asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	404596b790	channel: Fix issues in hangup scenarios caused by frame deferral ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8	9 years ago
Joshua Colp	19ca0e7fc8	Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded." into 13	9 years ago
zuul	b745c326c2	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source" into 13	9 years ago
Mark Michelson	2c031b67d3	res_format_attr_opus: Fix fmtp generation. res_format_attr_opus assumed that the string being passed into it was empty. It tried to determine if the only thing it had written was a=fmtp:<num> And if it had, it would reset the string. Its calculation was off when working with chan_sip, though. chan_sip passes the entire built SDP rather than an empty string. This resulted in always putting an empty fmtp line in the SDP. ASTERISK-26520 #close Reported by scgm11 Change-Id: Ib2e8712d26a47067e5f36d5973577added01dbb5	9 years ago
Joshua Colp	4c1b9fbefe	Merge "Revert "Revert "channel: Use frame deferral API for safe sleep.""" into 13	9 years ago
Joshua Colp	d0a72d3779	Merge "Revert "Revert "autoservice: Use frame deferral API""" into 13	9 years ago
zuul	36b59ee9c1	Merge "Revert "Revert "AGI: Only defer frames when in an interception routine.""" into 13	9 years ago
zuul	f68790d46a	Merge "Revert "Revert "Add API for channel frame deferral.""" into 13	9 years ago
zuul	f4f152f35c	Merge "apps/app_echo: Only relay a single video source change frame" into 13	9 years ago
Richard Mudgett	ed0f1afc8c	codec_opus: Fix warning when Opus negotiated but codec_opus not loaded. When Opus is negotiated but not loaded, the log is spammed with messages because the system does not know how to calculate the number of samples in a frame. * Suppress the warning by supplying a function that assumes 20ms of samples in the frame. For pass through support it doesn't really seem to matter what number of samples is returned anyway. ASTERISK-26605 #close Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f	9 years ago
Joshua Colp	4c3d25875f	Merge "Add X.509 subject alternative name support to TLS certificate verification." into 13	9 years ago
Joshua Colp	6672fd5faa	Merge "cli: Fix ast_el_read_char to work with libedit >= 3.1" into 13	9 years ago
Richard Mudgett	e632222bc4	res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak. Responding to authentication challenges leaks PJSIP memory pools. The leak was introduced with a pjproject 2.5.5 API change. https://trac.pjsip.org/repos/ticket/1929 changed the API usage of pjsip_auth_clt_init() to require the new API pjsip_auth_clt_deinit() to clean up cached authentication allocations that get allocated with pjsip_auth_clt_reinit_req(). ASTERISK-26516 #close Change-Id: I4473141b8c3961d0dc91c382beb3876b3efb45c8	9 years ago
Joshua Colp	c4baa50f9a	Merge "file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type" into 13	9 years ago
Alexei Gradinari	cf6d13180e	chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d	9 years ago
George Joseph	c92dcc76da	file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type One of the code paths in __ast_file_read_dirs will only get executed if the OS doesn't support dirent->d_type OR if the filesystem the particular file is on doesn't support it. So, while standard Linux systems support the field, some filesystems like XFS do not. In this case, we need to call stat() to determine whether the directory entry is a file or directory so we append the filename to the supplied directory path and call stat. We forgot to truncate path back to just the directory afterwards though so we were passing a complete file name to the callback in the dir_name parameter instead of just the directory name. The logic has been re-written to only create a full_path if we need to call stat() or if we need to descend into another directory. Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba	9 years ago
Maciej Szmigiero	7b96e8cc3d	Add X.509 subject alternative name support to TLS certificate verification. This way one X.509 certificate can be used for hosts that can be reached under multiple DNS names or for multiple hosts. Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name> ASTERISK-25063 #close Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f	9 years ago
Matt Jordan	0790aa528a	pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS The PJ_ICE_MAX_CHECKS constant is used by pjproject to determine how many pairs of local/remote candidates will be made. If for some reason we reach this upper bound, ICE will generally fail and no media will flow between the browser and Asterisk. This patch makes PJ_ICE_MAX_CHECKS set to the total possible number of pairs of candidates we'd theoretically allow, which is PJ_ICE_MAX_CAND^2. Prior to this patch, we simply multiplied PJ_ICE_MAX_CAND by two; on systems with multiple interfaces (I blame Docker), this is far too low to allow WebRTC calls to succeed. Setting this to be PJ_ICE_MAX_CAND^2 allowed WebRTC calls to succeed even when the system Asterisk was running on had quite a few virtual interfaces. Change-Id: Icd4f17de0ac9d3a83dddfc8bf1cb7616bc107d55	9 years ago
Matt Jordan	993a6f96c7	apps/app_echo: Only relay a single video source change frame In `9785e8d0`, app_echo was updated to relay video source updates to the channel for the purposes of displaying video in WebRTC tests. Unfortunately, this can cause a Kafkaesque nightmare if two or more Local channels are in a bridge together where their ends are in app_echo. When this situation occurs, a video update sent into app_echo will cause the video update to be relayed to the other Local channels, causing another round of video updates, etc. In not much time at all, the channel length queues will be overwhelmed, channel alert pipes will fail, and all hell will break loose as Asterisk merrily continues to throw more video update requests onto the channels. This patch updates app_echo to only relay a single video update. Once a video update has been made, all further video updates are dropped. This meets the intended purpose of the original patch: if we get a video update and we're in app_echo, go ahead and ask the sender to update themselves. However, once we've got that video stream sync'd up, don't keep spamming the world. Change-Id: I9210780b08d4c17ddb38599d1c64453adfc34f74	9 years ago
Matt Jordan	d23b4af477	res/ari/resource_bridges: Add the ability to manipulate the video source In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	9 years ago
George Joseph	404a62eeee	Revert "Revert "channel: Use frame deferral API for safe sleep."" This reverts commit `58c88cfbaa`. Change-Id: I72692e2b2e83ef6da9390075ff20b138b2c374b6	9 years ago
George Joseph	09d8febc91	Revert "Revert "autoservice: Use frame deferral API"" This reverts commit `1df434e2b4`. Change-Id: Id2b8a8bccbb4bbdd82b792275d4cd6f32563e401	9 years ago
George Joseph	ffad2b44df	Revert "Revert "AGI: Only defer frames when in an interception routine."" This reverts commit `6be5d8de0d`. Change-Id: I4b548137f52ae0686d8f09e21496b778d1c6a797	9 years ago
George Joseph	2fefb6187f	Revert "Revert "Add API for channel frame deferral."" This reverts commit `6b5a7ced13`. Change-Id: I61d1dbb2e69e1977f684b7dfc8e98211024e1cd1	9 years ago
zuul	a58d359701	Merge "res_pjsip.c: Rework endpt_send_request() req_wrapper code." into 13	9 years ago
George Joseph	5e0c224043	cli: Fix ast_el_read_char to work with libedit >= 3.1 Libedit 3.1 is not build with unicode on as a default and so the prototype for the el_gets callback changed from expecting a char buffer to accepting a wchar buffer. If ast_el_read_char isn't changed, the cli reads garbage from teh terminal. Added a configure test for (el_rfunc_t)(EditLine , wchar_t *) and updated ast_el_read_char to use the HAVE_ define to detemrine whether to use char or wchar. ASTERISK-26592 #close Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a	9 years ago
Joshua Colp	ec17e96784	Merge "Fix closing rtp ports after call finished in chan_unistim." into 13	9 years ago
Joshua Colp	d5774005fe	Merge "res_pjsip: Fix tdata leaks in off nominal paths." into 13	9 years ago
Igor Goncharovskiy	3faca1d4ff	Fix closing rtp ports after call finished in chan_unistim. Fix ASTERISK-26565 by adding ast_rtp_instance_stop before rtp instance destroy for chan_unistim. Also several fixes for displayed text translation. Change-Id: If42a03eea09bd1633471406bdc829cf98bf6affc	9 years ago
Joshua Colp	10e64a5fb8	Merge "res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp." into 13	9 years ago
zuul	cf08676c00	Merge "build: Fix default values for some SANITIZER options" into 13	9 years ago
Richard Mudgett	412d43fa21	res_pjsip.c: Rework endpt_send_request() req_wrapper code. * Don't hold the req_wrapper lock too long in endpt_send_request(). We could block the PJSIP monitor thread if the timeout timer expires. sip_get_tpselector_from_endpoint() does a sorcery access that could take awhile accessing a database. pjsip_endpt_send_request() might take awhile if selecting a transport. * Shorten the time that the req_wrapper lock is held in the callback functions. * Simplify endpt_send_request() req_wrapper->timeout code. * Removed some redundant req_wrapper->timeout_timer->id assignments. Change-Id: I3195e3a8e0207bb8e7f49060ad2742cf21a6e4c9	9 years ago
Richard Mudgett	2e7fc56d3c	res_pjsip: Fix tdata leaks in off nominal paths. Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b	9 years ago
Richard Mudgett	da68b185b3	res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94	9 years ago
George Joseph	ee73af1d88	Update for 13.12.2	9 years ago
Joshua Colp	b70eb07c53	res_pjsip_sdp_rtp: Reject offer of required SRTP without res_srtp. When optimistic SRTP was on it was possible for us to still set up a call without an audio stream if an offer was received with required SRTP. This change makes it so this scenario will now fail with a 488 response. ASTERISK-26575 Change-Id: I7d14187037681f48879bd20319ac79d0877318f3	9 years ago
Joshua Colp	71dc333565	app_queue: Add mention of 'ABANDON' variable to CHANGES. ASTERISK-26558 Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e	9 years ago
George Joseph	ccc8a37584	Merge "Revert "autoservice: Use frame deferral API"" into 13	9 years ago
George Joseph	07e7ac5119	Merge "Revert "Add API for channel frame deferral."" into 13	9 years ago
George Joseph	06045fc29e	Merge "Revert "AGI: Only defer frames when in an interception routine."" into 13	9 years ago
George Joseph	9e13c3e2a1	Merge "Revert "channel: Use frame deferral API for safe sleep."" into 13	9 years ago
George Joseph	6b5a7ced13	Revert "Add API for channel frame deferral." This reverts commit `9231a56cf3`. Multiple testsuite failures were detected after the fact. Change-Id: I3bac8d7c3ddb69a4ddf6c5d6de0ffa5ff7ff3af7	9 years ago
George Joseph	6be5d8de0d	Revert "AGI: Only defer frames when in an interception routine." This reverts commit `5c10091f3d`. Multiple testsuite failures were detected after the fact. Change-Id: I397a841acc17ae230c512449cd6bed89d2ef3b73	9 years ago
George Joseph	1df434e2b4	Revert "autoservice: Use frame deferral API" This reverts commit `2e3a354575`. Multiple testsuite failures were detected after the fact. Change-Id: Ia45fa4633fae74dca345b24bb6722737c63035de	9 years ago
George Joseph	58c88cfbaa	Revert "channel: Use frame deferral API for safe sleep." This reverts commit `44f7e25239`. Multiple testsuite failures were detected after the fact. Change-Id: I56299087da22128a95f0c8f3955f740890d7ca65	9 years ago
Joshua Colp	44fd077739	Merge "app_queue: new variable set when abandoned" into 13	9 years ago
George Joseph	a562fbe618	build: Fix default values for some SANITIZER options 2 of the sanitizers didn't have default values so in systems that don't support sanitizers menuselect would spit out warnings. They were harmless but confusing. They've now been set to "0". Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58	9 years ago
George Joseph	dfcb2b6c24	Merge "res_pjsip_session: Do not call session supplements when it's too late." into 13	9 years ago
Sebastian Gutierrez	7fd5031c1c	app_queue: new variable set when abandoned sets the variable ABANDONED to TRUE if the call was not answered. ASTERISK-26558 Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3	9 years ago
Mark Michelson	e043d1a55c	res_pjsip_session: Do not call session supplements when it's too late. res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92	9 years ago

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