request, To, From or Contact URIs are now rejected with PJSIP_SC_UNSUPPORTED_URI_SCHEME (416). This is performed in pjsip_message_filter (formerly pjsip_message_ip_updater) and is done at pjproject's "TRANSPORT" layer before a request can even reach the distributor. URIs read by res_pjsip_outbound_publish from pjsip.conf are now also checked for both length and sip(s) scheme. Those URIs read by outbound registration and aor were already being checked for scheme but their error messages needed to be updated to include scheme failure as well as length failure. Change-Id: Ibb2f9f1d2dc7549da562af4cbd9156c44ffdd460	8 years ago
Jenkins2	77c6bab990	Merge "chan_rtp: Use μ-law by default instead of signed linear" into 13	8 years ago
Jenkins2	df7211421e	Merge "res_pjsip: Add handling for incoming unsolicited MWI NOTIFY" into 13	8 years ago
Sean Bright	db785ddb92	res_calendar: Various fixes ... * The way that we were looking at XML elements for CalDAV was extremely fragile, so use SAX2 for increased robustness. * Don't complain about a 'channel' not be specified if autoreminder is not set. Assume that if 'channel' is not set, we don't want to be notified. * Fix some truncated CLI output in 'calendar show calendar' and make the 'Autoreminder' description a bit more clear ASTERISK-24588 #close Reported by: Stefan Gofferje ASTERISK-25523 #close Reported by: Jesper Change-Id: I200d11afca6a47e7d97888f286977e2e69874b2c	8 years ago
Sean Bright	0688f61a01	chan_rtp: Use μ-law by default instead of signed linear ... Multicast/Unicast RTP do not use SDP so we need to use a format that cleanly maps to one of the static RTP payload types. Without this change, an Originate to a Multicast or Unicast channel without a format specified would produce no audio on the receiving device. ASTERISK-21399 #close Reported by: Tzafrir Cohen Change-Id: I97e332b566e85da04b0004b9b0daae746cfca0e3	8 years ago
George Joseph	ed2a4ee81e	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	8 years ago
Jenkins2	1aef4fe9df	Merge "alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode" into 13	8 years ago
Richard Mudgett	044674c0cd	res_rtp_asterisk.c: Add doxygen to RTCP payload types. ... Change-Id: I3f20ce428777cc4ce9c13b2f808d29ff8c873998	8 years ago
Jenkins2	55af8326b5	Merge "alembic: Add support for MS-SQL" into 13	8 years ago
George Joseph	5ff2d06aa6	alembic: Fix typo in add_auto_info_to_endpoint_dtmf_mode ... The downgrade function was missing "_v2" at the end of the alter column type. Change-Id: Iaa9bcef48d6f3590ce07a61342d8e66f00263d8e	8 years ago
Walter Doekes	babb617f20	res/res_pjsip: Fix localnet checks in pjsip, part 2. ... In 45744fc53, I mistakenly broke SDP media address rewriting by misinterpreting which address was checked in the localnet comparison. Instead of checking the remote peer address to decide whether we need media address rewriting, we check our local media address: if it's local, then we rewrite. This feels awkward, but works and even made directmedia work properly if you set local_net. (For the record: for local peers, the SDP media rewrite code is not called, so the comparison does no harm there.) ASTERISK-27248 #close Change-Id: I566be1c33f4d0a689567d451ed46bab9c3861d4f	8 years ago
Florian Floimair	ad606844be	alembic: Add support for MS-SQL ... MS-SQL has no native Enum-type support and therefore needs to work with constraints. Since these constraints need unique names the suggested approach referenced in the following alembic documentation has been applied: http://bit.ly/2x9r8pb ASTERISK-27255 #close Change-Id: I4a399ba3eed41a33ce8cb294968ad340221580ee	8 years ago
Jenkins2	63211cc0ff	Merge "app_waitforsilence: Cleanup & don't treat missing frames as 'noise'" into 13	8 years ago
Joshua Colp	2971c2f333	Merge "chan_sip: when getting sip pvt return failure if not found" into 13	8 years ago
Jenkins2	584f6abc4e	Merge "res_srtp: Add support for libsrtp2.1." into 13	8 years ago
Jenkins2	bab59a8190	Merge "chan_sip: Do not change IP address in SDP origin line (o=) in SIP reINVITE" into 13	8 years ago
Jenkins2	1161fce1a1	Merge "func_cdr: honour 'u' flag on dummy channel" into 13	8 years ago
Jacek Konieczny	2aefc6e5fe	func_cdr: honour 'u' flag on dummy channel ... Fixes ${CDR(...,u)} when used in cdr_custom.conf ASTERISK-27165 #close Change-Id: Ia4e0b6ba93e03d27886354c279737790e2cd6a83	8 years ago
Scott Griepentrog	c0d4f1880e	chan_sip: when getting sip pvt return failure if not found ... In handle_request_invite, when processing a pickup, a call is made to get_sip_pvt_from_replaces to locate the pvt for the subscription. The pvt is assumed to be valid when zero is returned indicating no error, and is dereferenced which can cause a crash if it was not found. This change checks the not found case and returns -1 which allows the calling code to fail appropriately. ASTERISK-27217 #close Reported-by: Bryan Walters Change-Id: I6bee92b8b8b85fcac3fd66f8c00ab18bc1765612	8 years ago