29493 Commits (5d3420a2deb11773062ddff47473669dfcbde10a)

 

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Author	SHA1	Message	Date
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
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
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	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
Joshua Colp	8d70c081a4	Merge "format_mp3: Re-work menuselect/build issues"	8 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	8 years ago
Jenkins2	77f7c1b98f	Merge "Sqlite3: make busy_timeout configurable."	8 years ago
Jenkins2	f166074524	Merge "test_json: Fix test names with reserved words"	8 years ago
Sean Bright	001f4ddda4	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	f6eeaaafd5	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	8 years ago
Sean Bright	9dce4a947b	stasis_recording: Correct ast_asprintf error checking ... ASTERISK-27021 #close Reported by: Tim Morgan Change-Id: I0ac061f040093e806c3b1f4e2340864f3ce4dd75	8 years ago
Sean Bright	5c27fe2187	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
Mark Michelson	39d14834f8	Confbridge: Add "sfu" video mode to bridge profile options. ... A previous commit added plumbing to bridge_softmix to allow for an SFU experience with Asterisk. This commit adds an option to app_confbridge that allows for a confbridge to actually make use of the SFU video mode. SFU mode is implemented in a "set it and forget it" kind of way. That is, when the bridge is created, if SFU mode is enabled, then the video mode gets set to SFU and cannot be changed. Future improvements may allow for a hybrid experience (e.g. forward multiple video streams, specifically those of the most recent talkers), but for this addition, no such capability is present. Change-Id: I87bbcb63dec6dbbb42488f894871b86f112b2020	8 years ago
Mark Michelson	2da869408a	Add primitive SFU support to bridge_softmix. ... This sets up the "plumbing" in bridge_softmix to be able to accommodate Asterisk asking as an SFU (selective forwarding unit) for conferences. The way this works is that whenever a channel enters or leaves a conference, all participants in the bridge get sent a stream topology change request. The topologies consist of the channels' original topology, along with video destination streams corresponding to each participants' source video streams. So for instance, if Alice, Bob, and Carol are in the conference, and each supplies one video stream, then the topologies for each would look like so: Alice: Audio, Source video(Alice), Destination Video(Bob), Destination video (Carol) Bob: Audio, Source video(Bob) Destination Video(Alice), Destination video (Carol) Carol: Audio, Source video(Carol) Destination Video(Alice), Destination video (Bob) This way, video that arrives from a source video stream can then be copied out to the destination video streams on the other participants' channels. Once the bridge gets told that a topology on a channel has changed, the bridge constructs a map in order to get the video frames routed to the proper destination streams. This is done using the bridge channel's stream_map. This change is bare-bones with regards to SFU support. Some key features are missing at this point: * Stream limits. This commit makes no effort to limit the number of streams on a specific channel. This means that if there were 50 video callers in a conference, bridge_softmix will happily send out topology change requests to every channel in the bridge, requesting 50+ streams. * Configuration. The plumbing has been added to bridge_softmix, but there has been nothing added as of yet to app_confbridge to enable SFU video mode. * Testing. Some functions included here have unit tests. However, the functionality as a whole has only been verified by hand-tracing the code. * Selectivenss. For a "selective" forwarding unit, this does not currently have any means of being selective. * Features. Presumably, someone might wish to only receive video from specific sources. There are no external-facing functions at the moment that allow for users to select who they receive video from. * Efficiency. The current scheme treats all video streams as being unidirectional. We could be re-using a source video stream as a desetnation, too. But to simplify things on this first round, I did it this way. Change-Id: I7c44a829cc63acf8b596a337b2dc3c13898a6c4d	8 years ago
Sean Bright	045d7b8cb7	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	80206cdc65	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
Jenkins2	cf60a0c5bd	Merge "format_mp3: Don't try to build format_mp3 if we don't have sources"	8 years ago
Joshua Colp	9c4f63263c	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	9 years ago
Sean Bright	1f136fe885	res_srtp: Add support for libsrtp2 ... ASTERISK-25294 #close Reported by: Tzafrir Cohen ASTERISK-26976 #close Reported by: Alex Change-Id: I789b1c3d1ed31365bbd9339fa58ef36f48833c40	9 years ago
Jenkins2	56b6a71548	Merge "asterisk: Audit locking of channel when manipulating flags."	9 years ago
George Joseph	6bb3cedb1b	Merge "res_agi: Prevent crash when SET VARIABLE called without arguments"	9 years ago
George Joseph	366b10a500	Merge "res_agi: Allow configuration of audio format of EAGI pipe"	9 years ago
Jenkins2	915acf1e5d	Merge "res_agi: Fix malformed AGI usage response"	9 years ago
Jenkins2	24318aac45	Merge "unittests: Add a unit test that causes a SEGV and..."	9 years ago
Sean Bright	59348aa182	format_mp3: Don't try to build format_mp3 if we don't have sources ... ASTERISK-23951 #close Reported by: Tzafrir Cohen Change-Id: Iebf181d44bb735787fde4b5be863c4d7e2478a30	9 years ago
Martin Tomec	44c5a144ce	Sqlite3: make busy_timeout configurable. ... Enables runtime configuration of busy_timeout for sqlite databases. Default timeout remains 1000ms. ASTERISK-27014 #close Change-Id: I8921a3aac3c335843be4cb17d2dd0a5c157a36da	9 years ago
Jenkins2	d6992daaef	Merge "res_agi: Clarify 'RECORD FILE' documentation"	9 years ago
George Joseph	08edd54c1b	unittests: Add a unit test that causes a SEGV and... ... ...that can only be run by explicitly calling it with 'test execute category /DO_NOT_RUN/ name RAISE_SEGV' This allows us to more easily test CI and debugging tools that should do certain things when asterisk coredumps. To allow this a new member was added to the ast_test_info structure named 'explicit_only'. If set by a test, the test will be skipped during a 'test execute all' or 'test execute category ...'. Change-Id: Ia3a11856aae4887df9a02b6b081cc777b36eb6ed	9 years ago
Joshua Colp	2e7866ebb7	Merge "app_queue: Add QUEUE_RAISE_PENALTY feature"	9 years ago
Jenkins2	c247625012	Merge "chan_sip: Better ICE handling for RTCP-MUX"	9 years ago
Jenkins2	cd0e6a2324	Merge "res_rtp_asterisk: rtcp mux using the wrong srtp unprotecting algorithm"	9 years ago