asterisk

Commit Graph

Author	SHA1	Message	Date
Matt Jordan	0e15760795	res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter Per RFC 7118 5.2, the SIP URI 'transport' parameter should advertise 'ws' when WebSockets are to be used as the transport. This applies to both secure and insecure WebSockets. There were two bugs in Asterisk with respect to this: (1) The most egregious occurs in res_pjsip. There, we advertise 'ws' for insecure websockets and 'wss' for secure websockets. While this would seem to make sense - since 'WS' and 'WSS' are used for the Via Transport parameter - this is not the case for the SIP URI. This patch corrects that by registering the secure websockets with pjproject using the shorthand 'WS', and by returning 'ws' when asked for the transport parameter. Note that in pjproject, it is perfectly valid to have multiple transports use the same shorthand. (2) In chan_sip, we return an upper-case version of the transport 'WS' instead of 'ws'. Since we should be strict in what we send and liberal in what we accept (within reason), this patch lower-cases the transport before appending it to the parameter. ASTERISK-24330 #close Reported by: cervajs, Inaki Baz Castillo Change-Id: Iff77b645f8cc3b7cd35168a6676c26b147f22f42	9 years ago
Joshua Colp	606ab90c25	Merge "autoconf: more variants for OSARCH linux-gnu"	9 years ago
Joshua Colp	acdae08cf7	Merge "codec_dahdi: Fix poll.h include."	9 years ago
Joshua Colp	c9cc64b911	Merge "ast_format: Adds an identifier for interleaved audio formats to the ast_format"	9 years ago
Joshua Colp	e3dae763ee	iostream: Move include of asterisk.h The asterisk.h header file needs to be included first or else some things go awry, such as: implicit declaration of function 'vasprintf' Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c	9 years ago
gestoip2	d9b24cce0a	res_rtp_asterisk: RTT miscalculation in RTCP When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f	9 years ago
Joshua Colp	038158bf7b	Merge "addons/chan_mobile: do not use strerror_r"	9 years ago
Joshua Colp	b1f7cc4223	Merge "Add support for older name resolving version libraries like openBSD"	9 years ago
George Joseph	abae3dc36e	pjproject_bundled: Use $(LIB_RT) for link of libasteriskpj libasteriskpj was hard coded to use -lrt but librt is linux specific so we now use the LIB_RT variable which gets set by configure. Change-Id: I41148884517e3031f7675a413d524c86e8614694	9 years ago
zuul	aa9d91c290	Merge "pjproject_bundled: Improve reliability of pjproject download"	9 years ago
Joshua Colp	84e508c999	Merge "main/app.c: Transmit Silence on ControlPlayback pause"	9 years ago
zuul	120a4999f0	Merge "Add support for building RADIUS with radcli"	9 years ago
snuffy	b546497fe0	Add support for older name resolving version libraries like openBSD Fix support of OS's like openBSD that use an older nameser.h, this change reverts the defines to the older style which on other systems is found in nameser_compat.h Tested on openBSD 6.0, Debian 8 ASTERISK-26608 #close Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a	9 years ago
Mark Michelson	7a8d6bc81b	Bump ARI version to 2.0.0 In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665	9 years ago
zuul	782985631e	Merge "build: Various OpenBSD issues"	9 years ago
George Joseph	d3f070c7a2	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
Joshua Colp	98b3b500dc	Merge "manager: update minor version"	9 years ago
misha	e822a50f86	main/app.c: Transmit Silence on ControlPlayback pause ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8	9 years ago
Joshua Colp	d3dba74017	Merge "Implement internal abstraction for iostreams"	9 years ago
Mark Michelson	d670ea6297	manager: update minor version Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: Idf84507354170400813cda780906c94c9f1b60b4	9 years ago
Timo Teräs	349e08cb48	codec_dahdi: Fix poll.h include. POSIX defines poll.h. sys/poll.h should not be used as it is c-library internal header which may or may not exist. Notably in musl including sys/poll.h generates warning of being incorrect. Change-Id: Ib318c1c7142a737bcf3caa4d8d72560bebe39252	9 years ago
Joshua Colp	ed143a3b7c	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak."	9 years ago
Joshua Colp	09d1958448	Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded."	9 years ago
zuul	8fbdedb36e	Merge "res_format_attr_opus: Fix fmtp generation."	9 years ago
George Joseph	935f5d003b	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	97b2ba472d	Merge "channel: Fix issues in hangup scenarios caused by frame deferral"	9 years ago
George Joseph	b8e91bb9cc	Merge "Revert "Revert "channel: Use frame deferral API for safe sleep."""	9 years ago
George Joseph	99e97154bb	Merge "Revert "Revert "autoservice: Use frame deferral API"""	9 years ago
George Joseph	013e7dd4a6	Merge "Revert "Revert "AGI: Only defer frames when in an interception routine."""	9 years ago
George Joseph	ac0a1ee6da	Merge "Revert "Revert "Add API for channel frame deferral."""	9 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	9 years ago
Mark Michelson	dc8f99ee27	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	732ab6a045	Merge "apps/app_echo: Only relay a single video source change frame"	9 years ago
George Joseph	89e79a487a	Merge "file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type"	9 years ago
Richard Mudgett	ed9ced0531	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	1c26117dff	Merge "cli: Fix ast_el_read_char to work with libedit >= 3.1"	9 years ago
Richard Mudgett	0cd0e70c16	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	24d35ff74a	Merge "pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS"	9 years ago
George Joseph	3017f09f22	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
Joshua Colp	6911f9891f	Merge "manager: Bump AMI version number."	9 years ago
Joshua Colp	ceefe483cf	Merge "res_ari: Add support for channel variables in ARI events."	9 years ago
Timo Teräs	070a51bf7c	Implement internal abstraction for iostreams fopencookie/funclose is a non-standard API and should not be used in portable software. Additionally, the way FILE's fd is used in non-blocking mode is undefined behaviour and cannot be relied on. This introduces internal abstraction for io streams, that allows implementing the desired virtualization of read/write operations with necessary timeout handling. ASTERISK-24515 #close ASTERISK-24517 #close Change-Id: Id916aef418b665ced6a7489aef74908b6e376e85	9 years ago
Joshua Colp	d3b61a98f4	manager: Bump AMI version number. During the development of Asterisk 14 the behavior of the Command AMI action was altered such that the result was returned on lines with a prefix of "Output: ". While this was documented in the UPGRADE.txt file it is also reasonable that this should bump the AMI version number. ASTERISK-26556 Change-Id: Idf1bf01608e53f7bfdf43ddb4d0683e53f74ee42	9 years ago
Matt Jordan	edd7ae85e8	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	cc86329228	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	a72ef38113	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	7263a17ca0	channel: Fix issues in hangup scenarios caused by frame deferral ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 (cherry picked from commit `0be46aaf6b`)	9 years ago
George Joseph	0dc4567133	Revert "Revert "channel: Use frame deferral API for safe sleep."" This reverts commit `e5365dada5`. Change-Id: Icc40cf0c7687454760762912dd29e4ae79e8e9ee	9 years ago
George Joseph	6d61f7bfd1	Revert "Revert "autoservice: Use frame deferral API"" This reverts commit `edca6911f3`. Change-Id: I76030b87333a2c390cd05392b74b75678d78ddfa	9 years ago
George Joseph	f62c9c42fa	Revert "Revert "AGI: Only defer frames when in an interception routine."" This reverts commit `6bce938c2f`. Change-Id: Iadbf462bf2a52e8b2fa9ebc75b37b1f688ba51d9	9 years ago

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