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Author	SHA1	Message	Date
Richard Mudgett	d416dd7f65	res_pjsip_outbound_registration.c: Filter redundant statsd reporting. ... Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646	9 years ago
zuul	1de7e0eefa	Merge "tcptls: Use new certificate upon sip reload" into 14	9 years ago
Joshua Colp	7b1d990683	Merge "PJPROJECT logging: Made easier to get available logging levels." into 14	9 years ago
Joshua Colp	73f4df291c	Merge "res_rtp: Fix regression when IPv6 is not available." into 14	9 years ago
zuul	194e327336	Merge "res_calendar_caldav: Add support reading gmail calendar" into 14	9 years ago
Joshua Colp	f1a7cf64e2	Merge "Frame deferral: Re-queue deferred frames one-at-a-time." into 14	9 years ago
Joshua Colp	02e3a69b07	Merge "OpenSSL 1.1.0 support" into 14	9 years ago
Guido Falsi	798179995b	res_rtp: Fix regression when IPv6 is not available. ... The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e	9 years ago
Joshua Colp	12517ad1f3	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 14	9 years ago
Eduardo S. Libardi	e147c91a65	res_calendar_caldav: Add support reading gmail calendar ... The response from gmail calendar includes the string name "caldav:calendar-data". res_calendar_caldav implements the example included in RFC 4791: string "C:calendar-data". When reading the calendar, res_calendar_caldav compare the string and if does not match just discards the event. This commit compares the response to both strings, successfully loading gmail calendar events. Writing to gmail calendar is working prior to this fix. ASTERISK-26624 Reported by: Eduardo S. Libardi Change-Id: Ia1eef10552ae616efb645d390f5ffe81260d7d4a	9 years ago
Richard Mudgett	846c77b28c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	9 years ago
Mark Michelson	2bc0b595a1	Frame deferral: Re-queue deferred frames one-at-a-time. ... The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d	9 years ago
Joshua Colp	36d380d71c	Merge "chan_sip: Fix segfault during module unload" into 14	9 years ago
Tzafrir Cohen	0f33fa4ead	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
Alexei Gradinari	d49fbc5f4e	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
Joshua Colp	a152581464	Merge "res/res_pjsip: Fix documentation whitespace issues" into 14	9 years ago
Joshua Colp	628b69dd00	Merge "build_tools: Fix download_externals to handle certified branches" into 14	9 years ago
Matt Jordan	dc7d8037f3	res/res_pjsip: Fix documentation whitespace issues ... Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	9 years ago
George Joseph	6c15f1cac9	Merge "autoconf: more variants for OSARCH linux-gnu" into 14	9 years ago
Matt Jordan	9fe1a62f7f	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	a51199123f	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	0169b09bee	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	04ceef0e83	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	cbdf9cc948	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
Joshua Colp	40becc5c84	Merge "addons/chan_mobile: do not use strerror_r" into 14	9 years ago
gestoip2	f8f197a6ee	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	cc619d30bf	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
zuul	bba4204fa1	Merge "Add support for older name resolving version libraries like openBSD" into 14	9 years ago
Timo Teräs	1dc4936c38	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	34a763415f	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	7ec27528b2	Merge "pjproject_bundled: Improve reliability of pjproject download" into 14	9 years ago
snuffy	729f1b63e2	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
Joshua Colp	002a8e4f10	Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 14	9 years ago
Joshua Colp	184524a8cf	Merge "Bump ARI version to 2.0.0" into 14	9 years ago
Mark Michelson	7af1aae57f	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
Joshua Colp	3a151b5205	Merge "build: Various OpenBSD issues" into 14	9 years ago
George Joseph	6d34d0ab36	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	4e64e4f35f	main/app.c: Transmit Silence on ControlPlayback pause ... ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8	9 years ago
Mark Michelson	7a665c5c6e	manager: update minor version ... Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: Idf84507354170400813cda780906c94c9f1b60b4	9 years ago
zuul	38d023c628	Merge "res_pjsip_outbound_authenticator_digest.c: Fix memory pool leak." into 14	9 years ago
George Joseph	329c554e7b	Merge "res_format_attr_opus: Fix fmtp generation." into 14	9 years ago
George Joseph	54d7e65014	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
Joshua Colp	0ee5b4dbe9	Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded." into 14	9 years ago
George Joseph	0825528322	Revert "Revert "AGI: Only defer frames when in an interception routine."" ... This reverts commit 7155020c96. Change-Id: I4e1681d2ca8bf36eda849f129bc0dcbc39ef5bc6	9 years ago
George Joseph	6b2efc116f	Revert "Revert "autoservice: Use frame deferral API"" ... This reverts commit 8b9996a90c. Change-Id: Ibb2fccfc37188dc54f3776c2d5b41cc425d5cd9c	9 years ago
George Joseph	f6acb765a8	Revert "Revert "channel: Use frame deferral API for safe sleep."" ... This reverts commit 97679ee846. Change-Id: Ib68f76e75882c0230378c4fed97964988e690f3c	9 years ago
Joshua Colp	fd52740044	Merge "pjproject: Use a much higher limit for PJ_ICE_MAX_CHECKS" into 14	9 years ago
Joshua Colp	00329115e2	Merge "apps/app_echo: Only relay a single video source change frame" into 14	9 years ago
Joshua Colp	aa1482aed4	Merge "channel: Fix issues in hangup scenarios caused by frame deferral" into 14	9 years ago
Joshua Colp	caa207bc4f	Merge "Revert "Revert "Add API for channel frame deferral.""" into 14	9 years ago