29720 Commits (15fbcc74d8fd1b3655f4c34a280fe76665bd0c5c)

 

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Author	SHA1	Message	Date
Richard Mudgett	716abaf33d	SDP: Search for the ice-lite attribute in the right place. ... * Pulled finding the rtcp-mux attribute flag out of the ICE candidate for loop. Also ordered the RTCP ICE candidate skip test to fail earlier. Change-Id: I8905d9c68563027a46cd3ae14dbcc27e9c814809	8 years ago
Richard Mudgett	a95584d079	SDP: Set the remote c= line in RTP instance. ... Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c	8 years ago
Richard Mudgett	06265b8c8a	stream: Add ast_stream_topology_del_stream() and unit test. ... Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9	8 years ago
Richard Mudgett	0fdb99c268	SDP: Add t= line in sdp_create_from_state() ... Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141	8 years ago
Richard Mudgett	4797a8bb81	stream: Ignore declined streams for some topology calls. ... * Made ast_format_cap_from_stream_topology() not include any formats from declined streams. * Made ast_stream_topology_get_first_stream_by_type() ignore declined streams to return the first active stream of the type. * Updated unit tests to check these changes have the expected effect. Change-Id: Iabbc6a3e8edf263a25fd3056c3c614407c7897df	8 years ago
George Joseph	15c04db3bd	Merge "app_voicemail.c: Fix compile error when IMAP enabled."	8 years ago
George Joseph	f743e2e84c	Merge "app_voicemail: IMAP logout on MWI unsubscribe"	8 years ago
Jenkins2	2c3c862cee	Merge "res_pjsip_refer/session: Calls dropped during transfer"	8 years ago
Joshua Colp	bd16c3c524	channel: Fix reference counting in ast_channel_suppress. ... The ast_channel_suppress function wrongly decremented the reference count of the underlying structure used to keep track of what should be suppressed on a channel if the function was called multiple times on the same channel. This change cleans up the reference counting a bit so this no longer occurs. ASTERISK-27016 Change-Id: I2eed4077cb4916e6626f9f120b63b963acc5c136	8 years ago
George Joseph	54a08a2e43	Merge "res_rtp_asterisk: Fix ssrc change for rtcp srtp"	8 years ago
Jenkins2	4681b9baef	Merge "res_pjsip_session: Correct inverted test in session_outgoing_nat_hook"	8 years ago
Jenkins2	433e876317	Merge "res_pjsip_transport_websocket: Add NULL check in get_write_timeout"	8 years ago
George Joseph	c27acb4c99	Merge "pjproject_bundled: Use the asterisk github mirror for download"	8 years ago
Richard Mudgett	b8b0b61a24	app_voicemail.c: Fix compile error when IMAP enabled. ... Change-Id: I2703f15b4099b4210c68eccf293105d1975c1fc1	8 years ago
Alexei Gradinari	023eede265	app_voicemail: IMAP logout on MWI unsubscribe ... Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd	8 years ago
George Joseph	65ed2ea311	res_pjsip_pubsub: Fix reference to released endpoint ... destroy_subscription was attempting to get the id of the subscription tree's endpoint after we'd already called ao2_cleanup on it causing a segfault. Moved the cleanup until after the debug statement and since endpoint could also be NULL at this point, check for that as well. ASTERISK-27057 #close Reported-by: Ryan Smith Change-Id: Ice0a7727f560cf204d870a774c6df71e159b1678	8 years ago
George Joseph	ea3f8c6889	res_pjsip_session: Correct inverted test in session_outgoing_nat_hook ... There was a typo introduced in commit 776ffd77 which was preventing the transport's external media address from being used. ASTERISK-27024 #close Reported-by: Christopher van de Sande patches: patch.diff submitted by Florian Floimair (license 6892) Change-Id: I7ec617171eaa2d86d2680b00cf37d5088adafc27	8 years ago
Joshua Colp	c33168c147	Merge "CFLAGS for BIND8 support"	8 years ago
George Joseph	88f18faf2a	res_rtp_asterisk: Fix ssrc change for rtcp srtp ... It looks like there was a copy/paste error in ast_rtp_change_source where if there was a rtcp srtp instance, instead of updating its ssrc we were updating the srtp instance ssrc twice. ASTERISK-27022 #close Reported-by: Michael Walton Change-Id: Ic88f3aee7227b401c58745ac265ff92c19620095	8 years ago
Joshua Colp	d6386a8f0c	bridge: Add a deferred queue. ... This change adds a deferred queue to bridging. If a bridge technology determines that a frame can not be written and should be deferred it can indicate back to bridging to do so. Bridging will then requeue any deferred frames upon a new channel joining the bridge. This change has been leveraged for T.38 request negotiate control frames. Without the deferred queue there is a race condition between the bridge receiving the T.38 request negotiate and the second channel joining and being in the bridge. If the channel is not yet in the bridge then the T.38 negotiation fails. A unit test has also been added that confirms that a T.38 request negotiate control frame is deferred when no other channel is in the bridge and that it is requeued when a new channel joins the bridge. ASTERISK-26923 Change-Id: Ie05b08523f399eae579130f4a5f562a344d2e415	8 years ago
Kevin Harwell	9e53c30610	res_pjsip_refer/session: Calls dropped during transfer ... When doing an attended transfer it's possible for the transferer, after receiving an accepted response from Asterisk, to send a BYE to Asterisk, which can then be processed before Asterisk has time to start and/or complete the transfer process. This of course causes the transfer to not complete successfully, thus dropping the call. This patch makes it so any BYEs received from the transferer, after the REFER, that initiate a session end are deferred until the transfer is complete. This allows the channel that would have otherwise been hung up by Asterisk to remain available throughout the transfer process. ASTERISK-27053 #close Change-Id: I43586db79079457d92d71f1fd993be9a3b409d5a	8 years ago
George Joseph	b2fd7e5069	pjproject_bundled: Use the asterisk github mirror for download ... We now mirror the pjproject tarball and md5 at https://github.com/asterisk/third-party/tree/master/pjproject To improve download reliability, we now get the tarball from our mirror instead of from pjsip.org. ASTERISK-27052 #close Reported-by: 'alex' Change-Id: I60236587a8935bfa71fcc391f4e2ecb31918c08a	8 years ago
Alexei Gradinari	42f738e052	res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled ... If sending unsolicited mwi to all endpoints on startup is disabled (mwi_disable_initial_unsolicited=yes) do not need to create subscriptions. If there are many (thousands) realtime endpoints configured with unsolicited mwi and Vociemail Storage configured as ODBC or IMAP there will be huge number of DB/IMAP requests on startup. ASTERISK-26230 #close Change-Id: I50ae909639e3ee298b931a54def4b2b9e0fb86c5	8 years ago
Jenkins2	abe1dd3039	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing."	8 years ago
Jenkins2	5d3420a2de	Merge "BuildSystem: Add patches to allow building with recent LibreSSL"	8 years ago
Jenkins2	b17e05b53a	Merge "BuildSystem: Fix build on FreeBSD due to missing crypt.h"	8 years ago
Jenkins2	44a0edcf71	Merge "codecs.conf.sample: Fix max_bandwidth speling error"	8 years ago
Joshua Colp	ac5f40f797	Merge "eventfd: Disable during cross compilation"	8 years ago
Sean Bright	847087a4ff	codecs.conf.sample: Fix max_bandwidth speling error ... Reported by Sylvain Boily via asterisk-dev mailing list. Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d	8 years ago
Jørgen H	8d1f54b92e	res_pjsip_transport_websocket: Add NULL check in get_write_timeout ... Added check for NULL return value when calling ast_sorcery_retrieve_by_id in function get_write_timeout ASTERISK-27046 Change-Id: I9357717278da631c3a1cb502c412693929b0cb41	8 years ago
Guido Falsi	d27168d36f	BuildSystem: Add patches to allow building with recent LibreSSL ... Add some #if defined checks which allow building against LibreSSL. These patchess come from OpenBSD ports: https://cvsweb.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/patches/ ASTERISK-27043 #close Reported by: OpenBSD ports Change-Id: I2f6c08a5840b85ad4d2b75370b947ddde7a9a572	8 years ago
David M. Lee	fcb1a0d7e8	CFLAGS for BIND8 support ... Some systems (like macOS) require BIND_8_COMPAT to be defined so that the nameser libraries are, well, BIND8 compatible. Change-Id: If79fc27a64f90de1835b5aa3aadfa9be22bd16b0	8 years ago
Jenkins2	db5245f9fd	Merge "CHANGES: correct version for a new option 'refer_blind_progress'"	8 years ago
Guido Falsi	7b668297f3	BuildSystem: Fix build on FreeBSD due to missing crypt.h ... FreeBSD does not include a crypt.h include file. Definitions for crypt() and crypt_r() are in unistd.h ASTERISK-27042 #close Change-Id: Ib307ee5e384870c6af50efa89fb73722dd0c3a7e	8 years ago
Joshua Colp	5b80496b42	chan_pjsip: Update device state when in early media. ... The chan_pjsip module uses a calculation approach for determining device state. This means that in situations where we would expect device state to change we need to tell the core to query. A scenario that was missed is when early media was signaled. This change adds the notification for the core to query device state when we are told that early media is being provided. ASTERISK-27039 Change-Id: Iafebfd152894966344ff2e950a3cee9f59a3eb6f	8 years ago
Sean Bright	e497a76d24	eventfd: Disable during cross compilation ... Reported by Lonnie Abelbeck <lonnie@abelbeck.com> via private e-mail. Change-Id: Icc80f12b8d8d591e14a8e0ed9f1c02cbd193a89b	8 years ago
Alexei Gradinari	19da99df2f	CHANGES: correct version for a new option 'refer_blind_progress' ... Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97	8 years ago
Joshua Colp	d3e951edf5	pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. ... PJSIP support in Asterisk differs from chan_sip in that it allows media to be sent as-is without transcoding provided the codecs were negotiated in the SDP. This is allowed according to the RFC. Support for this differs quite a lot though and some endpoints do not handle it well. This change extends the 'asymmetric_rtp_codec' option to also cover this case. When set to no (the default) the code behaves as chan_sip does - the best codec is selected and we will only ever send that, unless we change what we are sending if the remote side changes. When set to yes we will send media as-is without transcoding if the codec has been negotiated in the SDP. ASTERISK-26996 Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51	8 years ago
Joshua Colp	9f054955f2	Merge "res_pjsip: Add support for returning only reachable contacts and use it."	8 years ago
Jenkins2	29f87a5530	Merge "channel: ast_write frame wrongly freed after call to audiohooks"	8 years ago
Sean Bright	b3ca24d216	res_rtp_multicast: Use consistent timestamps when possible ... When a frame destined for a MulticastRTP channel does not have timing information (such as when an 'originate' is done), we generate the RTP timestamps ourselves without regard to the number of samples we are about to send. Instead, use the same method as res_rtp_asterisk and 'predict' a timestamp given the number of samples. If the difference between the timestamp that we generate and the one we predict is within a specific threshold, use the predicted timestamp so that we end up with timestamps that are consistent with the number of samples we are actually sending. Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f	8 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	8 years ago
Jenkins2	452e6315bb	Merge "format: Reintroduce smoother flags"	8 years ago
Joshua Colp	1a24543124	Merge "Confbridge: Add "sfu" video mode to bridge profile options."	8 years ago
Jenkins2	bb2f6234da	Merge "Add primitive SFU support to bridge_softmix."	8 years ago
Joshua Colp	97abf6d475	Merge "res_srtp: Add support for libsrtp2"	8 years ago
Kevin Harwell	d8802a6a0f	channel: ast_write frame wrongly freed after call to audiohooks ... ASTERISK-26419 introduced a bug when calling ast_audiohook_write_list in ast_write. It would free the frame given to ast_write if the frame returned by ast_audiohook_write_list was different than the given one. The frame give to ast_write should never be freed within that function. It is the caller's resposibility to free the frame after writing (or when it its done with it). By freeing it within ast_write this of course led to some memory corruption problems. This patch makes it so the frame given to ast_write is no longer freed within the function. The frame returned by ast_audiohook_write_list is now subsequently used in ast_write and is freed later. It is freed either after translate if the frame returned by translate is different, or near the end of ast_write prior to function exit. ASTERISK-26973 #close Change-Id: Ic9085ba5f555eeed12f6e565a638c3649695988b	8 years ago
Joshua Colp	0d0b1730af	Merge "pbx_builtin: Properly handle hangup during Background"	8 years ago
George Joseph	b3590c9602	Merge "app_confbridge: Race between removing and playing name recording while leaving"	8 years ago
Jenkins2	1dc1af4aa5	Merge "stasis_recording: Correct ast_asprintf error checking"	8 years ago