29082 Commits (1cf2c79f37ac3dfedf27f96358b09bb45429f22d)

 

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Author	SHA1	Message	Date
Jenkins2	e26d15fabb	Merge "CFLAGS for BIND8 support" into 13	8 years ago
Jørgen H	e16a669c70	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
Jenkins2	44d4b55697	Merge "res_pjsip_mwi: don't create mwi subscriptions if initial unsolicited disabled" into 13	8 years ago
George Joseph	7dafe82751	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	e414833f6e	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	6cdf3191d3	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	0bde568669	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
Joshua Colp	bc51d4324a	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing." into 13	8 years ago
Alexei Gradinari	08be5e01e8	app_voicemail: IMAP logout on MWI unsubscribe ... Closing IMAP connection on MWI unsubscribe. ASTERISK-24052 #close Change-Id: I4ff964026002b2817b48c20fb4239f0a880228fd	8 years ago
Joshua Colp	d715beae16	Merge "codecs.conf.sample: Fix max_bandwidth speling error" into 13	8 years ago
Joshua Colp	052cd049bc	Merge "BuildSystem: Fix build on FreeBSD due to missing crypt.h" into 13	8 years ago
Jenkins2	1f7aedf6f7	Merge "eventfd: Disable during cross compilation" into 13	8 years ago
Alexei Gradinari	59c9bbe696	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
David M. Lee	68de35a6a0	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
Sean Bright	da3312457e	codecs.conf.sample: Fix max_bandwidth speling error ... Reported by Sylvain Boily via asterisk-dev mailing list. Change-Id: Idc7623f335aea3e144dd369ba383b9a757480a9d	8 years ago
Guido Falsi	6a64f65fe6	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
Jenkins2	85dff8e26e	Merge "CHANGES: correct version for a new option 'refer_blind_progress'" into 13	8 years ago
Guido Falsi	44cee2f4a1	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	1f10c6b3b0	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	590ffcaf0b	eventfd: Disable during cross compilation ... Reported by Lonnie Abelbeck <lonnie@abelbeck.com> via private e-mail. Change-Id: Icc80f12b8d8d591e14a8e0ed9f1c02cbd193a89b	8 years ago
Alexei Gradinari	5520b6c201	CHANGES: correct version for a new option 'refer_blind_progress' ... Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97	8 years ago
Joshua Colp	996a4791ff	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
Jenkins2	812f5b51cb	Merge "res_pjsip: Add support for returning only reachable contacts and use it." into 13	8 years ago
Jenkins2	f0400ed858	Merge "channel: ast_write frame wrongly freed after call to audiohooks" into 13	8 years ago
Sean Bright	c093bf8072	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	746c2c5745	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	95b2b542f6	Merge "format: Reintroduce smoother flags" into 13	8 years ago
Joshua Colp	3d7e9375fb	Merge "res_srtp: Add support for libsrtp2" into 13	8 years ago
Kevin Harwell	adfb28882b	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: I463d4ac3b736ced95de986ee74a489c7c7ab103b	8 years ago
Jenkins2	cee5b78be8	Merge "pbx_builtin: Properly handle hangup during Background" into 13	8 years ago
Jenkins2	174a9b5d99	Merge "stasis_recording: Correct ast_asprintf error checking" into 13	8 years ago
Jenkins2	1e086cf018	Merge "format_mp3: Re-work menuselect/build issues" into 13	8 years ago
Jenkins2	3e8eea0325	Merge "res_pjsip: New endpoint option "refer_blind_progress"" into 13	8 years ago
Jenkins2	a76b473c49	Merge "app_confbridge: Race between removing and playing name recording while leaving" into 13	8 years ago
Joshua Colp	b276810aa1	Merge "sip.conf.sample: Clarify where DTLS settings are permitted" into 13	8 years ago
Jenkins2	3f336ca053	Merge "test_json: Fix test names with reserved words" into 13	8 years ago
Sean Bright	283cc59af7	pbx_builtin: Properly handle hangup during Background ... Before this patch, when a user hung up during a Background, we would stuff 0xff into a char and attempt a dialplan lookup of it. This caused problems for some realtime engines which interpreted the value as the beginning of an invalid UTF-8 sequence. ASTERISK-19291 #close Reported by: Andrew Nowrot Change-Id: I8ca6da93252d61c76ebdb46a4aa65e73ca985358	8 years ago
Joshua Colp	dc05183f4b	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	8 years ago
Sean Bright	cf6cf59646	stasis_recording: Correct ast_asprintf error checking ... ASTERISK-27021 #close Reported by: Tim Morgan Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75	8 years ago
Sean Bright	70e5887906	format: Reintroduce smoother flags ... In review 4843 (ASTERISK-24858), we added a hack that forced a smoother creation when sending signed linear so that the byte order was adjusted during transmission. This was needed because smoother flags were lost during the new format work that was done in Asterisk 13. Rather than rolling that same hack into res_rtp_multicast, re-introduce smoother flags so that formats can dictate their own options. Change-Id: I77b835fba0e539c6ce50014a984766f63cab2c16	8 years ago
Sean Bright	97b003f5e2	format_mp3: Re-work menuselect/build issues ... Rather than removing format_mp3 from ALL_C_MODS (which caused format_mp3 to not show up in menuselect), use .PHONY targets when the necessary source files are not present. ASTERISK-23951 Reported by: Tzafrir Cohen Change-Id: I0a7512c51acc9e86043671795020b0de725bd9e8	8 years ago
George Joseph	c10341646d	test_json: Fix test names with reserved words ... Some of the test names were actually reserved words (true, false, int, null, string, bool). When the jenkins test results analyzer does its thing it tries to create a map using the test names as keys and fails because they're reserved words. Added "type_" to those test names. Change-Id: I90d809f46969c78a1c605b736ff0635196a2cf1b	8 years ago
Joshua Colp	1e65857e6f	Merge "format_mp3: Don't try to build format_mp3 if we don't have sources" into 13	8 years ago
Joshua Colp	b07b216235	manager: Clear the flag on the other channel. ... During the channel flag audit an incorrect change was done. The flag should be cleared on the second channel. ASTERISK-26469 Change-Id: I770c5a389550a2fb5a6ade942fccbb2e1d9199c8	8 years ago
Sean Bright	5e9cd1f20d	res_srtp: Add support for libsrtp2 ... ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40	8 years ago
Jenkins2	d4ccd3a6c0	Merge "asterisk: Audit locking of channel when manipulating flags." into 13	8 years ago
Jenkins2	5715360ba5	Merge "res_agi: Fix malformed AGI usage response" into 13	8 years ago
George Joseph	a8f8c5d857	Merge "res_agi: Allow configuration of audio format of EAGI pipe" into 13	8 years ago
George Joseph	a2d15b93f1	Merge "unittests: Add a unit test that causes a SEGV and..." into 13	8 years ago
Jenkins2	558199e5dd	Merge "res_agi: Prevent crash when SET VARIABLE called without arguments" into 13	8 years ago