5738 Commits (45a1f4e2ae3749d16820d7f9ea1e7dfbe028bbdf)

Author	SHA1	Message	Date
Kevin Harwell	45a1f4e2ae	bridge: stuck channel(s) after failed attended transfer ... If an attended transfer failed it was possible for some of the channels involved to get "stuck" because Asterisk was not hanging up the transfer target. This patch ensures Asterisk hangs up the transfer target when an attended transfer failure occurs. ASTERISK-27075 #close Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9	9 years ago
Joshua Colp	57bbba7d43	Merge "res_stasis: Plug reference leak on stolen channels"	9 years ago
George Joseph	3f5bf287a2	Merge "SDP: Add get/set option calls for RTP sched context per type."	9 years ago
George Joseph	854a6de819	res_stasis: Plug reference leak on stolen channels ... When a stasis channel is stolen by another app, the control structure is unreffed but never unlinked from the app_controls container. This causes the channel reference to leak. Added OBJ_UNLINK to the callback in channel_stolen_cb. Also added some additional channel lifecycle debug messages to channel.c. ASTERISK-27059 #close Repoorted-by: George Joseph Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14	9 years ago
Frederic LE FOLL	0ad95bc8a0	Core/PBX: Deadlock between dialplan execution and application unregistration. ... Not easy to reproduce, but we have noticed deadlocks when unloading a module while dialplan is handling a request. The deadlock is between : 1) Dialplan execution: pbx_extension_helper() first taking conlock, then pbx_findapp() [when called] asking for lock on apps list. 2) Application unregistration: ast_unregister_application() first taking lock on apps list, then unreference_cached_app() [when called] asking for conlock. As a protection, I suggest to modify ast_unregister_application(), so that it anticipates the need of conlock, before taking the lock on apps list. The side effect is a longer unavailability of conlock when unregistering an application. ASTERISK-27041 Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2	9 years ago
Joshua Colp	0405185357	Merge "SDP: Search for the ice-lite attribute in the right place."	9 years ago
Jenkins2	d81293a5dd	Merge changes from topic 'sdp_api_adjustments' ... * changes: SDP: Set the remote c= line in RTP instance. SDP: Add t= line in sdp_create_from_state() stream: Ignore declined streams for some topology calls.	9 years ago
Jenkins2	2f684eb6a5	Merge "stream: Add ast_stream_topology_del_stream() and unit test."	9 years ago
Joshua Colp	41bd01c861	Merge "channel: Fix reference counting in ast_channel_suppress."	9 years ago
Richard Mudgett	e563a1920e	SDP: Add get/set option calls for RTP sched context per type. ... Change-Id: I82dc75c63c48904e9e5a49e2205dcc06e88487e4	9 years ago
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	9 years ago
Richard Mudgett	a95584d079	SDP: Set the remote c= line in RTP instance. ... Change-Id: I23b646392082deab65bedeb19b12dcbcb9216d0c	9 years ago
Richard Mudgett	06265b8c8a	stream: Add ast_stream_topology_del_stream() and unit test. ... Change-Id: If07e3c716a2e3ff85ae905c17572ea6ec3cdc1f9	9 years ago
Richard Mudgett	0fdb99c268	SDP: Add t= line in sdp_create_from_state() ... Change-Id: I4060391328a893101ed87d0d9bacbbab4fd8b141	9 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	9 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	9 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	9 years ago
Jenkins2	5d3420a2de	Merge "BuildSystem: Add patches to allow building with recent LibreSSL"	9 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	9 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	9 years ago
Jenkins2	29f87a5530	Merge "channel: ast_write frame wrongly freed after call to audiohooks"	9 years ago
Jenkins2	452e6315bb	Merge "format: Reintroduce smoother flags"	9 years ago
Joshua Colp	1a24543124	Merge "Confbridge: Add "sfu" video mode to bridge profile options."	9 years ago
Jenkins2	bb2f6234da	Merge "Add primitive SFU support to bridge_softmix."	9 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	9 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	9 years ago
Joshua Colp	f6eeaaafd5	channel / app_meetme: Fix parentheses. ... ASTERISK-27025 Change-Id: Id736b0aa4ec6b6b0f04663d64fa8d151f81fdbed	9 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	9 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	9 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	9 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
Jenkins2	56b6a71548	Merge "asterisk: Audit locking of channel when manipulating flags."	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
Kevin Harwell	51375686f7	core/conversions: Added string to unsigned integer and long conversions ... Added functions that convert a string to an unsigned integer or unsigned long. A couple of unit test were also created to test the routines. The reasons for adding these conversion utilities (and hopefully eventually more) are as follows: * Conversion routines are functionally contained with consistent and better error checking * The function names offer a better description of what is happening * It encourages code reuse for easier bug fixing at a single source * It's simpler to use * It's unit testable For instance, currently in a lot of places when converting to an integer or similar the "sscanf" function is used. When using "sscanf" it may not be immediately clear what's happening as it lacks semantic naming. Limited error checking is usually done as well. For example, most of the time a check is done to make sure the value converted, but does not check for overflows or negative valued conversions when converting unsigned numbers. Why use/wrap "strtoul" and not "sscanf" then? Primarily, it lacks some of the built in error handling that "strtoul" has. For instance "strtoul" contains overflow checks. Less so, but can still factor as reasons, "sscanf" is slightly more complex in its use. And maybe a bit controversial, but it may be ("big if") potentially slower than "strtoul" in some cases. Change-Id: If7eaca4a48f8c7b89cc8b5a1f4bed2852fca82bb	9 years ago
Joshua Colp	5a7af00e80	asterisk: Audit locking of channel when manipulating flags. ... When manipulating flags on a channel the channel has to be locked to guarantee that nothing else is also manipulating the flags. This change introduces locking where necessary to guarantee this. It also adds helper functions that manipulate channel flags and lock to reduce repeated code. ASTERISK-26789 Change-Id: I489280662dba0f4c50981bfc5b5a7073fef2db10	9 years ago
George Joseph	ce4d8dac91	Merge changes from topic 'sdp_api_adjustments' ... * changes: SDP: Make process possible multiple fmtp attributes per rtpmap. SDP: Explicitly stop a RTP instance before destoying it. SDP: Rework merge_capabilities(). SDP: Update ast_get_topology_from_sdp() to keep RTP map.	9 years ago
George Joseph	28d4e6be9b	Merge "SDP: Remove sdp_state.remote_capabilities"	9 years ago
Jenkins2	f09e079294	Merge "SDP: Add interface_address to specify our address to use."	9 years ago
Jenkins2	542dd7d795	Merge "logger: Added logger_queue_limit to the configuration options."	9 years ago
Jenkins2	8b15719a11	Merge "tcptls: Improve error messages for TLS connections."	9 years ago
Richard Mudgett	b8659be9b0	SDP: Make process possible multiple fmtp attributes per rtpmap. ... Change-Id: Ie7511008d82b59590e0eb520a21b5e1da4bd7349	9 years ago
Richard Mudgett	c2906dfa05	SDP: Remove sdp_state.remote_capabilities ... The sdp_state.remote_capabilities was only used inside merge_sdps() and subsequent calls to merge_sdps() by re-INVITE's would leak them. Change-Id: I0ceb7838ea044cc913e8ad4a255c39c9740ae0ce	9 years ago
Richard Mudgett	16785c0908	SDP: Add interface_address to specify our address to use. ... When we optionally set the interface_address we are forcing the media to go out a specific interface address. This allows us to optionally have the media go out the interface that SIP signalling came in on or if we are configured to have the media always go out a specific address. Change-Id: I160d9fac322a075bd2557b430632544178196189	9 years ago
Richard Mudgett	367042bd3e	SDP: Explicitly stop a RTP instance before destoying it. ... * Made sdp_add_m_from_rtp_stream() and sdp_add_m_from_udptl_stream() handle generating disabled/declined streams. * Added /main/sdp/sdp_merge_asymmetric unit test. It currently does not check the offerer side negotiated SDP because that isn't the purpose of this patch and there is much to be done to handle declined/dummy streams. * Added T.38 image streams to the /main/sdp/sdp_merge_symmetric and /main/sdp/sdp_merge_crisscross unit tests. Change-Id: Ib4dcb3ca4f9a9133b376f4e3302f9a1f963f2b31	9 years ago
Richard Mudgett	be5809fac8	SDP: Rework merge_capabilities(). ... * Tried to give better variable names. * Made our SDP answer use the offer's RTP payload types as the SDP RFC says we SHOULD. * Updating the local topology now takes the stream format caps. We are likely preparing to send an offer. Change-Id: I34d3be8e3036402a8575ffcae3eebc5ce348d7c0	9 years ago
Richard Mudgett	ae7689f093	SDP: Update ast_get_topology_from_sdp() to keep RTP map. ... * Add failure exits to ast_get_topology_from_sdp(). Change-Id: I4cc85c1ede8d712766ed20f544dbcef04c8c1049	9 years ago
Joshua Colp	cbbd119c21	tcptls: Improve error messages for TLS connections. ... This change uses the functions provided by OpenSSL to query and better construct error messages for situations where the connection encounters a problem. ASTERISK-26606 Change-Id: I7ae40ce88c0dc4e185c4df1ceb3a6ccc198f075b	9 years ago
Joshua Elson	10a4439ac9	Prevent Undefined Capath Crash ... It is possible to initialize a valid config without a capath or cafile definition. This will cause a crash on a reload. This fix ensures capath is always allocated. ASTERISK-26983 #close Change-Id: I63ff715d9d9023427543a5b8a4ba7b0d82533c12	9 years ago
Joshua Colp	c62b5721b3	Merge "stream: ast_stream_clone() cannot copy the opaque user data."	9 years ago
George Joseph	201346fb7d	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago