asterisk

Commit Graph

Author	SHA1	Message	Date
Martin Tomec	d13be4eff6	app_queue: Ensure member is removed from pending when hanging up. In some cases member is added to pending_members, and the channel is hung up before any extension state change. So the member would stay in pending_members forever. So when we call do_hang, we should also remove member from pending. ASTERISK-26621 #close Change-Id: Iae476b5c06481db18ebe0fa594b3e80fdc9a7d54	10 years ago
Joshua Colp	791c15942b	Merge "Fix typo in chan_sip" into 13	10 years ago
Joshua Colp	c7eb439953	Merge "chan_sip: Delete unneeded check" into 13	10 years ago
zuul	5f9316e143	Merge "Small code cleanup in chan_sip" into 13	10 years ago
zuul	949a4a443a	Merge "res_pjsip: Fix 'A = B != C' kind." into 13	10 years ago
Joshua Colp	939010fc15	Merge "res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command" into 13	10 years ago
Badalyan Vyacheslav	22820e10fe	chan_sip: Delete unneeded check P is always true. We check it before Change-Id: Iee61cda002a9f61aee26b9f66c5f9b59e3389efb	10 years ago
Badalyan Vyacheslav	6aa2c5e5f9	Small code cleanup in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I2cf7c317b106ec14440c7f1b5dcfbf03639f748a	10 years ago
Badalyan Vyacheslav	b596fac838	Fix typo in chan_sip The conditional expressions of the 'if' operators situated alongside each other are identical. Change-Id: I652b6dcddb3be007e669a6aa8107edb31a1ddafb	10 years ago
Badalyan Vyacheslav	483ed9f1aa	res_pjsip: Fix 'A = B != C' kind. Consider reviewing the expression of the 'A = B != C' kind. The expression is calculated as following: 'A = (B != C)' Change-Id: Ibaa637dfda47d51a20e26069d3103e05ce80003d	10 years ago
Kevin Harwell	f9644de0bd	Merge "res_format_attr_opus: Fix crash when fmtp contains spaces." into 13	10 years ago
Walter Doekes	41c6319c4e	chan_sip: Do not allow non-SP/HTAB between header key and colon. RFC says SIP headers look like: HCOLON = ( SP / HTAB ) ":" SWS SWS = [LWS] ; sep whitespace LWS = [WSP CRLF] 1WSP ; linear whitespace WSP = SP / HTAB ; from rfc2234 chan_sip implemented this: HCOLON = ( LOWCTL / SP ) ":" SWS LOWCTL = %x00-1F ; CTL without DEL This discrepancy meant that SIP proxies in front of Asterisk with chan_sip could pass on unknown headers with \x00-\x1F in them, which would be treated by Asterisk as a different (known) header. For example, the "To\x01:" header would gladly be forwarded by some proxies as irrelevant, but chan_sip would treat it as the relevant "To:" header. Those relying on a SIP proxy to scrub certain headers could mistakenly get unexpected and unvalidated data fed to Asterisk. This change fixes so chan_sip only considers SP/HTAB as valid tokens before the colon, making it agree on the headers with other speakers of SIP. ASTERISK-26433 #close AST-2016-009 Change-Id: I78086fbc524ac733b8f7f78cb423c91075fd489b	10 years ago
Joshua Colp	888142e891	res_format_attr_opus: Fix crash when fmtp contains spaces. When an opus offer or answer was received that contained an fmtp line with spaces between the attributes the module would fail to properly parse it and crash due to recursion. This change makes the module handle the space properly and also removes the recursion requirement. ASTERISK-26579 Change-Id: I01f53e5d9fa9f1925a7365f8d25071b5b3ac2dc3	10 years ago
George Joseph	ebc67d3053	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	10 years ago
Joshua Colp	738203f6e6	Merge "Bundled pjproject: Fix finding SIP transactions." into 13	10 years ago
Richard Mudgett	d506874477	Bundled pjproject: Fix finding SIP transactions. Occasionally SIP message transactions are not found when they should be. In the particular case an incoming INVITE transaction is CANCELed but the INVITE transaction cannot be found so a 481 response is returned for the CANCEL. The problematic calls have a '_' character in the Via branch parameter. The problem is in the pjproject PJ_HASH_USE_OWN_TOLOWER feature's code. The problem with the "own tolower" code is that it does not calculate the same hash value as when the pj_tolower() function is used. The "own tolower" code will erroneously modify the ASCII characters '@', '[', '\\', ']', '^', and '_'. Calls to pj_hash_calc_tolower() can use the PJ_HASH_USE_OWN_TOLOWER substitute algorithm when enabled. Calls to pj_hash_get_lower(), pj_hash_set_lower(), and pj_hash_set_np_lower() call find_entry() which never uses the PJ_HASH_USE_OWN_TOLOWER algorithm. As a result you may not be able to find a hash tabled entry because the calculated hash values would differ. * Simply disable PJ_HASH_USE_OWN_TOLOWER. ASTERISK-26490 #close Change-Id: If89bfdb5f301b8b685881a9a2a6e0c3c5af32253	10 years ago
George Joseph	4b233675d8	pjproject_bundled: Fix missing inclusion of symbols Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to the CFLAGS. Not sure how they went missing. Also fixed an uninstall problem where we weren't removing the symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was there, I fixed it for libasteriskssl as well. Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556	10 years ago
Joshua Colp	f08095ef18	Merge "res_pjsip_outbound_registration.c: Filter redundant statsd reporting." into 13	10 years ago
Joshua Colp	b9a79a56c7	Merge "Remove files that got merged in error somehow to the 13 branch." into 13	10 years ago
Richard Mudgett	580f83dac7	Remove files that got merged in error somehow to the 13 branch. Change-Id: Id79e2226c31084f9252d5aede9050d3cf13322c8	10 years ago
Richard Mudgett	61ba2a014a	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	10 years ago
Joshua Colp	fdf0a2afb0	Merge "res_pjsip/chan_sip: Advertise 'ws' in the SIP URI transport parameter" into 13	10 years ago
Joshua Colp	8d56016ae4	Merge "tcptls: Use new certificate upon sip reload" into 13	10 years ago
Joshua Colp	28b76ed667	Merge "PJPROJECT logging: Made easier to get available logging levels." into 13	10 years ago
Joshua Colp	c3a509be7e	Merge "res_rtp: Fix regression when IPv6 is not available." into 13	10 years ago
Joshua Colp	cd5e9a89d1	Merge "res_calendar_caldav: Add support reading gmail calendar" into 13	10 years ago
Joshua Colp	bddc60df73	Merge "Frame deferral: Re-queue deferred frames one-at-a-time." into 13	10 years ago
Joshua Colp	662a4741e2	Merge "OpenSSL 1.1.0 support" into 13	10 years ago
Guido Falsi	2ceb609edb	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	10 years ago
Eduardo S. Libardi	53459cdaa9	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	10 years ago
Richard Mudgett	44fe4a5769	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	10 years ago
Mark Michelson	17b0b91afa	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	10 years ago
zuul	eec82c6221	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 13	10 years ago
zuul	3d43e5ed3c	Merge "chan_sip: Fix segfault during module unload" into 13	10 years ago
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	10 years ago
Matt Jordan	a33ed3327a	res/res_pjsip: Fix documentation whitespace issues Tabs > Spaces. Change-Id: If1e43a71822615a898e958e0f8b2e882606f0bd0	10 years ago
Mark Michelson	2bae7f4ed9	Merge "build_tools: Fix download_externals to handle certified branches" into 13	10 years ago
zuul	e6ecbdf001	Merge "autoconf: more variants for OSARCH linux-gnu" into 13	10 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	10 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	10 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	10 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	10 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	10 years ago
zuul	5027b0d5e6	Merge "addons/chan_mobile: do not use strerror_r" into 13	10 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	10 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	10 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	10 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	10 years ago
Joshua Colp	b83c9ff2ad	Merge "pjproject_bundled: Improve reliability of pjproject download" into 13	10 years ago
Joshua Colp	5c9680da9d	Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 13	10 years ago

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