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Author	SHA1	Message	Date
George Joseph	2a40c3a867	pjproject_bundled: Improve reliability of pjproject download ... The download process now has a timeout which will cause wget to retry if it stops retrieving data for 5 seconds and fetch and curl to timeout if the whole retrieval take smore than 30 seconds. If the tarball retrieval works, the MD5SUM file is retrieved from the downloads site and the md5 checksum is verified. If either the tarball retrieval or MD5SUM retrieval fails, or the checksums don't match, the entire process is retried once. If it fails again, any incomplete tarball is deleted. .DELETE_ON_ERROR: was also added to the Makefile. Not only does this delete the tarball on failure, it till also delete corrupted library files from the pjproject source directory should they fail to build correctly. Tested all the way back to FreeBSD 9, CentOS 6, Debian 6 and Ubuntu 14. Change-Id: Iea7d33b96a31622ab1b6e54baebaf271959514e1	9 years ago
misha	12c4e664bc	main/app.c: Transmit Silence on ControlPlayback pause ... ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8	9 years ago
Mark Michelson	bde3d022a3	manager: update minor version ... Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: I02586bd6cafc0baa33ea98c2f75356c0f5e03435	9 years ago
zuul	3135a745e3	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak." into 13	9 years ago
George Joseph	d3f921ed51	Merge "res_format_attr_opus: Fix fmtp generation." into 13	9 years ago
George Joseph	b213045fe4	build: Various OpenBSD issues ... OpenBSD's 'find' doesn't take the -delete argument so you have to pipe through 'xargs rm -rf'. 'echo -e' doesn't like \t starting a line. It just prints 't' which causes the libasteriskpj.exports file to be garbage. They were just cosmetic so they were removed. librt doesn't exist so the link of libasteriskpj.so fails. It's not actually needed for linux anyway so -lrt was removed from the link. res_rtp_asterisk was failing to load because of an undefined DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if so DTLSv1_method is used instead. ASTERISK-26608 Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c	9 years ago
George Joseph	c877801e5d	Merge "channel: Fix issues in hangup scenarios caused by frame deferral" into 13	9 years ago
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