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Author	SHA1	Message	Date
Tzafrir Cohen	b0c9f07f04	OpenSSL 1.1.0 support ... OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b	9 years ago
Matt Jordan	a33ed3327a	res/res_pjsip: Fix documentation whitespace issues ... Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	9 years ago
Mark Michelson	2bae7f4ed9	Merge "build_tools: Fix download_externals to handle certified branches" into 13	9 years ago
zuul	e6ecbdf001	Merge "autoconf: more variants for OSARCH linux-gnu" into 13	9 years ago
Matt Jordan	09c36a6535	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
George Joseph	29e887e9e1	build_tools: Fix download_externals to handle certified branches ... download_externals wasn't handling the "certified/13.x" version correctly. Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a	9 years ago
Tzafrir Cohen	bfb8c962c4	autoconf: more variants for OSARCH linux-gnu ... There are quite a few odd GNU/Linux platforms. Just call all of them linux-gnu. Specifically this fixes building the Debian platforms mips64el and x32. And maybe also others. ASTERISK-26546 #close Change-Id: I06ec4bd7f0ee1c84b6b24d81538223b07c4174b1	9 years ago
Timo Teräs	a1fa909033	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
Michael Kuron	0cc8351484	chan_sip: Fix segfault during module unload ... If a TCP/TLS connection was pending (not accepted and not timed out) during unload of chan_sip, Asterisk would segfault when trying to send a signal to a thread whose thread ID hadn't been recorded yet. This commit fixes that by recording the thread ID before calling the blocking connect() syscall. This was a regression introduced by 776a14386a. The above wasn't enough to fix the segfault, which was now delayed to the point where connect() timed out. Therefore, it was necessary to also remove the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be used to interruput the connect() syscall. This was a regression introduced by 5d313f51b9. ASTERISK-26586 #close Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b	9 years ago
zuul	5027b0d5e6	Merge "addons/chan_mobile: do not use strerror_r" into 13	9 years ago
gestoip2	8756ce64b7	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
Michael Kuron	8e77d6f520	tcptls: Use new certificate upon sip reload ... Previously, a TLS server socket would only be restarted upon sip reload if the bind address had changed. This commit adds checking for changes to TLS parameters like certificate, ciphers, etc. so they get picked up without requiring a reload of the entire chan_sip module. This does not affect open connections in any way, but new connections will use the new TLS parameters. The changes also apply to HTTP and Manager. ASTERISK-26604 #close Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6	9 years ago
Timo Teräs	86d824b7ff	addons/chan_mobile: do not use strerror_r ... The two reasons why it might be used are that some systems do not implement strerror in thread safe manner, and that strerror_r returns the error code in the string in case there's no error message. However, all of asterisk elsewhere uses strerror() and assumes it to be thread safe. And in chan_mobile the errno is also explicitly printed so neither of the above reasons are valid. The reasoning to remove usage is that there are actually two versions of strerror_r: XSI and GNU. They are incompatible in their return value, and there's no easy way to figure out which one is being used. glibc gives you the GNU version if _GNU_SOURCE is defined, but the same feature test macro is needed for other symbols. On all other systems you assumedly get XSI symbol, and compilation warnings as well as non-working error printing. Thus the easiest solution is to just remove strerror_r and use strerror as rest of the code. Alternative is to introduce ast_strerror in separate translation unit so it can request the XSI symbol in glibc case, and replace all usage of strerror. Change-Id: I84d35225b5642d85d48bc35fdf399afbae28a91d	9 years ago
George Joseph	425da14927	build: Backport addition of librt check to configure.ac ... A while back, a master-only change was made to check for librt which should probably have been cherry-picked to 13 at that time. Sometime between then and now, part of that change did make it into 13 but it was incomplete and non-functional. This patch backports the rest of the librt check and allows the link of libasteriskpj to use the results. Change-Id: I1424008fd8c90f389dda53162ec4a340b253a3c1	9 years ago
Joshua Colp	b83c9ff2ad	Merge "pjproject_bundled: Improve reliability of pjproject download" into 13	9 years ago
Joshua Colp	5c9680da9d	Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 13	9 years ago
Joshua Colp	9e5f76c1e1	Merge "build: Various OpenBSD issues" into 13	9 years ago
Joshua Colp	51d2d7c359	Merge "Bump ARI version to 1.10.0" into 13	9 years ago
Mark Michelson	cb624b10ae	Bump ARI version to 1.10.0 ... The video-related bridge changes mean that the version needs to be bumped. Change-Id: I41c4495068562bef03aa76728f188b8ac4bd393d	9 years ago
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