29899 Commits (a0a1f95abfe04d51a52cfb3b9db1e69a9e12c659)

 

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Author	SHA1	Message	Date
hajekd	a0a1f95abf	res/res_ari.c Fix: Memory leaks in ARI when using Content-Type: application/json ... ASTERISK-27305 Reported by: David Hajek Tested by: David Hajek Change-Id: Ife3e289062e6cf7d0e7d342dbf79ed96feff441e	8 years ago
Joshua Colp	79deaa6a96	Merge "vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED."	8 years ago
Jenkins2	98076de152	Merge "main/strings: Fix uninitialized value."	8 years ago
Jenkins2	d4be387d41	Merge "res_pjsip: Fix leak of fake_auth references."	8 years ago
Corey Farrell	3bd00c4a7e	vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED. ... Use temporary variable to prevent multiple evaluations of elem argument. This resolves a memory leak in res_pjproject startup. ASTERISK-27317 #close Change-Id: Ib960d7f5576f9e1a3c478ecb48995582a574e06d	8 years ago
Corey Farrell	b35ac9e566	res_pjsip: Fix leak of fake_auth references. ... pjsip_distributor leaks references to fake_auth when the default realm has not changed. ASTERISK-27306 Change-Id: I3fcf103b3680ad2d1d4610dcd6738eeaebf4d202	8 years ago
Corey Farrell	0f3e725503	main/strings: Fix uninitialized value. ... ast_strings_match uses sscanf and checks for non-zero return to verify a token was parsed. This is incorrect as sscanf returns EOF (-1) for errors. ASTERISK-27318 #close Change-Id: Ifcece92605f58116eff24c5a0a3b0ee08b3c87b1	8 years ago
Daniel Tryba	59b6e8467a	res_pjsip_caller_id chan_sip: Comply to RFC 3323 values for privacy ... Currently privacy requests are only granted if the Privacy header value is exactly "id" (defined in RFC 3325). It ignores any other possible value (or a combination there of). This patch reverses the logic from testing for "id" to grant privacy, to testing for "none" and granting privacy for any other value. "none" must not be used in combination with any other value (RFC 3323 section 4.2). ASTERISK-27284 #close Change-Id: If438a21f31a962da32d7a33ff33bdeb1e776fe56	8 years ago
Joshua Colp	3ef6834c3b	Merge "app_queue.c: Fix announcements when announce-to-first-user not enabled."	8 years ago
krells	2301447a20	res_calendar_icalendar: Filter out occurrences superceded by another VEVENT ... When we are loading the calendars, we call libical's icalcomponent_foreach_recurrence method for each VEVENT component that we have in our calendar. That method has no knowledge concerning the existence of the other VEVENT components and will feed our callback with all ocurrences matching the requested time span. The occurrences generated by icalcomponent_foreach_recurrence while expanding a recurring VEVENT's RRULE and RDATE properties can be superceded by an other VEVENT sharing the same UID. I use an external iterator (in libical terminology) to avoid messing with the internal ones from the calling function, and search for VEVENTS which could supersede the current occurrence. The event which can invalidate this occurence needs to have: - the same UID as our recurrent component (comp) - a RECURRENCE-ID property, which represents the start time of this occurrence If one component is found, just clean and return. ASTERISK-27296 #close Reported by: Benoît Dereck-Tricot Change-Id: I8587ae3eaa765af7cb21eda3b6bf84e8a1c87af8	8 years ago
Jenkins2	7156477ba0	Merge "heap.c: No need to calloc heap pointer array."	8 years ago
Joshua Colp	323762abf3	Merge "logger: Bring back ability to turn debug on by source file"	8 years ago
Richard Mudgett	b2dbfe23ef	app_queue.c: Fix announcements when announce-to-first-user not enabled. ... The previous patch for ASTERISK-27216 made it so you wouldn't get any position or periodic announcements unless you had announce-to-first-user enabled. The announce-to-first-user feature was added by ASTERISK_21782 as a result of the patch which introduced the redundant announcements that ASTERISK-27216 removes. * By noting that the makeannouncement variable is used to suppresses the first user announcement, we set its initial value to the announce-to-first-user enable setting. ASTERISK-27216 Change-Id: Ieaeb7dbea8ae7073086b775fbafe0625b000b10a	8 years ago
Richard Mudgett	80097676e7	heap.c: No need to calloc heap pointer array. ... Change-Id: I5ae2f316229f336eb90d99c7af7ed07a33097e68	8 years ago
Jenkins2	5307659e96	Merge "pjsip_message_filter: Fix regression causing bad contact address"	8 years ago
Joshua Colp	47d68401d7	Merge "res_stasis: Add 'video_sfu' as a requested bridge type."	8 years ago
Joshua Colp	dd4c573e6d	Merge "res_pjsip_session: outgoing call did not offer all configured codecs"	8 years ago
George Joseph	d1de7948fe	logger: Bring back ability to turn debug on by source file ... Somewhere along the way we lost the ability to debug individual source files. For modules, this wasn't a big deal but all the source files in ./main are in the one "core" module so debugging individual core capabilities was almost impossible. * Added a test to DEBUG_ATLEAST that also checks __FILE__ instead of just module name. Any source file will work even if it's in a module subdirectory. Change-Id: Icc0af41837f3b1679dec7af21fa32cd1f7469f6e	8 years ago
Jenkins2	7c5e55f15d	Merge "pjproject: Patch to correct STUN FINGERPRINT usage"	8 years ago
Joshua Colp	a78ffe96a8	Merge "res_rtp_asterisk.c: Fix bridge_p2p_rtp_write() reentrancy potential."	8 years ago
Joshua Colp	4ea4364eef	Merge "res_rtp_asterisk: Trim trailing byte off of SDES packet"	8 years ago
Joshua Colp	f21408c866	res_stasis: Add 'video_sfu' as a requested bridge type. ... This change adds 'video_sfu' as a requested bridge type when creating a bridge. By specifying this a mixing type bridge is created that exchanges video in an SFU fashion. Change-Id: I2ada47cf5f3fc176518b647c0b4aa39d55339606	8 years ago
Richard Mudgett	a6dc0527a2	res_pjsip_outbound_publish.c: Fix misplaced parenthesis. ... The pjsip_publishc_init() call was referenced with a misplaced parentheses. As a result, outbound publication messages went out with an expiration of 1 second. ASTERISK-27298 Change-Id: I93622eabc8ee83e7a22e98c107f921284c605a08	8 years ago
George Joseph	61ea872233	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	9d65057cdf	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
Sean Bright	c9e972a26a	res_rtp_asterisk: Trim trailing byte off of SDES packet ... This could have been fixed by subtracting 1 from the final value of 'len' but the way the packet was being constructed was confusing so I took the opportunity to (I think) make it more clear. We were sending 1 extra byte at the end of the SDES RTCP packet which caused Chrome to complain (in its debug log): Too little data (1 byte) remaining in buffer to parse RTCP header (4 bytes). We now send the correct number of bytes. Change-Id: I9dcf087cdaf97da0374ae0acb7d379746a71e81b	8 years ago
Joshua Colp	c8a8933558	Merge "webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file"	8 years ago
Joshua Colp	f9bcd3287d	Merge "channel.c: Fix invalid reference in conditionaled out code."	8 years ago
Joshua Colp	2a09b3b76a	Merge "app_queue: Only do announcement logic between ringing cycles"	8 years ago
Joshua Colp	87c4a72f16	Merge "res_pjsip_session: Reduce (and improve) SDP renegotiation."	8 years ago
Sean Bright	721947ebae	webrtc: Allow 'webrtc' to be set on endpoints without dtls_ca_file ... If using a legitimate certificate from a trusted certificate authority, you don't need to provide CA file. Change-Id: I8623973b4209b44889243716d7880274caed8a6d	8 years ago
Sean Bright	0cbeaa5589	pjproject: Patch to correct STUN FINGERPRINT usage ... Change-Id: I0e453253dff1388b0186b36c754457c1d0d12db6	8 years ago
Joshua Colp	0133d29c83	Merge "build: A few gcc 7 error fixes"	8 years ago
Kevin Harwell	b74cbadd05	res_pjsip_session: outgoing call did not offer all configured codecs ... For some scenarios when an outgoing call was made only a subset of the configured codecs were offered. If the codecs being offered happened to not have a codec supported by the phone then the call would fail. For instance Alice and Bob both are configured in Asterisk for g722 and ulaw( allow=!all,g722,ulaw). Alice's endpoint however only supports g722 while Bob's only supports ulaw. When Alice calls Bob, Alice negotiates g722 fine with Asterisk. But when Asterisk sends the outgoing offer to Bob it only contains g722 and not both g722 and ulaw, so the call ends. This patch makes it so all the audio codecs configured on the endpoint always get sent, and not just a subset. However priority is given to those codecs that are compatible with the "other side". ASTERISK-27259 #close Change-Id: Iffabc373bd94cd1dc700925dcfe406e12918c696	8 years ago
Joshua Colp	360bd11c30	Merge "res_pjsip_session: Don't end session when receiving a 500 on a reinvite"	8 years ago
Richard Mudgett	08e67f814b	channel.c: Fix invalid reference in conditionaled out code. ... ASTERISK-27289 Change-Id: I7a415948116493050614d9f4fa91ffbe0c21ec4c	8 years ago
Joshua Colp	9cb5899781	Merge "app_stream_echo: Don't echo declined streams"	8 years ago
George Joseph	4275ca16a1	build: A few gcc 7 error fixes ... Change-Id: I7b5300fbf1af7d88d47129db13ad6dbdc9b553ec	8 years ago
Joshua Colp	1b468b543d	Merge "res_pjsip: Use ast_sip_is_content_type() where appropriate"	8 years ago
StefanEng86	c3c73b3511	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
Sean Bright	0fad11f21c	app_stream_echo: Don't echo declined streams ... Discovered while experimenting with Cyber Mega Phone 2K Ultimate Dynamic Edition after accepting the audio request but declining the video one. Change-Id: Iaa86d41fccfbc1b559a30ccf740d78a3b5f8a98c	8 years ago
Joshua Colp	601e0c563f	res_pjsip_session: Reduce (and improve) SDP renegotiation. ... When pruning a request to change the topology of a channel be more intelligent about the resulting topology that is actually used for SDP renegotiation. In a case where a stream has not already been negotiated we don't need to renegotiate and offer a declined stream. This can occur if something in Asterisk (such as ConfBridge) requests to add video to a PJSIP channel that has no video codecs configured. In this case since the stream did not already exist we can safely remove the stream from the requested topology, resulting in no renegotiation occurring. In a case where a renegotiation is requested with a codec that is not supported we can reuse the formats of the existing stream if it exists to ensure that the stream continues to flow, instead of removing it. Change-Id: I636540798d55922377318fe619c510fb6ed125fb	8 years ago
Kevin Harwell	36690c26f8	res_pjsip_session: Don't end session when receiving a 500 on a reinvite ... During a reinvite, if a remote endpoint error occurs and it returns a 500 the session would end. This patch makes it so the session is not terminated, but continues as it was. The reason for this is because some endpoints may send non session terminating "server errors" like a failed codec negotiation. So in this case instead of ending the call it can hopefully continue. In the case of a real server error the session is already "doomed", will be known soon enough and appropriately ended by Asterisk later. Change-Id: Ifeedae86b8cb44b92d52c79046522ec5f0aff1d5	8 years ago
Joshua Colp	f91a905003	Merge "res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint"	8 years ago
Joshua Colp	feadd654af	Merge "res_pjsip_session: Change some asserts to warning/debug messages"	8 years ago
Sean Bright	ebd0a4bebf	res_pjsip: Use ast_sip_is_content_type() where appropriate ... Change-Id: If3ab0d73d79ac4623308bd48508af2bfd554937d	8 years ago
George Joseph	6c0e13da22	res_pjsip_session/BUNDLE: Handle no audio codecs on endpoint ... When an INVITE came in with both audio and video streams but there were no audio codecs defined for the endpoint, we weren't declining the audio stream. Since it's usually the first/transport stream, when the video stream was processed and tried to use the transport, it was empty and caused a crash. We now decline the the stream if there are no matching codecs so when the video stream is processed, it's now the first/transport stream and processes normally. Change-Id: Ic854eda54c95031e66b076ecfae3041d34daa692	8 years ago
Joshua Colp	63a3d17f0b	Merge "res_rtp_asterisk.c: Fix bundled SSRC handling."	8 years ago
Joshua Colp	8b7aed0025	Merge "res_config_pgsql: Fix removed support to previous for versions PostgreSQL 9.1"	8 years ago
Joshua Colp	f77a2367e7	Merge "bridge: Change participant SFU streams when source streams change."	8 years ago