5765 Commits (6edefc9480dac80895eec53f19343933b5bd531e)

Author	SHA1	Message	Date
Corey Farrell	7c3cdc81f7	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	9 years ago
Joshua Colp	1c07c80eef	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues." into 15	9 years ago
Jenkins2	fab609deeb	Merge "Core: Add support for systemd socket activation." into 15	9 years ago
Joshua Colp	451d86d62e	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	9 years ago
Jenkins2	29a746c534	Merge "core: Add VP9 passthrough support." into 15	9 years ago
Joshua Colp	927fc6bbd9	core: Add VP9 passthrough support. ... This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	9 years ago
Rusty Newton	715d79b60d	say.c: Fix file locations for second, seconds, minute, minutes files ... The seconds and minutes files have always existed in the base language directory of the Core package. So say.c has always been calling the wrong location (under digits/) for those two files and in the case of second and minute they didn't exist in the Core packages at all. The 1.6 sounds release moves the second and minute files into Core from Extra for the languages that already had them. A future release will include the second and minute files for languages that didn't already have them. This patch just changes all the target locations for second, seconds, minute, and minutes that were under the digits subdir to be under the root of sounds instead. Which is where the sounds will be for some languages after 1.6 sounds and for all languages after a future release. ASTERISK-25810 #close Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 Reported-by: Nicolas Riendeau	9 years ago
Corey Farrell	eea9da2f42	Core: Add support for systemd socket activation. ... This change adds support for socket activation of certain SOCK_STREAM listeners in Asterisk: * AMI / AMI over TLS * CLI * HTTP / HTTPS Example systemd units are provided. This support extends to any socket which is initialized using ast_tcptls_server_start, so any unknown modules using this function will support socket activation. Asterisk continues to function as normal if socket activation is not enabled or if systemd development headers are not available during build. ASTERISK-27063 #close Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d	9 years ago
Joshua Colp	f43fc91911	Merge "core: Add digit filtering to ast_waitfordigit_full"	9 years ago
Jenkins2	62c381afdb	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."	9 years ago
Jenkins2	594c7a50af	Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."	9 years ago
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	9 years ago
Corey Farrell	78a50b0343	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. ... This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	9 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	9 years ago
Corey Farrell	6b138046e7	core: Add digit filtering to ast_waitfordigit_full ... This adds a parameter to ast_waitfordigit_full which can be used to only stop waiting when certain expected digits are received. Any unexpected DTMF digits are simply ignored. This also creates a new dialplan application WaitDigit. ASTERISK-27129 #close Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9	9 years ago
Joshua Colp	95b35cb1cb	Merge "Avoid setting maxfiles for a remote asterisk"	9 years ago
Jenkins2	fbcfa6b4b2	Merge "http.c: Reduce log spam"	9 years ago
Joshua Colp	7f09fd2c2f	bridge/core_unreal: Fix SFU bugs with forwarding frames. ... This change fixes a few things uncovered during SFU testing. 1. Unreal channels incorrectly forwarded video frames when no video stream was present on them. This caused a crash when they were read as the core requires a stream to exist for the underlying media type. The Unreal channel will now ensure a stream exists for the media type before forwarding the frame and if no stream exists then the frame is dropped. 2. Mapping of frames during bridging from the stream number of the underlying channel to the stream number of the bridge was done in the wrong location. This resulted in the frame getting dropped. This mapping now occurs on reading of the frame from the channel. 3. Bridging was using the wrong ast_read function resulting in it living in a non-multistream world. 4. In bridge_softmix when adding new streams to existing channels the wrong stream topology was copied resulting in no streams being added. Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8	9 years ago
Tzafrir Cohen	d58ef31acd	Avoid setting maxfiles for a remote asterisk ... Setting maxfiles (maximum number of open files) has no practical effect on a remote asterisk (rasterisk, rasterisk -x). It has an ill effect of printing an extra message, which may be annoying in case of -x. ASTERISK-27105 #close Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2	9 years ago
George Joseph	303f935a50	http.c: Reduce log spam ... Messages like "fwrite() failed: Connection reset by peer" are no help whatsoever, especially since they can be caused simply by a client disconnecting. * Make those WARNINGs DEBUGs. * Check the return from ast_iostream_printf of headers. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b	9 years ago
Richard Mudgett	03ae8b0105	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	9 years ago
Jenkins2	d6c08cc559	Merge "core: Remove 'Data Retrieval API'"	9 years ago
Sean Bright	325eeced6a	core: Remove 'Data Retrieval API' ... This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83	9 years ago
Corey Farrell	50ddb56dad	channel: Clear channel flag in error branch. ... Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when ast_read returns NULL. ASTERISK-27100 #close Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d	9 years ago
Mark Michelson	45df25a579	chan_pjsip: Add support for multiple streams of the same type. ... The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7	9 years ago
Jenkins2	01536546e2	Merge "bridge: stuck channel(s) after failed attended transfer"	9 years ago
Kevin Harwell	27dae55fb6	core_local: local channel data not being properly unref'ed and unlocked ... In an earlier version of Asterisk a local channel [un]lock all functions were added in order to keep a crash from occurring when a channel hung up too early during an attended transfer. Unfortunately, when a transfer failure occurs and depending on the timing, the local channels sometime do not get properly unlocked and deref'ed after being locked and ref'ed. This happens because the underlying local channel structure gets NULLed out before unlocking. This patch reworks those [un]lock functions and makes sure the values that get locked and ref'ed later get unlocked and deref'ed. ASTERISK-27074 #close Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09	9 years ago
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