2017-03-07 18:43 +0000 Asterisk Development Team * asterisk certified/13.13-cert2 Released. 2016-11-15 15:01 +0000 [44cac45610] Alexei Gradinari * 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 (cherry picked from commit cf6d13180effc92a2483dccc68f2f188689a40fa) 2017-02-13 17:11 +0000 [0d7f99a087] Richard Mudgett * pjproject: Fixes to resolve DNS SRV crashes. * Re #1945 (misc): Don't trigger SRV complete callback when there is a parse error. * srv_resolver.c: Don't try to send query if already considered resolved. ** In resolve_hostnames() don't try to resolve a query that is already considered resolved. ** In resolve_hostnames() fix DNS typo in comments. ** In build_server_entries() move a common expression assigning to cnt earlier. * sip_transport.c: Fix tdata object name to actually contain the pointer. It helps if the logs referencing a tdata object buffer actually have a name that includes the correct pointer as part of the name. Also since the tdata has its own pool it helps if any logs referencing the pool have the same name as the tdata object. This change brings tdata logging in line with how tsx objects are named. ASTERISK-26669 #close ASTERISK-26738 #close Change-Id: I56af2ded25476b3e870ca586ee69ed6954ef75af 2017-02-20 13:38 +0000 [35881858db] Richard Mudgett * pjproject: Increase SENDER_WIDTH column size for 64-bit system logs. ASTERISK-26669 ASTERISK-26738 Change-Id: Ibae6fc8cae69a1f04df0c577c4c11200499d6fe0 2017-02-06 14:26 +0000 [0d4412f2b3] Richard Mudgett * pjsip_distributor.c: Update some debug messages to get transaction name. * Removed overloaded unmatched response ignore. We obviously sent the request so we shouldn't ignore it because it isn't new work. ASTERISK-26669 ASTERISK-26738 Change-Id: I55fb5cadc83a8e6699b347c6dc7fa32c5a617d37 2017-02-04 16:00 +0000 [4af241ab60] Richard Mudgett * res_pjsip: Record the serializer earlier on the tdata. When PJPROJECT needs to do a DNS resolution and there is not a cached entry available, the SIP request message goes out on the PJSIP monitor thread instead of the original serializer thread. Thus when the response comes back it does not get processed by the original sending serializer. This patch records the serializer on tdata before passing a request message to PJPROJECT where it can in Asterisk code. There are several places in PJPROJECT for outbound registration and publishing support that would need to record the serializer. Unfortunately, without replacing the PJPROJECT DNS resolver as was done in v14 we cannot fix those without modifying PJPROJECT. Even if we backported the DNS resolver from v14, the outbound registration refresh timer does not go out on a serializer thread but the PJSIP monitor thread. Fortunately, Asterisk's outbound publish support doesn't use the auto refresh timer that would also not go out under the serializer thread. This patch is v13 only. ASTERISK-26669 ASTERISK-26738 Change-Id: I9997b9ed6dbcebd2c37d6a67dc6dcee9c78914a4 2017-02-13 19:25 +0000 Asterisk Development Team * asterisk certified/13.13-cert1 Released. 2017-02-08 12:58 +0000 [0ef6b6960d] gtjoseph * Update for certified/13.13-cert1-rc4 2017-02-08 11:50 +0000 [7603c4f32b] Mark Michelson * Revert "Update qualifies when AOR configuration changes." This reverts commit 6492e91392b8fd394193e411c6eb64b45486093f. The change in question was intended to prevent the need to reload in order to update qualifies on contacts when an AOR changes. However, this ended up causing a deadlock instead. Change-Id: I1a835c90a5bb65b6dc3a1e94cddc12a4afc3d71e 2017-02-03 13:58 +0000 [47febcb927] Mark Michelson * Update for certified/13.13-cert1-rc3 2017-01-31 18:28 +0000 [5c90c1e9f5] Richard Mudgett * channel.c: Fix unbalanced read queue deadlocking local channels. Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6 2017-01-31 16:38 +0000 [5c2b7e34ff] Richard Mudgett * res_agi: Prevent an AGI from eating frames it should not. (Re-do) A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43 2017-01-31 16:32 +0000 [7d291e9ef7] Richard Mudgett * Frame deferral: Revert API refactoring. There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496 2017-01-10 17:37 +0000 [640d3b21d1] Richard Mudgett * astobj2.c: Add excessive ref count trap. Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a 2017-01-31 11:17 +0000 [107c8a7e19] Joshua Colp * res_pjsip: Handle invocation of callback on outgoing request when error occurs. There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91 2017-02-01 13:54 +0000 [3eb5f42090] Mark Michelson * Update qualifies when AOR configuration changes. Prior to this change, qualifies would only update in the following cases: * A reload of res_pjsip.so was issued. * A dynamic contact was re-registered after its AOR's qualify_frequency had been changed This does not work well if you are using realtime for your AORs. You can update your database to have a new qualify_frequency, but the permanent contacts on that AOR will not have their qualifies updated. And the dynamic contacts on that AOR will not have their qualifies updated until the next registration, which could be a long time. This change seeks to fix this problem by making it so that whenever AOR configuration is applied, the contacts pertaining to that AOR have their qualifies updated. Additions from this patch: * AOR sorcery objects now have an apply handler that calls into a newly added function in the OPTIONS code. This causes all contacts associated with that AOR to re-schedule qualifies. * When it is time to qualify a contact, the OPTIONS code checks to see if the AOR can still be retrieved. If not, then qualification is canceled on the contact. Alterations from this patch: * The registrar code no longer updates contact's qualify_frequence and qualify_timeout. There is no point to this since those values already get updated when the AOR changes. * Reloading res_pjsip.so no longer calls the OPTIONS initialization function. Reloading res_pjsip.so results in re-loading AORs, which results in re-scheduling qualifies. Change-Id: I2e7c3316da28f389c45954f24c4e9389abac1121 2017-01-31 12:46 +0000 [0611290911] gtjoseph * debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c 2017-01-25 06:50 +0000 [805928c98b] gtjoseph * debug_utilities: Add ast_logescalator The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543 2017-01-25 15:26 +0000 [1997157e7e] Mark Michelson * Add reload options to CLI/AMI stale object commands. Marking an object as stale in a memory cache is supposed to prime the cache so that the next time the item is retrieved, the stale item is deleted from the cache and a background task is run to re-populate the cache with a fresh version of the object. The problem is, there are some object types out there for which there is no natural reason that they would be retrieved from the backend with any regularity. Outbound PJSIP registrations are a good example of this. At startup, they are read, and an object-specific state is created that refers to the initially-retrieved object for all time. Adding the "reload" option to the CLI/AMI commands gives the cache the opportunity to manually re-retrieve the object from the backend, both storing the new object in the cache and applying the new object's configuration to the module that uses that object. Change-Id: Ieb1fe7270ceed491f057ec5cbf0e097bde96c5c8 2016-12-09 12:23 +0000 [92bdcfd57e] Martin Tomec * 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 (cherry picked from commit d13be4eff699449172efbd9fed0ee97f6a790b6a) 2017-01-20 21:13 +0000 [9a4434eb74] Richard Mudgett * PJPROJECT logging: Fix detection of max supported log level. The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4 2017-01-19 09:05 +0000 [6b0d734312] gtjoseph * ari: Implement 'debug all' and request/response logging The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf 2017-01-23 09:10 +0000 [28733bb0ab] gtjoseph * pjproject_bundled: Fix setting max log level An earlier attempt to prevent pjsua from spitting out an extra 6795 lines of debug output every time the testsuite called it was also turning off the ability for asterisk to output debug info when it needed to. This patch reverts the earlier fix and instead adds a pjproject patch that sets the startup log level to 1 for pjsua pjsystest and the pjsua python binding. This is an asterisk-only patch that does not affect pjproject functionality and will not be submitted upstream. Change-Id: I347a8b58b2626f2906ccfc1d339e907627a0c9e8 2017-01-13 11:03 +0000 [0d2f17b22c] gtjoseph * debug_utilities: Create ast_loggrabber ast_loggrabber gathers log files from customizable search patterns, optionally converts POSIX timestamps to a readable format and tarballs the results. Also a few tweaks were made to ast_coredumper. Change-Id: I8bfe1468ada24c1344ce4abab7b002a59a659495 (cherry picked from commit 5fa1c56d7e76999aa14f133a33f6b168e7c3b99c) 2017-01-19 13:18 +0000 [92876c1c2a] Mark Michelson * Update for certified/13.13-cert1-rc2 2017-01-08 10:29 +0000 [52bee5df9e] gtjoseph * pjproject_bundled: Fix compilation with MALLOC_DEBUG When MALLOC_DEBUG was specified, make was failing. Immediately remaking would work. The issues was in the ordering of the make dependencies. Change-Id: If6030b54fc693f3179f32bfd20c6b5d5f1b3f7cd (cherry picked from commit f8cd73ec3c159f2e6c464952c92d8fdb69394371) 2017-01-03 15:14 +0000 [08857b6e0e] gtjoseph * pjproject_bundled: Compile pjsua with max log level = 2 A while back, we changed config_site.h to set PJ_LOG_MAX_LEVEL = 6. This allowed us to control the log level better from inside Asterisk. An unfortunate side effect of this was that the pjsua binary and python bindings were also compiled with log level set to 6 so whenever a testsuite test that uses pjsua runs, it spits out 6795 lines of debug in an instant even before the test starts. I believe this overruns the Jenkins capture buffer and prevents the test from properly terminating. In turn, this results in the testsuite just hanging until the job is killed. It's more frequent on the higher end agents because they can spit out the messages faster. Unfortunately, the messages are all spit out before we have control of the python pj.Lib instance where we can set logging levels so the only alternative was to actually compile pjsua and _pjsua.so with an overridden PJ_LOG_MAX_LEVEL. Although defining a lower max level was done in the Makefile, the define in config_site.h had to be wrapped with "#ifndef" so the change would take effect. Change-Id: I2af9e7d48dde1927279c586c9c725d868fe6f3ff (cherry picked from commit d7e5a747c312de18647213359103ce6022776864) 2016-12-18 15:23 +0000 [7aacc0fc7f] gtjoseph * pjproject_bundled: Make build single threaded There were just too many issues in various environments with multi threaded building of pjproject. It doesn't really speed things up anyway since asterisk is already being compiled in parallel. Change-Id: Ie5648fb91bb89b4224b6bf43a0daa1af793c4ce1 (cherry picked from commit 815f7551550908c83220196ba08742af0c745772) 2016-11-23 18:27 +0000 [3a8a42b404] Richard Mudgett * 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 2017-01-10 18:10 +0000 [8e5e3c2b0c] gtjoseph * debug_utilities: Create the ast_coredumper utility This utility allows easy manipulation of asterisk coredumps. * Configurable search paths and patterns for existing coredumps * Can generate a consistent coredump from the running instance * Can dump the lock_infos table from a coredump * Dumps backtraces to separate files... - thread apply 1 bt full -> .thread1.txt - thread apply all bt -> .brief.txt - thread apply all bt full -> .full.txt - lock_infos table -> .locks.txt * Can tarball corefiles and optionally delete them after processing * Can tarball results files and optionally delete them after processing * Converts ':' in coredump and results file names '-' to facilitate uploading. Jira for instance, won't accept file names with colons in them. Tested on Fedora24+, Ubuntu14+, Debian6+, CentOS6+ and FreeBSD9+[1]. [1] For *BSDs, the "devel/gdb" package might have to be installed to get a recent gdb. The utility will check all instances of gdb it finds in $PATH and if one isn't found that can run python, it prints a friendly error. Change-Id: I935d37ab9db85ef923f32b05579897f0893d33cd (cherry picked from commit 47474cfd54a9185c1433464ccfd6301427a03957) 2016-12-22 16:00 +0000 [cedf8a21a1] Joshua Colp * chan_pjsip: Use session for retrieving CHANNEL() information. The CHANNEL() dialplan function implementation for PJSIP allows querying of PJSIP specific information. This used the channel passed in to get the PJSIP session and associated information. It is possible for this channel to be masqueraded and end up as a different channel type by the time the information request is actually acted upon. This change retrieves the PJSIP session safely and accesses data from it (including channel). This provides a guarantee that the session and channel will not be altered when the request is being acted upon. ASTERISK-26673 Change-Id: I335e12b89e1820cafdd92b3e7526b8ba649eb7e6 2016-12-23 12:10 +0000 [92235dba88] Richard Mudgett * bridge_native_rtp.c: Fix native rtp bridge data race. native_rtp_bridge_compatible() didn't lock the bridge channels before checking the channels for native bridging ability. As a result, one of the channel's native format capabilities structure got replaced out from under the native bridge check. Use of a stale pointer to freed memory causes bad things to happen. MALLOC_DEBUG, DO_CRASH, and the tests/channels/pjsip/transfers/blind_transfer/caller_direct_media testsuite test caught this. * Add missing channel locking in native_rtp_bridge_compatible(). Change-Id: If25fdb3ac8e85563c4857fb8216b3d9dc3d0fa53 2016-12-21 16:28 +0000 [d8747659f0] Richard Mudgett * res_rtp_asterisk.c: Fix uninitialized memory crash. ast_rtp_remote_address_set() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor(). * Optimized out the 'us' struct variable. ASTERISK-26672 #close Change-Id: I4acea5dcdf0813da2c7d3e11c2d6067d160d17dc 2016-12-21 17:54 +0000 [a9400da2d3] Richard Mudgett * chan_rtp.c: Fix uninitialized memory crash. unicast_rtp_request() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort the UnicastRTP channel request if it fails. ASTERISK-26672 Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0 2016-12-21 17:55 +0000 [a2c695cd18] Richard Mudgett * res_rtp_asterisk.c: Initialize ourip passed to ast_find_ourip(). We access uninitialized memory when the 'ourip' parameter does not have an initial guess to our IP address. ASTERISK-26672 Change-Id: I35507ea1ad7455d2be188f6ccdd4add7bd150e15 2016-12-21 16:25 +0000 [a3502c1885] Richard Mudgett * acl.c: Improve ast_ouraddrfor() diagnostic messages. * Made not generate strings unless they will actually be used. ASTERISK-26672 Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 2016-12-14 14:21 +0000 [a3da3bb406] Richard Mudgett * chan_dahdi.c: Fix bounds check regression. Caused by ASTERISK-25494 Change-Id: I1fc408c1a083745ff59da5c4113041bbfce54bcb 2016-12-13 14:34 +0000 [1bb47bc3b0] Richard Mudgett * res_pjsip: Add/update ERROR msg if invalid URI. ASTERISK-24499 Change-Id: Ie305153e47e922233b2ff24715e0e326e5fa3a6c 2016-12-12 18:38 +0000 [ee9c8d0c97] Richard Mudgett * MESSAGE: Flush Message/ast_msg_queue channel alert pipe. ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 2016-12-13 14:06 +0000 [a209faa94f] gtjoseph * res_sorcery_memory_cache: Change an error to a debug message When a sorcery user calls ast_sorcery_delete on an object that may have already expired from the cache, res_sorcery_memory_cache spits out an ERROR. Since this can happen frequently and validly when an inbound registration expires after the cache entry expired, the errors are unnecessary and misleading. Changed to a debug/1. Change-Id: Idf3a67038c16e3da814cf612ff4d6d18ad29ecd7 2016-11-30 09:31 +0000 [2021b5380d] Walter Doekes * 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] 1*WSP ; 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 2016-11-14 18:18 +0000 [d27eae001d] Joshua Colp * 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 2016-12-06 14:54 +0000 [f243f7fb4b] gtjoseph * 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 2016-12-06 16:45 +0000 [c7c2db5a29] Richard Mudgett * 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 2016-12-06 12:06 +0000 [221e838b26] gtjoseph * 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 2016-11-30 10:48 +0000 [492b37429c] Mark Michelson * 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 2016-11-28 11:03 +0000 [d4d1909077] gtjoseph * build_tools: Fix download_externals to handle certified branches download_externals wasn't handling the "certified/13.x" version correctly. Change-Id: I124d195bb117ca36fd7bf1150c630f3b474a9d9a 2016-11-23 15:58 +0000 [33a0d64eab] Kevin Harwell * Update for certified/13.13-cert1-rc1 2016-11-23 15:20 +0000 [907160ee21] Kevin Harwell * app_talkdectect: Now core supported, enable for cert Change-Id: Ic0b2cacb21a6e11a25ebbff7e508e106ea156f6c 2016-11-23 15:01 +0000 [0cd0495732] Kevin Harwell * Disable extended support modules Change-Id: Ib6b4f9451b5b68b738d8ab07a27de1c87c28f819 2016-11-23 14:57 +0000 [854196eea9] Kevin Harwell * .version: Update for certified/13.13 Change-Id: Ia1a0f035359d88b8885c7aca22f0d70b73aeb05d 2016-11-23 09:26 +0000 [fdde690e0f] Kevin Harwell * Update for 13.13.0 2016-11-22 12:02 +0000 [f93e55d124] Kevin Harwell * Update for 13.13.0-rc2 2016-11-21 09:40 +0000 [e246b36a3c] gtjoseph * 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 2016-11-22 11:20 +0000 [855f05e525] Kevin Harwell * Update for 13.13.0 2016-11-18 12:59 +0000 [751d43e8e4] Joshua Colp * Update for 13.13.0-rc1 2016-11-18 09:45 +0000 [cb624b10ae] Mark Michelson * Bump ARI version to 1.10.0 The video-related bridge changes mean that the version needs to be bumped. Change-Id: I41c4495068562bef03aa76728f188b8ac4bd393d 2016-11-17 10:50 +0000 [bde3d022a3] Mark Michelson * manager: update minor version Based on bridge video AMI event changes, bump the minor version of AMI. Change-Id: I02586bd6cafc0baa33ea98c2f75356c0f5e03435 2016-11-16 20:24 +0000 [b213045fe4] gtjoseph * 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 2016-11-14 18:45 +0000 [404596b790] gtjoseph * channel: Fix issues in hangup scenarios caused by frame deferral ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 2016-11-16 15:42 +0000 [2c031b67d3] Mark Michelson * 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: 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 2016-11-15 16:23 +0000 [ed0f1afc8c] Richard Mudgett * 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 2016-11-14 14:36 +0000 [e632222bc4] Richard Mudgett * 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 2016-11-15 12:01 +0000 [c92dcc76da] gtjoseph * 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 2015-05-14 17:12 +0000 [7b96e8cc3d] Maciej Szmigiero * 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 ASTERISK-25063 #close Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f 2016-11-14 15:57 +0000 [0790aa528a] Matt Jordan * 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 2016-11-14 15:32 +0000 [993a6f96c7] Matt Jordan * 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 2016-11-08 10:11 +0000 [d23b4af477] Matt Jordan * 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 2016-11-14 14:22 +0000 [404a62eeee] gtjoseph * Revert "Revert "channel: Use frame deferral API for safe sleep."" This reverts commit 58c88cfbaa80cb43419cde9186d643d1c5d24baf. Change-Id: I72692e2b2e83ef6da9390075ff20b138b2c374b6 2016-11-14 14:22 +0000 [09d8febc91] gtjoseph * Revert "Revert "autoservice: Use frame deferral API"" This reverts commit 1df434e2b4bd7cc34b9b4addf405a3caa7ac16b8. Change-Id: Id2b8a8bccbb4bbdd82b792275d4cd6f32563e401 2016-11-14 14:21 +0000 [ffad2b44df] gtjoseph * Revert "Revert "AGI: Only defer frames when in an interception routine."" This reverts commit 6be5d8de0da7e804544507f70382425af9a07b3f. Change-Id: I4b548137f52ae0686d8f09e21496b778d1c6a797 2016-11-14 14:21 +0000 [2fefb6187f] gtjoseph * Revert "Revert "Add API for channel frame deferral."" This reverts commit 6b5a7ced136b7178ae0b2ba39221eba1cd2e37c9. Change-Id: I61d1dbb2e69e1977f684b7dfc8e98211024e1cd1 2016-11-14 12:16 +0000 [5e0c224043] gtjoseph * 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 2016-11-11 02:41 +0000 [3faca1d4ff] Igor Goncharovskiy * 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 2016-09-23 17:54 +0000 [412d43fa21] Richard Mudgett * 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 2016-09-21 15:10 +0000 [2e7fc56d3c] Richard Mudgett * res_pjsip: Fix tdata leaks in off nominal paths. Change-Id: Ie83e06e88c2d60157775263b07e40b61718ac97b 2016-10-24 12:41 +0000 [da68b185b3] Richard Mudgett * res_pjsip_registrar_expire.c: Remove extra linefeed in debug message. Change-Id: I1f9adb911f23376503396ec8867e8005b755eb94 2016-11-10 10:57 +0000 [b70eb07c53] Joshua Colp * 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 2016-11-10 08:33 +0000 [71dc333565] Joshua Colp * app_queue: Add mention of 'ABANDON' variable to CHANGES. ASTERISK-26558 Change-Id: I1127010181e79c8ac291f72f036cb8e430dc7f7e 2016-11-10 07:41 +0000 [6b5a7ced13] gtjoseph * Revert "Add API for channel frame deferral." This reverts commit 9231a56cf3d6f5eca1bf2d37d827453400690773. Multiple testsuite failures were detected after the fact. Change-Id: I3bac8d7c3ddb69a4ddf6c5d6de0ffa5ff7ff3af7 2016-11-10 07:41 +0000 [6be5d8de0d] gtjoseph * Revert "AGI: Only defer frames when in an interception routine." This reverts commit 5c10091f3d1430c6fc04015226f8c3e3aa9d8282. Multiple testsuite failures were detected after the fact. Change-Id: I397a841acc17ae230c512449cd6bed89d2ef3b73 2016-11-10 07:41 +0000 [1df434e2b4] gtjoseph * Revert "autoservice: Use frame deferral API" This reverts commit 2e3a3545754749de21873bfdc6d1a40ec7d8893f. Multiple testsuite failures were detected after the fact. Change-Id: Ia45fa4633fae74dca345b24bb6722737c63035de 2016-11-10 07:40 +0000 [58c88cfbaa] gtjoseph * Revert "channel: Use frame deferral API for safe sleep." This reverts commit 44f7e252397fd87420b3374df26941d7436401b3. Multiple testsuite failures were detected after the fact. Change-Id: I56299087da22128a95f0c8f3955f740890d7ca65 2016-11-09 18:18 +0000 [a562fbe618] gtjoseph * build: Fix default values for some SANITIZER options 2 of the sanitizers didn't have default values so in systems that don't support sanitizers menuselect would spit out warnings. They were harmless but confusing. They've now been set to "0". Change-Id: I08dc495e3b83f1feac3160b421f538c375fc5d58 2016-11-06 06:04 +0000 [7fd5031c1c] Sebastian Gutierrez * app_queue: new variable set when abandoned sets the variable ABANDONED to TRUE if the call was not answered. ASTERISK-26558 Change-Id: I4729af9bff4eba436d8a776afd3374065d0036d3 2016-11-08 10:48 +0000 [e043d1a55c] Mark Michelson * res_pjsip_session: Do not call session supplements when it's too late. res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 2016-11-03 16:46 +0000 [44f7e25239] Mark Michelson * channel: Use frame deferral API for safe sleep. This is another case where manual frame deferral can be replaced with centralized routines instead. Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e 2016-11-03 16:46 +0000 [2e3a354575] Mark Michelson * autoservice: Use frame deferral API Rather than use manual frame deferral, just let the channel API do it for us. ASTERISK-26343 Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49 2016-11-03 16:42 +0000 [5c10091f3d] Mark Michelson * AGI: Only defer frames when in an interception routine. AGI recently was modified to defer important frames. This was because when AGI was used in a connected line interception routine, the resulting connected line frame would end up getting discarded by the AGI. However, this caused bad behavior in other cases. Specifically, during a transfer, if someone attempted to manually set the Caller ID on a channel in an AGI, the deferred connected line frame would end up overwriting what had been manually set in the AGI. Since the initial issue was specific to interception routines, this change removes the manual frame deferral from AGI and instead uses the new frame deferral API in interception routines. ASTERISK-26343 #close Reported by Morton Tryfoss Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208 2016-11-03 16:36 +0000 [9231a56cf3] Mark Michelson * Add API for channel frame deferral. There are several places in Asterisk that have duplicated logic for deferring important frames until later. This commit adds a couple of API calls to facilitate this automatically. ast_channel_start_defer_frames(): Future reads of deferrable frames on this channel will be deferred until later. ast_channel_stop_defer_frames(): Any frames that have been deferred get requeued onto the channel. ASTERISK-26343 Change-Id: I3e1b87bc6796f222442fa6f7d1b6a4706fb33641 2016-11-03 07:42 +0000 [a9ac1f5de4] Alexander Anikin * chan_ooh323: Fixes to work right with Cisco devices Changed output packets queue processing algo to one read-one write instead of all read-all send Remove h.245 tunneling parameter from ReleaseComplete packet ASTERISK-24400 #close Reported by: Dmitry Melekhov Tested by: Dmitry Melekhov Change-Id: I0b31933b062a21011dbac9a82b8bcfe345f406f6 2016-11-03 13:10 +0000 [0ee249075a] Alexander Anikin * chan_ooh323: reset rrq count on gk registration reset registration attempts count on success registration on gatekeeper Change-Id: I5f47351852e0ca76c9ac78421659600e0f106336 2016-11-06 03:46 +0000 [59c23e1768] Michael Kuron * automon: restore mixing of the both channels after recording stops This is a regression over Asterisk 11, introduced by 2dc8a060064f359a17f5ebcd515d85fe5203c019. Previously, recordings started via the automon DTMF code would automatically be mixed together using sox because app_monitor would be called with the m option. This commit restores this behavior. Change-Id: Ibaf58684285c3f1b6ca3714524e6d638ae3b3759 2016-11-04 15:42 +0000 [e79acaeb75] Matt Jordan * res_http_websocket: Increase the buffer size for non-LOW_MEMORY systems Not surprisingly, using Respoke (and possibly other systems) it is possible to blow past the 16k limit for a WebSocket packet size. This patch bumps it up to 32k, which, at least for Respoke, is sufficient. For now. Because 32k is laughable on a LOW_MEMORY system (as is 16k, for that matter), this patch adds a LOW_MEMORY directive that sets the buffer to 8k for systems who have asked for their reduced memory availability to be considered. Change-Id: Id235902537091b58608196844dc4b045e383cd2e 2016-11-04 15:40 +0000 [7a83196985] Matt Jordan * res_stasis: Set a video source mode on Stasis created bridges When a bridge is created via ARI (through res_stasis), no video source mode is set by default. As a result, any endpoint sending video media won't ever see any video reflected back to it. This patch defaults a bridge to a 'follow the talker' video mode. Further work can be done to add routes that allow for the video mode to be controlled through the /bridges resource. Change-Id: I7e9d530a5d7a97a4524a9ee4e468e1a6b3443866 2016-11-04 15:37 +0000 [e7dc536b7a] Matt Jordan * main/bridge_channel: Fix channel reference leak on video source When a channel is made the video source, the bridge holds a reference to it. Whenever the video source changes, that reference is released. However, a ref leak does occur if the channel leaves the bridge (such as being hung up) while it is the video source, as the bridge never releases the ref in such a case. This patch adds a line to the bridge_channel_internal_join routine such that, when a channel finishes its time in the bridge, it notifies the bridge via ast_bridge_remove_video_src that if it is a video source its reference should be released. ASTERISK-26555 #close Change-Id: I3a2f5238a9d2fc49c591f0e65199d782ab0be76a 2016-11-04 15:36 +0000 [7c824b955d] Matt Jordan * main/bridge: Add some verbose logging for video source changes It's actually quite useful to see the source of a video stream change. This doesn't happen terribly often, even with talk detection - but when it does, it's nice to know which channel is now providing your video stream. As a verbose 5 level message, it shouldn't be terribly spammy or costly to have, and is 'lower level' then most other verbose messages that the bridge system emits. ASTERISK-26555 Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c 2016-11-04 15:33 +0000 [fd6af2dee8] Matt Jordan * bridges/bridge_softmix: Remove SSRC changes on join/leave; update video source WebRTC clients really, really want to know the SSRC of the media they're getting. Changing the SSRC is generally not a good thing. bridge_softmix, starting in Asterisk 12, started changing the SSRC of parties as they joined or left the bridge. With most phones, this isn't a problem: phones just play back the stream they're getting. With WebRTC clients, however, the SSRC is tied to a media stream that may be negotiated. When a new SSRC just shows up, the media can be dropped. As it turns out, the SSRC change shouldn't even be necessary. From the perspective of the client, it's still talking to Asterisk with the same media stream: why indicate that the far party has suddenly changed to a different source of media? This patch opts to just remove the SSRC changes. With this patch, video clients that join/leave a softmix bridge actually get the video stream instead of freaking out. ASTERISK-26555 Change-Id: I27fec098b32e7c8718b4b65f3fd5fa73527968bf 2016-10-28 15:11 +0000 [bd4d7d8ad0] Kevin Harwell * stasis_recording/stored: remove calls to deprecated readdir_r function. The readdir_r function has been deprecated and should no longer be used. This patch removes the readdir_r dependency (replaced it with readdir) and also moves the directory search code to a more centralized spot (file.c) Also removed a strict dependency on the dirent structure's d_type field as it is not portable. The code now checks to see if the value is available. If so, it tries to use it, but defaults back to using the stats function if necessary. Lastly, for most implementations of readdir it *should* be thread-safe to make concurrent calls to it as long as different directory streams are specified. glibc falls into this category. However, since it is possible that there exist some implementations that are not safe, locking has been added for those other than glibc. ASTERISK-26412 ASTERISK-26509 #close Change-Id: Id8f54689b1e2873e82a09d0d0d2faf41964e80ba 2016-11-04 10:57 +0000 [cb30963d22] Kevin Harwell * Revert "chan_sip: Fix lastrtprx always updated" This reverts commit 93332cb1d0eea18021ea6538237297e627d6e2fc. Unfortunately, the aforementioned commit caused a regression (incoming calls would eventually disconnect). Thus it is being removed. ASTERISK-26523 #close ASTERISK-25270 Change-Id: Ibf5586adc303073a8eac667a4cbfdb6be184a64d 2016-11-02 10:52 +0000 [3a1f9c5dab] Joshua Colp * res_stasis: Don't unsubscribe from a NULL bridge. A NULL bridge has special meaning in res_stasis for unsubscribing. It means that a subscription to ALL bridges should be removed. This should not be done as part of the normal subscription management in the res_stasis channel loop. ASTERISK-26468 Change-Id: I6d5bea8246dd13a22ef86b736aefbf2a39c15af0 2016-11-03 13:45 +0000 [eceab15f33] Alexander Anikin * chan_ooh323: Fix infinite loop on read second part of H.225 packet Fix logic on read second part of H.225 packet. There was infinite loop on wrong connections due to read before poll. Change-Id: I42b4bf75c46e4a5c5df5c5ca1f0bd74b8944e7ff 2016-11-03 11:55 +0000 [a9992da4aa] gtjoseph * pjproject_bundled: Fix issue with libasteriskpj needing libresample libresample is only needed by pjproject if we're building pjsua, which we only do if TEST_FRAMEWORK is selected. It's required by pjsua to process audio which is needed by some testsuite tests. Unfortunately, pjproject relies on a newer version of libresample than the version that ships by most distros so we need to compile the version that's bundled with pjproject. Since we only need it for pjsua, we DON'T want it's symbols exposed when we actually build asterisk. There was a problem however... TEST_FRAMEWORK is only known AFTER we've already run ./configure on both asterisk and pjproject but pjproject's ./configure needs to test it to know whether to set up to build libresample or not. The previous way of figuring this out was to always tell ./configure "yes" but not actually build the library. This caused an issue where building libasteriskpj was being told to include libresample but it wasn't actually there. The solution is to still do a default pjproject configure during an asterisk ./configure but if makeopts or menuselect.makeopts changes subsequently, we now reconfigure pjproject, taking into account the current state of TEST_FRAMEWORK. Previously, if makeopts or menuselect.makeopts changed, only a recompile of pjproject was done. Change-Id: I9b5d84c61384a3ae07fe30e85c49698378cc4685 2016-11-01 19:48 +0000 [714412f6c4] Sebastian Gutierrez * chan_sip: add missing account code Added missing account to AMI event of sip show peers ASTERISK-26176 #close Change-Id: Ieb6c2c80a838a1b59c82103eba4c63ba238dc482 2016-09-13 04:08 +0000 [0cf1778eed] Alexander Traud * rtp_engine: Allow more than 32 dynamic payload types. The dynamic range (96-127) allows 32 RTP Payload Types. RFC 3551 section 3 allows to reassign other ranges. Consequently, when the dynamic range is exhausted, you can go for "rtp_pt_dynamic = 35" (or 0) in asterisk.conf. This enables the range 35-63 (or 0-63) giving room for another 29 (or 64) payload types. ASTERISK-26311 #close Change-Id: I7bc96ab764bc30098a178b841cbf7146f9d64964 (cherry picked from commit 9ac53877f688c06acaa7c377f15da8770e4ee88b) 2016-11-02 09:15 +0000 [d971647949] Joshua Colp * app_dial: Fix incorrect device state when channel is picked up. Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f 2016-11-01 13:13 +0000 [afecb2cfc0] Richard Mudgett * bundled pjproject: Fix DNS write to freed memory. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patch below fixes a write to freed memory under cartain DNS lookup conditions. 0006-r5477-svn-backport-Fix-DNS-write-on-freed-memory.patch ASTERISK-26516 Reported by: Richard Mudgett Change-Id: Ifdfae9ecf1e41b53080f33aab44ce1a220f349c5 2016-11-01 06:56 +0000 [5f188bb7a8] Joshua Colp * res_pjsip_sdp_rtp: Limit number of formats to defined maximum. The res_pjsip_sdp_rtp module did not restrict the number of formats added to a media stream in the SDP to the defined limit. If allow=all was used with additional loaded codecs this could result in the next media stream being overwritten some. This change restricts the module to limit it to the defined maximum and also increases the maximum in our bundled pjproject. ASTERISK-26541 #close Change-Id: I0dc5f59d3891246cafa2f3df5ec406f088559ee8 2016-11-01 04:18 +0000 [94c9496ed5] Tzafrir Cohen * netsock.c: fix includes for HURD ASTERISK-25070 Change-Id: I43bf94d2d36d3d8a8d0df40cd6c027d65a462814 2016-11-01 04:00 +0000 [c1c9487375] Tzafrir Cohen * define PATH_MAX for HURD PATH_MAX is not guaranteed to be defined. In parctice, all but the HURD define it to a constant. It is indeed not safe to assume there won't be longer paths and Asterisk generally does err safely on such cases. So even for HURD we'll just pretend PATH_MAX is 4096. ASTERISK-25070 #close Change-Id: I53d10ba18c34c132bcb640a5fd8e0da1d9b22db3 2016-10-31 17:35 +0000 [50fa868ab8] Kevin Harwell * codecs.conf.sample: Add sample and option descriptions for codec_opus codecs.conf.sample was missing codec opus's configuration options, descriptions, and examples. This patch adds the configuration options and examples to codecs.conf.sample that can be used with codec_opus. ASTERISK-26538 #close Change-Id: I1d89bb5e01d3e3b5bd78951b8dd0ff077a83dc8b 2016-11-01 08:32 +0000 [b3f10b7b94] Grachev Sergey * chan_sip: Incorrect display option Outbound reg. retry 403 If in sip.conf (general section) set option register_retry_403=no, the command "sip show settings" return value: Outbound reg. retry 403:0 If in sip.conf (general section) set option register_retry_403=yes, the command "sip show settings" return value: Outbound reg. retry 403:-1 * In static char "sip show settings" for "Outbound.reg. retry 403" option use AST_CLI_YESNO ASTERISK-26476 #close Change-Id: I3c14272f05f1067bd2aeaa8b3ef9cf8fcb12dcf9 2016-10-20 07:27 +0000 [29692d4aa4] Matt Jordan * res/stasis: Add CLI commands for displaying/debugging ARI apps This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5 2016-10-31 16:12 +0000 [a36a7d0cf4] gtjoseph * pjproject_bundled: Fix compile of pjsua so it handles audio In order for pjsua and its python binding to actually negotiate audio for the testsuite tests, it needs g711 and resample. The pj* libraries themselves do not. Unfortunately, pjproject relies on a brand new libresample that most distros don't ship so we need to use the libresample already bundled with pjproject. Only the pjsua executable and the _pjsua.so python library are linked with it so it shouldn't interfere with asterisk itself. Also it was pointed out that apply_patches couldn't handle multiple patches that depended on each other during the dry-run, so the dry-run was removed. Change-Id: I24f397462b486dcdde0dcafe40e6c55a6593f098 2016-10-31 13:46 +0000 [42bd70b29f] Etienne Lessard * manager: Add documentation for NewConnectedLine event. The NewConnectedLine event has been added by commit fe7671f, but the documentation was missing. ASTERISK-26537 #close Change-Id: I7fc331f18caa28492da9303e576f70884ca8c9e6 2016-10-30 13:33 +0000 [30b1bc77d2] Corey Farrell * vector: Prevent NULL argument to memcpy. Headers declare that memcpy does not accept NULL argument for the first two parameters. Add a conditional block to prevent memcpy and ast_free from running on vectors with NULL element array. ASTERISK-26526 #close Change-Id: I988a476bb5fcfcbd3f6d6c6b3e7769e4f9629b71 2016-10-29 10:31 +0000 [b96f18560b] Corey Farrell * astobj2: Declare private variable data_size for AO2_DEBUG only. Every ao2 object contains storage for a private variable data_size, though the value is never read if AO2_DEBUG is disabled. This change makes the variable conditional, reducing memory usage. ASTERISK-26524 #close Change-Id: If859929e507676ebc58b0f84247a4231e11da07f 2016-10-28 16:59 +0000 [6b1c55dc9b] gtjoseph * pjproject_bundled: Fix issue where "/version.mak" wasn't found main/Makefile includes third-party/pjproject/build.mak but doesn't set PJDIR beforehand so "include $(PJDIR)/version.mak" evaluates to "/version.mak". Fix is to set PJDIR in main/Makefile before the include. Change-Id: I0f7c67d60209049056fe9c4b041bf0463aa95604 2016-10-28 14:55 +0000 [d7f457e4c1] Richard Mudgett * bundled pjproject: Crashes while resolving DNS names. PJPROJECT 2.5.5 introduced a race condition with the -r5349 IPv6 DNS patch. The patches below fix the DNS lookup race condition crash caused by attempting to send the same message twice for the single DNS lookup. 0006-r5471-svn-backport-Various-fixes-for-DNS-IPv6.patch 0006-r5473-svn-backport-Fix-pending-query.patch The patch below removes a cached DNS response from the hash table when another thread is referencing the old entry. The table still contained the entry when it was destroyed which can result in inexplicable crashes. 0006-r5475-svn-backport-Remove-DNS-cache-entry.patch ASTERISK-26344 #close Reported by: Ian Gilmour ASTERISK-26387 #close Reported by: Harley Peters Change-Id: I17fde80359e66f65a91341ceca58d914d0f61cc4 2016-10-28 09:50 +0000 [87903a6848] Rusty Newton * SAC documentation: don't specify transports for endpoints and registrations Removing explicit transport definition for endpoints and registrations. It isn't necessary and isn't generally advised. ASTERISK-26514 #close Change-Id: Ifdec5e631962438a4683600968dfa4bfd15909fb 2016-10-27 21:49 +0000 [f373de3020] Corey Farrell * Fix shutdown crash caused by modules being left open. It is only safe to run ast_register_cleanup callbacks when all modules have been unloaded. Previously these callbacks were run during graceful shutdown, making it possible to crash during shutdown. ASTERISK-26513 #close Change-Id: Ibfa635bb688d1227ec54aa211d90d6bd45052e21 2016-10-26 18:48 +0000 [61a5c3460e] gtjoseph * pjproject_bundled: Remove usage of tar's --strip-components option Older versions of tar don't support the --strip-components option so instead of doing 'tar --strip-components=1 -C source', we now just untar to the tarball's root directory (pjproject-) and rename that directory to 'source'. Also fixed an issue where the pjproject source directory is a hard coded absolute pathname. ASTERISK-26510 #close ASTERISK-22480 #close Change-Id: I9ec92952507a91ff4e4d01e0149e09fd8e8f32b0 2016-10-27 08:07 +0000 [675c71ae8c] Joshua Colp * res_pjsip_caller_id: Fix crash on session timers UPDATE on inbound calls. The res_pjsip_caller_id module wrongly assumed that a saved From header would always exist on sessions. This is true until an inbound call is received and a session timer causes an UPDATE to be sent. In this case there will be no saved From header and a crash will occur. This change makes it fall back to the From header of the outgoing request if no saved From header is present. ASTERISK-26307 #close Change-Id: Iccc3bc8d243b5ede9b81abf960292930c908d4fa 2016-10-26 07:51 +0000 [14496ce1e5] Joshua Colp * app_voicemail: Clear voice mailbox in MailboxExists and MAILBOX_EXISTS. When executing the MailboxExists dialplan application and MAILBOX_EXISTS dialplan function the passed in temporary voice mailbox was not cleared, causing it to try to free garbage. ASTERISK-26503 #close Change-Id: Ie21ccfa1b80b9c59318e596f6b8e17da2b5a7cb3 2016-10-23 07:38 +0000 [e0bc17edff] Joshua Colp * pjsip: Fix a few media bugs with reinvites and asymmetric payloads. When channel format changes occurred as a result of an RTP re-negotiation the bridge was not informed this had happened. As a result the bridge technology was not re-evaluated and the channel may have been in a bridge technology that was incompatible with its formats. The bridge is now unbridged and the technology re-evaluated when this occurs. The chan_pjsip module also allowed asymmetric codecs for sending and receiving. This did not work with all devices and caused one way audio problems. The default has been changed to NOT do this but to match the sending codec to the receiving codec. For users who want asymmetric codecs an option has been added, asymmetric_rtp_codec, which will return chan_pjsip to the previous behavior. The codecs returned by the chan_pjsip module when queried by the bridge_native_rtp module were also not reflective of the actual negotiated codecs. The nativeformats are now returned as they reflect the actual negotiated codecs. ASTERISK-26423 #close Change-Id: I6ec88c6e3912f52c334f1a26983ccb8f267020dc 2016-10-26 06:32 +0000 [f534f67f52] Joshua Colp * res_pjsip_sdp_rtp: Fix address family of explicit media_address. When an explicit media_address is provided the address family in the SDP needs to be set to reflect it. ASTERISK-26309 Change-Id: Ib9350cc91c120eb2f96f0623d3907d12af67eb79 2016-10-25 11:20 +0000 [3a2092b722] gtjoseph * test_astobj2_thrash: Fix multithreaded issues The test uses 4 threads to grow, count, lookup and shrink 15K objects in a container. If there's only 1 execution engine available, the test will complete in <50ms. If each threads gets its own execution engine, the test may timeout after 60 seconds because the count thread does a locked ao2_callback on the whole container in a tight loop with only a sched_yield to give up time. The lock contention makes the test execution times wildly variable and mostly timeout. 2 execution engines are OK, 3 results in about 33% failure rate and >=4 causes a 80% failure rate. To fix, the sched_yield was changed to a usleep(500). Also, the number of buckets specified for the container was an even number so that was changed to the next prime number greater than (MAX_HASH_ENTRIES / 100). That's 151 currently. Change-Id: I50cd2344161ea61bfe4b96d2a29a6ccf88385c77 2016-10-24 14:13 +0000 [640203802e] Pascal Cadotte Michaud * typo: s/paranthesis/parenthesis/ in a comment Change-Id: I7c1f4eb051177ee22cbe97e063d4a3effe29be30 2016-10-24 10:55 +0000 [9b3557e054] gtjoseph * pjproject_bundled: Fixed various build issues * CFLAGS is now properly set when using older gcc. * All third-party pjproject targets have been removed. This fixes an issue with older libsrtp in some distros. * Manually removing the source directory now causes a rebuild. * EXTERNALS_CACHE_DIR is now properly checked. * Whitespace fixes. Change-Id: I98fec6847efc5602a9f41cb95096fd660a49fa60 2016-09-19 06:13 +0000 [bb982480d8] Joshua Colp * pjsip: Support dual stack automatically. This change adds support for dual stack automatically. No configuration is required and the IP address and version in the SIP messages and SDP will be automatically changed based on the transport over which the message is being sent. RTP usage has also been changed to listen on both IPv4 and IPv6 simultaneously to allow media to flow, and to allow ICE support on both simultaneously. This also allows failover between IPv6 and IPv4 to work as expected. ASTERISK-26309 #close Change-Id: I235a421d8f9a326606d861b449fa6fe3a030572d 2016-10-17 14:18 +0000 [eff97808fb] Mark Michelson * ARI: Detect duplicate channel IDs ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06 2016-10-19 17:53 +0000 [c2036c827c] snuffy * Fix issue with CLI not returning to prompt after running "features show" ASTERISK-26444 #close Change-Id: I91d645b7e6e5dba35f8c410df2be77a8c0e3acb8 2016-10-04 18:24 +0000 [3c62b60e56] Michael Walton * res_rtp_asterisk: Add ice_blacklist option Introduces ice_blacklist configuration in rtp.conf. Subnets listed in the form ice_blacklist = , e.g. ice_blacklist = 192.168.1.0/255.255.255.0, are excluded from ICE host, srflx and relay discovery. This is useful for optimizing the ICE process where a system has multiple host address ranges and/or physical interfaces and certain of them are not expected to be used for RTP. Multiple ice_blacklist configuration lines may be used. If left unconfigured, all discovered host addresses are used, as per previous behavior. Documention in rtp.conf.sample. ASTERISK-26418 #close Change-Id: Ibee88f80d7693874fda1cceaef94a03bd86012c9 2016-10-18 16:30 +0000 [012fda29d2] Mark Michelson * CDR: Alter destruction pattern for CDR chains. CDRs form chains. When the root of the chain is destroyed, it then unreferences the next CDR in the chain. That CDR is destroyed, and it then unreferences the next CDR in the chain. This repeats until the end of the chain is reached. While this typically does not cause any sort of problems, it is possible in strange scenarios for the CDR chain to grow way longer than expected. In such a scenario, the destruction pattern can result in a stack overflow. This patch fixes the problem by switching from a recursive pattern to an iterative pattern for destruction. When the root CDR is destroyed, it is responsible for iterating over the rest of the CDRs and unreferencing each one. Other CDRs in the chain, since they are not the root, will simply destroy themselves and be done. This causes the stack depth not to increase. ASTERISK-26421 #close Reported by Andrew Nagy Change-Id: I3ca90c2b8051f3b7ead2e0e43f60d2c18fb204b8 2016-10-18 09:04 +0000 [6d462b9eaf] Alexei Gradinari * chan_pjsip: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk. This patch uses functions pjsip_inv_add_ref/pjsip_inv_dec_ref to inform pjproject that an INVITE session is in use. ASTERISK-26482 #close Change-Id: Ia2e3e2f75358cdb530252a9ce158af3d5d9fdf33 2016-10-18 03:01 +0000 [662b560c35] Alexander Traud * cli: Auto-complete File not Module for core set debug. Since Asterisk 1.8, the command "core set debug" on the command-line interface asks not for a file (.c) but a module name. This change shows modules (.so) on the auto-completion via a tabulator or the question mark. Now, when you partially type a module name, TAB or ?, you get the correct candidiates. ASTERISK-26480 Change-Id: I1213f1dd409bd4ff8de08ad80cb0c73cafb1bae0 2016-09-11 10:13 +0000 [6f5880913f] Tzafrir Cohen * menuselect: invalid test for GTK2 configuire.ac was only checking for the existence of pkg-config and not the gtk2 package itself. Now it calls AST_PKG_CONFIG_CHECK for gtk+-2.0. ASTERISK-26356 #close Change-Id: I8079d515d6ea99f9ab320a7eaa71c2aaa101ccd5 2016-10-17 11:39 +0000 [546ec4b038] gtjoseph * pjproject_bundled: Add patch to address SSL crash Addresses crashes when an attempt is made to operate on an SSL socket after the socket has been closed. ASTERISK-26477 #close Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 2016-10-13 02:06 +0000 [644fad7477] Moises Silva * chan_rtp: Set a sane default rtp engine for unicast. ASTERISK-26439 Change-Id: I7f5ee2eeba8906e9ecb3293dbe3a747770bb5011 2016-10-15 20:05 +0000 [42cfdcd1b7] Matt Jordan * res/ari: Add the Asterisk EID field to outgoing events This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d 2016-10-16 17:25 +0000 [74d9385273] gtjoseph * utils.c: Fix ast_set_default_eid for multiple platforms ast_set_default_eid was searching for ethX, emX, enoX, ensX and even pciD#U interface names. While this was a good attempt, it wasn't inclusive enough to capture interfaces like enp6s0 or ens6d1, etc. Rather than relying on interface names, we now simply find the first interface returned by the OS that has a hardware address and that address isn't all 0x00 or all 0xff. The code IS different for BSD, Solaris and Linux based on what method is available for enumerating interfaces. Tested on: FreeBSD9 CentOS6 Ubuntu14 Fedora24 I was unable to test on Solaris at this time but the code for Solaris is used elsewhere at Digium. Change-Id: Iaa6db87ca78a9a375e47d70e043ae08c1448cb72 2016-10-15 04:58 +0000 [f1fd873df0] Michael Kuron * chan_sip: Only send video on outgoing channel if incoming channel supports it Previously, the settings videosupport=always and videosupport=yes behaved identically and unconditionally caused a video offer to be sent in the SDP on an outgoing call. This was a regression introduced with commit 5a1d90e1fbfc4b48927aad55311f3b38efbf1f54 in Asterisk 1.6.1. This commit restores correct behavior: videosupport=always causes a video offer to be sent unconditionally, while videosupport=yes will only offer video on an outbound channel if the incoming channel it is bridged to also supports video. That way, the device receiving the outgoing call can display the correct user interface elements for audio or video and will not unnecessarily show a blank video window on an audio-only call. ASTERISK-17470 #close Change-Id: I782f4409d436114dbc97061c3570c0cd24f7c3ae 2016-10-13 14:09 +0000 [0306869399] Leandro Dardini * app_queue: Added initialization for "context" parameter When using Asterisk Realtime Architecture, empty fields are skipped and the default values are used. If the "context" parameter in queue was set and then cleared from the database, the old value remains in memory and it continues to be used. This change initialize the "context" parameter with an empty value, allowing clearing the parameter. ASTERISK-26462 #close Change-Id: I64be73d5044ce38dd02408bd0e53de965ef65905 2016-10-14 00:18 +0000 [ce4cfd2eca] Corey Farrell * Fix issues with bundled pjproject cached download. Previously when testing I had a preexisting makeopts in ASTTOPDIR. The ordering of configure.ac causes --with-externals-cache to be processed after third-party configure. In cases where the Asterisk clone is cleaned it would cause pjproject to be downloaded to /tmp. This moves processing of the externals cache and sounds cache to happen before third-party configure. This also addresses a possible issue with the third-party Makefile. If TMPDIR is set by the environment it would override the path given to --with-externals-cache. ASTERISK-26416 Change-Id: Ifab7f35bfcd5a31a31a3a4353cc26a68c8c6592d 2016-10-12 16:24 +0000 [3c54328c57] Richard Mudgett * Audit ast_json_pack() calls for needed UTF-8 checks. Added needed UTF-8 checks before constructing json objects in various files for strings obtained outside the system. In this case string values from a channel driver's peer and not from the user setting channel variables. * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json object construction. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096 2016-10-12 16:20 +0000 [7f8f125738] Richard Mudgett * json: Check party id name, number, subaddresses for UTF-8. * Updated unit test as ast_json_name_number() is now NULL tolerant. ASTERISK-26466 #close Reported by: Richard Mudgett Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6 2016-10-11 18:14 +0000 [9621c9bcbc] Richard Mudgett * json: Add UTF-8 check call. Since the json library does not make the check function public we recreate/copy the function in our interface module. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99 2016-10-12 17:42 +0000 [e4bb9f9a37] Richard Mudgett * aoc.c: Whitespace cleanup * In s_to_json() removed unnecessary ast_json_ref() to ast_json_null() when creating the type json object. The ref is a noop. Change-Id: I2be8b836876fc2e34a27c161f8b1c53b58a3889a 2016-10-12 17:27 +0000 [bcac905bd3] Richard Mudgett * app_queue.c: Fix clearing of pause reason string. The pause reason is not always cleared when it should be cleared. * Made set_queue_member_pause() always clear pause reason if not pausing with a reason string. Change-Id: I993dad19626ec017478a230e980989438b778c53 2016-10-12 16:22 +0000 [ee4ae2b648] Richard Mudgett * app_minivm.c: Fix malformed ast_json_pack() call. Change-Id: I082b239022fac462666e52a14a44304748908dc0 2016-10-12 16:30 +0000 [90ae4e4337] gtjoseph * res_config_mysql: Fix several issues related to recent table changes Unlike any of the other database drivers, res_config_mysql checks that the table definition matches the requirements for every insert and update statement. Since all requirements are forced to 'char', any column that isn't a char, like ps_contacts' expiration_time, qualify_timeout, etc., will throw a warning. It's kinda harmless but very misleading. Since no other driver does those checks on insert or update, they've been removed from res_config_mysql. Also, all the logic that actually attempted to ALTER the table to fix the issue has been removed. With the move to alembic, the auto-alter functionality is not only unnecessary, it's also dangerous. The other issue is that res_config_mysql calls the mysql_insert_id function inside store_mysql. Presumably the intention was to return the number of rows inserted DESPITE A NOTE IN THE CODE THAT THE VALUE IS NON_PORTABLE AND MAY CHANGE. That value is then returned to config realtime as the number of rows inserted. Guess what? The value changed. It now only returns the number of rows inserted if there's an auto increment column on the table, which ps_contacts doesn't have. Otherwise it returns 0. So now, the insert worked but we tell config realtime and sorcery that no rows were inserted. That call to mysql_insert_id was removed and we now always return 1 if the insert succeeded. We're only inserting 1 row at a time anyway. If the insert fails, we still return -1. ASTERISK-26362 #close Reported-by: Carlos Chavez Change-Id: I83ce633efdb477b03c8399946994ee16fefceaf4 2016-09-29 13:08 +0000 [86c15db6a1] Torrey Searle * res_fax: Fix a tight race condition causing fax to crash in audio fallback When T.38 gets rejected and G711 failback occurs there is a period of time where neither AST_FAX_TECH_T38 nor AST_FAX_TECH_AUDIO is set, leading to a crash. Change-Id: Icc3f457b2292d48a9d7843dac0028347420cc982 2016-09-30 16:29 +0000 [29b7a5b00f] Rodrigo Ramírez Norambuena * Add text of cdr directory into README.md for ast-db-manage Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636 2016-10-06 09:58 +0000 [f919edc4e2] gtjoseph * app_dial: Add the "Q" option to set the cause on unanswered channels The "Q" option will set the cause on the unanswered channels when another channel answers. It overrides the default of ANSWERED_ELSEWHERE. NOTE: chan_sip does not support setting the cause on a CANCEL to anything other than ANSWERED_ELSEWHERE. ASTERISK-26446 #close Change-Id: I71742e0919aaa16784c30a2b2e73fbeed7672e47 2016-10-11 06:55 +0000 [a859bcb49c] Alexander Traud * chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. In the SIP channel driver chan_sip, auto_comedia was expected to be used in tandem with auto_force_rport. Or stated differently: Only when auto_force_rport was chosen (the default), auto_comedia worked. This change allows auto_comedia to be set independently of the state of (auto_)force_rport. For example, nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments when IPv6 clients are behind a Firewall. ASTERISK-26457 #close Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2 2016-10-10 16:59 +0000 [a884b26392] Badalyan Vyacheslav * vector: After remove element recheck index Small fix. It is necessary to double-check the index that we just removed because there is a new element. ASTERISK-26453 #close Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7 2016-09-29 12:52 +0000 [349c34f72a] Torrey Searle * res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge If a bridge switched to P2P when a DTMF was in progress it was possible for the DTMF to continue being sent indefinitely. Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29 2016-10-10 10:59 +0000 [9da3489d24] Badalyan Vyacheslav * res_pjsip_config_wizard: Memory leak in module_unload Fixed a memory leak. It removes only the first element. Added a useful feature in vector.h to remove all items under the CMP through a callback function / macro. ASTERISK-26453 #close Change-Id: I84508353463456d2495678f125738e20052da950 2016-10-09 21:53 +0000 [fa2885b3ff] Badalyan Vyacheslav * cel_odbc: Fix memory leak on module unload Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715 2016-10-03 11:30 +0000 [e6b0053d75] gtjoseph * bundled_pjproject: Add tests for programs used by the Makefile, et al. Added tests for bzip2, tar, patch, sed and nm to configure.ac. Set DOWNLOAD_TO_STDOUT to a working command line regardless of whether the download program is wget, curl or fetch. Added a 'configure.m4' file to the third-party directory which takes care of calling any third-party project setup. Had to move some pjproject_bundled stuff up in configure.ac so it was called before the third-party configure macro. The pjproject tarball is now downloaded to the externals_cache_dir if it was specified on the ./configure command line Removed regeneration of the pjproject aconfigure file. It was only needed for an old patch that no longer applies. Converted the tests for symbols to explicit tests since we know that they're now available in the bundled version. Saves a little time during configure. ASTERISK-26416 #close Reported-by: Corey Farrell Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b 2016-10-05 14:53 +0000 [0dc0356e39] gtjoseph * pjproject_bundled: Add MALLOC_DEBUG capability pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0 2016-10-07 17:32 +0000 [dd873bcada] Corey Farrell * astobj2: Add backtrace to log_bad_ao2. * Compile __ast_assert_failed unconditionally. * Use __ast_assert_failed to log messages from log_bad_ao2 * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751 2016-10-04 16:59 +0000 [86550f9c17] gtjoseph * alembic: Allow cdr, config and voicemail to exist in the same schema cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_'. Regardless, the script then continues with the migration using 'alembic_version_' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898 2016-10-05 04:25 +0000 [f166681c12] Alexander Traud * chan_sip: Honor support of Symmetric Response (rport) for SIP requests. In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6 2016-10-04 20:46 +0000 [430f6e5388] Michael Walton * audiohooks: Remove redundant codec translations when using audiohooks The main frame read and write handlers in main/channel.c don't use the optimum placement in the processing flow for calling audiohooks callbacks, as far as codec translation is concerned. This change places the audiohooks callback code: * After the channel read translation if the frame is not linear before the translation, thereby increasing the chance that the frame is linear as required by audiohooks * Before the channel write translation if the frame is linear at this point This prevents the audiohooks code from instantiating additional translation paths to/from linear where a linear frame format is already available, saving valuable CPU cycles ASTERISK-26419 Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f 2016-09-29 14:02 +0000 [2449d2877c] Kevin Harwell * Remove "format_ogg_opus: New format" This reverts commit 40aa28131bc30b4516da2b20eb1a1e043920169c. ASTERISK-26426 #close Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5 2016-09-27 16:10 +0000 [f0a2e628d6] gtjoseph * download_externals: Fix issue with re-install Needed to ignore an xmlstarlet return code for optional element. Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 Reported-by: Dan Jenkins 2016-09-22 09:49 +0000 [5258c067ae] gtjoseph * codec_opus: Add download ability to menuselect Updated codecs/codecs.xml to add codec_opus to the external download list. ASTERISK-26409 Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4 2016-07-23 14:50 +0000 [a5af8709c8] gtjoseph * codec_opus: Replace res_format_attr_opus with the one from codec_opus Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 (cherry picked from commit 59f7662a93bf9c07204fb50e1020a0f5bfbbd5c9) 2016-07-23 15:56 +0000 [44c0c51cf1] gtjoseph * format_ogg_opus: New format Add Ogg/Opus playback support. This uses libopusfile in order to be able to read .opus files and play them back. Writing/recording support is not present at this time. ASTERISK-26409 Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955 2016-09-24 19:05 +0000 [0ab443007b] gtjoseph * build_tools: Add ability to download variants to download_externals Some external packages have multiple variants that apply to different builds of asterisk. The DPMA for instance has a "bundled" variant that needs to be downloaded if asterisk was configured with --with-pjproject-bundled. There are 2 ways to specify variants: If you need the user to make the decision about which variant to download, simply create multiple menuselect "member" entries like so... external xmlstarlet bash no external xmlstarlet bash no Note that the second entry has "-" appended to the name. You can then use the existing menuselect facilities to restrict which members to enable or disable. Youy probably don't want the user to enable multiple at the same time. If you want to hide the details of the variants, the better way to do it is to create 1 member with "variant" elements. external xmlstarlet bash no The condition must be a bash expression suitable for use with an "if" statement. Any environment variable can be used plus those available in makeopts. In this case, if asterisk was configured with --with-pjproject-bundled the bundled variant will be automatically downloaded. Otherwise the normal version will be downloaded. Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e 2016-09-22 01:40 +0000 [a0a17a8c6f] Aaron An * channels/chan_pjsip: fix HANGUPCAUSE function bug. HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61 2016-09-23 09:54 +0000 [0502675e5c] Alessandro Crespi * chan_sip: Resolve externhost not to IPv6; instead go for IPv4. For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac 2016-09-20 09:42 +0000 [0056bcaebd] gtjoseph * chan_sip: Address runaway when realtime peers subscribe to mailboxes Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0 2016-09-21 15:03 +0000 [323aff3a09] Joshua Colp * core: Ensure presencestate subtype and message are NULL. When retrieving presence state information there is no guarantee that the subtype and message passed in are set to NULL. This change ensures they are. ASTERISK-26397 #close Change-Id: I61f8187972d5d8bbd7d6b7f4daa4f4f7e8237b23 2016-09-21 10:48 +0000 [10c180760c] Joshua Colp * res_odbc: Make pooling option deprecation notice more useful. This changes the notice for the deprecation of the old pooling options to point to the new option for doing pooling. This gives a clearer direction as to what to look into. ASTERISK-26389 #close Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10 2016-09-12 07:37 +0000 [42cc267016] Tzafrir Cohen * cdr_mysql: fix UTC support * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778 2016-09-21 08:46 +0000 [f16ab19292] Joshua Colp * odbc: Remove options that are no longer applicable. The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6 2016-09-20 15:17 +0000 [c9ce299b64] Corey Farrell * core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4 2016-09-20 10:05 +0000 [610eb4c189] Corey Farrell * logger: Fix default console settings. When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3 2016-06-27 14:26 +0000 [36092ee3a0] Tzafrir Cohen * sd_notify (systemd status notifications) support sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 (cherry picked from commit 07b95f7c65b7c083724f1af2b26f93cc22cad58c) 2016-09-19 14:21 +0000 [9372d32100] Walter Doekes * asterisk.c: Non-root users also get the astcanary after core restart. Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55 2016-09-19 09:40 +0000 [e96448e991] Walter Doekes * asterisk.c: When astcanary dies on linux, reset priority on all threads. Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39 2016-09-09 06:35 +0000 [01884a7af6] Timo Teräs * Fix showing of swap details when sysinfo() is available If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant || defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca 2016-09-12 18:00 +0000 [cdbad152c7] Richard Mudgett * res_config_odbc.c: Fix buffer size limitation creating invalid SQL. Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae 2016-09-14 08:42 +0000 [449719be00] Joshua Colp * res_pjsip_multihomed: Change Contact port to listening port. The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab 2016-09-07 14:48 +0000 [4d64b176eb] gtjoseph * pjproject_bundled: Prevent SERVFAIL from marking name server bad A name server that returns "Server Failure" is indicating only that the server couldn't process that particular request. We should NOT assume that the name server is incapable of serving other requests. Here's the scenario we've been encountering... * 2 local name servers configured in resolv.conf. * An OPTIONS request causes a request for A and AAAA records to go out to both nameservers. * The A responses both come back successfully resolved. * Because of an issue at some upstream nameserver, the AAAA responses for that particular query come back as "SERVFAIL" from both local name servers. * Both local servers are marked as bad and no further queries can be sent until the 60 second ttl expires. Only previously cached results can be used. * In this case, 60 seconds is just enough time for another OPTIONS request to go out to the same host so the cycle repeats. We could set the bad ttl really low but that also affects REFUSED and NOTAUTH which probably DO signal a real server issue. Besides, even a really low bad ttl would be an issue on a pbx. Although we use our own resolver in 14 and master and don't have this issue there, Teluu has merged this patch upstream so it's appropriate to cherry-pick to 14 and master to keep pjproject consistent. Change-Id: Ie03ba902288e274aff23f9b9bb2786e1e8be09e0 2016-09-14 07:59 +0000 [1cac856e17] Joshua Colp * rtp: Preserve timestamps on video frames. Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd 2016-09-14 09:51 +0000 [9df4056d70] Joshua Colp * res_pjsip_transport_management: Convert time in log message to seconds. ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc 2016-09-13 05:34 +0000 [98e42cc662] Steve Davies * chan_sip: Fix session timeout on retransmit of non-UDP packets Change-Id I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 Enable Session-Timers for SIP over TCP (and TLS) also disables SIP retransmits in chan_sip for non-UDP connections, allowing the TCP layer to handle the retransmits. Unfortunately, this caused sessions to be terminated with a retransmit timeout becasue it stopped at the point of the first retrans call. This patch waits for the 64*T1 timer to expire instead. ASTERISK-19968 Change-Id: I844f26801aada10bc94e9bebe6e151f0a8443204 2016-09-12 12:25 +0000 [0388882cdb] Richard Mudgett * app_queue: Fix CLI "queue show" and AMI Queues action output truncation. The output of CLI "queue show" and AMI Queues action is truncated and "failed to extend from 240 to 327" messages are generated if the queue member and interface names are lengthy. * Increase the string buffer size from 240 to 512 in order to accommodate for more information fields added to the output since v1.8. ASTERISK-26360 #close Reported by: Richard Mudgett Change-Id: Id99c03cf5362453b80491a4b3b0434cb67aa966d 2016-09-12 03:28 +0000 [da8ba990d1] Walter Doekes * chan_sip: Allow target refresh (Contact update) on re-INVITE. Previously, the Contact was stored only on initial INVITE and on any 18X and 200. That meant that after re-INVITEs from *us* the Contact could get updated, but after re-INVITEs from the *peer*, it did not. This changeset fixes this inconsistency, properly allowing target refreshes through re-INVITES (RFC3261, 12.2). If your strictrtp setting allows it, this change allows you to switch the source IP of a connected/calling device mid-call with a simple re-INVITE from the new IP. ASTERISK-26358 #close Change-Id: Ibb8512054ab27c8c3d2514022568fde943bf2435 2016-08-31 15:22 +0000 [e9ddab4685] Richard Mudgett * sip_to_pjsip.py: Map legacy_useroption_parsing. Map the sip.conf general section legacy_useroption_parsing to the new pjsip.conf global ignore_uri_user_options. ASTERISK-26316 Reported by: Kevin Harwell Change-Id: I78108a31995db19d41f4e1a07b3324692c5363fc 2016-08-29 18:08 +0000 [30af92e78d] Richard Mudgett * res_pjsip: Add ignore_uri_user_options option. This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62 2016-09-09 06:26 +0000 [7ed5dc2c58] Walter Doekes * contrib: Let safe_asterisk script continue without /dev/tty9. If you use the safe_asterisk script, it uses hardcoded defaults before running configurable values from /etc/asterisk/startup.d. The hardcoded default has TTY=9. Some containerized environments don't have such a TTY, and safe_asterisk would stop. The custom configuration from /etc/asterisk/startup.d/* isn't read until after it stopped, so changing TTY in a custom config did not help. This changeset changes safe_asterisk to continue if the TTY setting was untouched and /dev/tty9 and /dev/vc/9 aren't found. Change-Id: I2c7cdba549b77f418a0af4cb1227e8e6fe4148fc 2016-09-09 05:39 +0000 [7580a736bb] Joshua Colp * res_pjsip: Only invoke unidentified endpoint logic when unidentified. The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f 2016-08-23 06:35 +0000 [efcfc4c1ee] Corey Farrell (license 5909) * chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 2016-08-16 15:34 +0000 [f1ffc22933] Mark Michelson * res_pjsip: Do not crash on ACKs from unknown endpoints. The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703 2016-09-06 11:46 +0000 [23d6ec7417] Richard Mudgett * res_pjsip_messaging.c: Misc cleanups and fixes. * Eliminated RAII_VAR in get_outbound_endpoint(). * Simplify update_to() coding. However, this function can only be a NoOp because the To string can only be a URI and not a name-address formatted string. * Simplify update_from() coding. Also fixed a code path modifying the from string when the caller could still want to use the original string. * Fixed msg_data_create() incompletely removing the "pjsip:" to then add back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" case because it left the colon after pjsip in the string. Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db 2016-09-07 16:00 +0000 [5f19657710] Joshua Colp * res_pjsip: Allow global headers to be overridden. Currently when you add global headers from the dialplan both the header in the dialplan and the globally configured header are added to the resulting SIP INVITE. This change makes it so the headers in the dialplan take precedence and are the only ones added. Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad 2016-08-11 12:10 +0000 [206d4f57dc] Tzafrir Cohen * followme: initialize all config items on reload Some configuration directives were not initialized on reload, and hence were not reset to default if they were removed from followme.conf. ASTERISK-26288 #close Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150 2016-08-01 20:55 +0000 [117a7741c8] gtjoseph * build: Add download capability for external packages The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a 2016-09-06 02:41 +0000 [d04ae7d1d8] Walter Doekes * chan_sip: Don't refuse calls with "optional crypto"; fall back to RTP. Certain SNOM phones send so-called "optional crypto" in their SDP body. Regular SRTP setup looks like this: m=audio 64620 RTP/SAVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... SNOM-style "optional crypto" looks like this: m=audio 61438 RTP/AVP 8 0 9 99 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:... A crypto line is supplied, but the m-line does not have SAVP. When res_srtp.so is *not* loaded, then chan_sip.so treats the optional crypto as regular RTP, but when res_srtp.so *is* loaded, it refuses the incoming call with the following message: WARNING: process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio For platforms that want to start providing SRTP this presents a compatibility problem. This changeset lets chan_sip handle the SDP as if no crypto-line was supplied: i.e. accept the call as regular RTP, just like it did before res_srtp was loaded. Now you'll get this informative warning instead: WARNING: Ignoring crypto attribute in SDP because RTP transport is insecure ASTERISK-23989 #close Reported by: Olle Johansson Change-Id: I91a15ae05a0296e398d6b65f53bb11afde1d80e2 2016-09-03 16:04 +0000 [df3d0188e4] Matt Jordan * apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d 2016-09-03 16:02 +0000 [a64063cc97] Matt Jordan * apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358 2016-08-30 16:40 +0000 [03fc438f6e] Richard Mudgett * res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09 2016-08-29 18:06 +0000 [b5e753227d] Richard Mudgett * pjsip_configuration.c: Ignore repeated identify by methods. Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838 2016-08-30 17:26 +0000 [9b7501b6ad] Richard Mudgett * config_global.c: Comments and a default expression adjustment. Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3 2016-08-31 15:14 +0000 [3314e1cec2] Richard Mudgett * sip_to_pjsip.py: Map canreinvite as directmedia alias. Change-Id: I48b8e150f96a3d2a24d8fc25fbe4f5aff9f4a6b2 2016-08-31 15:37 +0000 [6372f40ba0] Richard Mudgett * sip_to_pjsip.py: Fix typo converting outboundproxy registration. Change-Id: I6f30e5f9fcf8469ba0079fbf884047d54c2c0b15 2016-08-31 15:13 +0000 [11eb1afd2d] Richard Mudgett * sip_to_pjsip.py: Fix comment typo and tabs. Change-Id: If35174614545727817d329c60ba4456c028941b5 2016-08-31 15:56 +0000 [0f9b144c1a] Richard Mudgett * Sample configs: Eliminate false multiline comment block starts. Change-Id: Ie627def9604ae30abd80754f9e6f09874825aec6 2016-09-02 11:36 +0000 [8d1c535bd6] Richard Mudgett * format_cap.c: Fix CLI "core show channeltype Surrogate" crash. * Make ast_format_cap_get_names() NULL tolerant. ASTERISK-26331 #close Reported by: CGI.NET Change-Id: Id67e93936dc8ec2a33a9d33655843d43b59285a3 2016-08-18 14:45 +0000 [9bca895469] Alexei Gradinari * res_pjsip_session: segfault on already disconnected session On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7 2016-08-10 15:14 +0000 [63feffa126] Mark Michelson * ConfBridge: Make some announcements asynchronous. Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a 2016-08-31 12:23 +0000 [a002a4d2db] Michael Kuron * app_mp3: Use correct buffer size and the same sample rate as the channel Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0 2016-08-26 10:39 +0000 [308a65fe6c] Alexei Gradinari * res_pjsip: qualify/unqualify added/deleted realtime endpoints If the PJSIP endpoint's AOR with the permanent contact was deleted from the realtime storage the res_pjsip module continues trying to qualify this contact. The error 'Unable to find an endpoint to qualify contact' appeares every 'qualify_frequency' seconds. This patch deletes this contact in this case. The PJSIP endpoint's AOR with the permanent contact is never qualified if it is added to realtime storage after asterisk started. This patch adds qualifying for the AOR's permanent contacts on the first handling of this AOR. ASTERISK-26319 #close Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe 2016-08-17 02:51 +0000 [2fa168348e] chris de rock * app_macro: Consider '~~s~~' as a macro start extension. As described in issue ASTERISK-26282 the AEL parser creates macros with extension '~~s~~'. app_macro searches only for extension 's' so the created extension cannot be found. with this patch app_macro searches for both extensions and performs the right extension. ASTERISK-26282 #close Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb 2016-08-29 07:10 +0000 [27951792c4] Etienne Lessard * pbx.c: Prevent infinite recursion in manager_show_dialplan_helper. Previously, if context A was including context B and context B was including context A, i.e. if there was a circular dependency between contexts, then calling manager_show_dialplan_helper could lead to an infinite recursion, resulting in a crash. This commit applies the same solution as the one implemented in the show_dialplan_helper function. The manager_show_dialplan_helper and show_dialplan_helper functions contain lots of code in common, but the former was missing the "infinite recursion avoidance" code. ASTERISK-26226 #close Change-Id: I1aea85133c21787226f4f8442253a93000aa0897 2016-08-26 14:34 +0000 [fb82fdb013] gtjoseph * pjproject_bundled: Disable srtp use by pjmedia The reason for the disable is that while Asterisk works fine with older libsrtp versions, newer versions of pjproject won't compile with them. Debian 6 for instance, has libsrtp 1.4.4 which is older than what pjproject is expecting. We don't use most of pjmedia but we DO use it for SDP negotiation. Luckily disabling srtp in pjmedia doesn't interfere with it's ability to negitiate a secure channel. The proper crypto attributes are negotiated in both directions. ASTERISK-26279 #close Change-Id: Id25a92cdf3df97a26c53cffae65b6b82de33c8e2 2016-08-26 08:41 +0000 [847bd47ff0] Alexander Traud * channel: No hung-up on failing security requirements. In your Diaplan, if you specify same => n,Set(CHANNEL(secure_bridge_media)=1) same => n,Set(CHANNEL(secure_bridge_signaling)=1) only the SIP channel driver chan_sip supports this. All other channels drivers like res_pjsip fail. In case of failure, the original sRTP source code released the whole channel, even if not hung-up, yet. This change does not release the channel but instead hangs-up the channel. ASTERISK-26306 Change-Id: I0489f0cb660fab6673b0db8af027d116e70a66db 2016-08-20 09:04 +0000 [b59d3b48d0] Alexander Traud * sip_to_pjsip: Migrate IPv4/IPv6 (Dual Stack) configurations. When using the migration script sip_to_pjsip.py, and your sip.conf is configured with bindaddr=::, two transports are written to pjsip.conf, one for 0.0.0.0 (IPv4) and one for [::] (IPv6). That way, PJProject listens on the IPv4 and IPv6 wildcards; a IPv4/IPv6 Dual Stack configuration on a single interface like in chan_sip. Furthermore, the script internal functions "build_host" and "split_hostport" did not parse Literal IPv6 addresses as expected (like [::1]:5060). This change makes sure, even such addresses are parsed correctly. ASTERISK-26309 Change-Id: Ia4799a0f80fc30c0550fc373efc207c3330aeb48 2016-08-25 07:06 +0000 [f69f5cd3c4] Joshua Colp * app_queue: Ensure member is removed from pending when hanging up. When dialing channels it is possible that they may not ever leave the not in use state (Local channels in particular) by the time we cancel them. If this occurs but we know they were dialed we explicitly remove them from the pending members container so that subsequent call attempts occur. ASTERISK-26299 #close Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65 2016-08-04 20:11 +0000 [5cd583d7a2] Richard Mudgett * res_pjsip: Cache global config options. We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7 2016-08-23 11:02 +0000 [8b4b2500ee] Richard Mudgett * res_fax: Fix deadlock in ast_channel_get_t38_state(). ast_channel_get_t38_state() calls ast_channel_queryoption() with AST_OPTION_T38_STATE. If the passed in channel is a local channel then a deadlock can happen if a channel lock is held when called. * Made ast_channel_get_t38_state() callers not hold a channel lock before calling. * Update ast_channel_get_t38_state() doxygen to note that no channel locks can be held when calling the function. ASTERISK-26203 #close Reported by: Etienne Lessard ASTERISK-24822 #close Reported by: David Brillert ASTERISK-22732 #close Reported by: Richard Mudgett Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214 2016-08-23 10:39 +0000 [e8d4f40022] Richard Mudgett * res_fax: Fix deadlock setting FAXMODE channel variable. ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1 2016-08-22 12:31 +0000 [35cf6c7702] Richard Mudgett * res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407 2016-08-23 11:16 +0000 [50b2aa506f] Richard Mudgett * res_fax.c: Add chan locked precondition comments. Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7 2016-08-23 10:42 +0000 [038cbc0215] Richard Mudgett * ast_framehook_detach() must be called with the channel locked. The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584 2016-08-22 15:01 +0000 [88e9d05ef7] Richard Mudgett * ast_framehook_attach() must be called with the channel locked. The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438 2016-08-24 14:42 +0000 [c9e83f6d0b] gtjoseph * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362 2016-08-19 18:19 +0000 [cb8fd610e2] Corey Farrell * Fix checks for allocation debugging. MALLOC_DEBUG should not be used to check if debugging is actually enabled, __AST_DEBUG_MALLOC should be used instead. MALLOC_DEBUG only indicates that debugging is requested, __AST_DEBUG_MALLOC indicates it is active. Change-Id: I3ce9cdb6ec91b74ee1302941328462231be1ea53 2016-08-10 15:14 +0000 [b8b5d52b5e] Mark Michelson * ConfBridge: Rework announcer channel methodology NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0 2016-08-23 05:54 +0000 [d5d7cbfcfb] Joshua Colp * Revert "ConfBridge: Rework announcer channel methodology" This reverts commit 0cdeb2bfb0f4203384c08858951af3c77be8b9b3. Change-Id: I18ba73b6d4dc0b994f4ffb01ae0b6cfad36ac636 2016-08-22 17:08 +0000 [c16ef02318] Mark Michelson * res_pjsip: Default endpoints to the "offline" status. A recent change attempted to optimize startup by not updating contact status. Instead, code responsible for qualifying contacts updates the status as it becomes known. The code even accounts for contacts/AORs that are not set to be qualified. The problem, though, is when there are no contacts associated with an endpoint. A common case is when an endpoint is set to register its contacts but has not done so yet. In this case, prior to registration, the endpoint's device state will appear to be "not in use" and hints associated with that device will appear to be "idle". In actuality, the device state and hint should both appear as "unavailable". The reason for the failure is that the optimization change made all persistent endpoint states set to "unknown". The fix here is to change the hard-coded "unknown" to be "offline" instead. The default state will be offline until the qualifying code determines that the contact is actually online. This way, if there are no contacts at all, then the state stays as offline, and device state and hints appear correctly. ASTERISK-26269 #close Reported by nappsoft Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a 2016-08-20 14:51 +0000 [e54dcf4fd5] David M. Lee * res_odbc_transaction: add dep on generic_odbc When res_odbc_transaction depended on res_odbc, it got the generic_odbc headers and libs implicitly. Now that it no longer depends on res_odbc, its dependency on generic_odbc must be explicit. Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911 2016-08-20 11:18 +0000 [be38c95def] Alexander Traud * pjproject_bundled: Allow IPv4/IPv6 (Dual Stack) configurations. PJProject supports a lot of platforms even Windows, some with different defaults when it comes to IPv6. In many Linux platforms like Ubuntu 16.04 LTS, "/proc/sys/net/ipv6/bindv6only" is set to 0 (false). Different than in Windows. Because of this, if configured with just an IPv6 address/transport, PJProject listens to both IPv4 and IPv6. However, this is not supported by the PJProject team. As consequence, you end-up with IPv4-mapped IPv6 addresses in SDP, incompatible with IPv4-only clients. Technically, you end-up with an IPv6-only server which accepts incoming connections on IPv4. If you try to configure two transports, one with IPv4 and one with IPv6 on the same interface, as expected by the PJProject team, the IPv4 transport is not able to bind because the IPv6 transport listens to both already. One solution would be to change "/proc/sys/net/ipv6/bindv6only" system-wide. Then, you are able to configure two transports, one for each IP version on the same interface. That way, you get a server which works with IPv4 clients and IPv6 clients at the same time over the same interface. Here, this change sets this parameter directly within PJProject to match the expectations of the PJProject team in any case. This allows IPv4/IPv6 Dual Stack servers out of the box like in chan_sip. This change was accepted by the PJProject team as and is expected to arrive in the next version, PJProject 2.6.0. Until then, this change is incorporated in the bundled PJProject of Asterisk. ASTERISK-26309 Change-Id: I3335d8718f79f4b2feae91b5b005a3ce684a63ae 2016-08-10 15:14 +0000 [0cdeb2bfb0] Mark Michelson * ConfBridge: Rework announcer channel methodology One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5 2016-08-19 10:21 +0000 [b494b9f88c] Alexei Gradinari * compilation failed with -Werror=maybe-uninitialized The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580 2016-08-19 03:59 +0000 [a628009eb9] Alexander Traud * sip_to_pjsip: Add cert_file. When using the migration script sip_to_pjsip.py, cert_file was not migrated to pjsip.conf. A previous change regarding this contained a copy/paste error. ASTERISK-22374 Change-Id: I0fa72e9412117d53b4284fc6b83fa5b2b95ba03b 2016-08-18 09:21 +0000 [b1fe070d0b] Alexander Traud * sip.conf: tlsclientmethod is using sslv23 as default. When 'tlsclientmethod' is not specified in sip.conf, chan_sip uses the OpenSSL SSLv23_method. This was documented incorrectly in the file sip.conf.sample. SSLv23_method got its name in the 90s. Today, with OpenSSL 1.0.2, this method enables (just) the secure TLSv1.0 and TLSv1.2. Or stated differently, that function should have been called 'secure_method' or 'automatic_method' back in the 90s. Consequently please, specify 'tlsclientmethod=tlsv1' in your sip.conf only if you face a server which has problems like not falling back to TLSv1.0 automatically. ASTERISK-24425 Change-Id: I502ce6146b4504cadfd3973af8d6ec3994f54fa3 2016-08-18 17:16 +0000 [ff2378c735] Kevin Harwell * rest-api: Swagger scripts were not replacing format variable in file brief Given resource paths did not have 'json' substituted in for the '{format}'. For some auto generated documentation/comment strings it resulted in something like the following: "... REST handler for /api-docs/sounds.{format}" This patch makes sure the resource api's path is properly substituted. ASTERISK-25472 #close Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23 2016-08-16 15:57 +0000 [43f400ef95] Jason Parker (license 4993) * res_format_attr_g729: Add annexb=no format parameter to SDPs Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0 2016-08-18 15:15 +0000 [4c1ae07d51] gtjoseph * res_odbc: Correct the dependency relationship with res_odbc_transaction The MODULEINFO dependencies between these 2 modules was reversed. res_odbc should depend on res_odbc_transaction, not the other way around. ASTERISK-25984 #close Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f 2016-08-18 12:04 +0000 [cab6975b02] Kevin Harwell * sip_to_pjsip: Set correct tls transport method A recent update had a copy/paste error where the unused variable 'val' was being passed to the set_value function instead of the 'method' value itself. This patch passes in the right variable. ASTERISK-22374 Change-Id: I895b7b3779ce4442bc58b8ec40d59dd29bb43f06 2016-08-18 08:19 +0000 [2381ddde63] Alexander Traud * sip_to_pjsip: Map the TLS method correctly. When using the migration script sip_to_pjsip.py and tlsclientmethod is not set in sip.conf, the default value of chan_sip (sslv23) is copied to pjsip.conf, to overwrite the default of the PJProject (tlsv1). This makes sure, res_pjsip is offering/using not just TLSv1.0 but TLSv1.2 as well. ASTERISK-22374 Change-Id: Ie530a3dae9926ae14f3920a21be1e2edb15bda4f 2016-08-18 08:17 +0000 [6500f5e138] Alexander Traud * sip_to_pjsip: Add compactheaders, timerb, timert1, and useragent. When using the migration script sip_to_pjsip.py, no section of type=system or type=general were created. Therefore the keys compactheaders, timerb, timert1, and useragent were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I318a453843227ea36bf130d392d4abd7bd26b5a1 2016-08-18 08:16 +0000 [21e9c69e56] Alexander Traud * sip_to_pjsip: Map (session-)timers correctly. When using the migration script sip_to_pjsip.py, session-timers=accept and session-timers=refuse were mapped to wrong values. ASTERISK-22374 Change-Id: Ie4e90d5f6a29aff07837b7fe5bc8aea5fb6fc092 2016-08-18 08:15 +0000 [c9a97398f7] Alexander Traud * sip_to_pjsip: Write username even without authname. When using the migration script sip_to_pjsip.py, now the (mandatory) username is written to pjsip.conf, even if there was no (optional) authname in the register string in sip.conf. ASTERISK-22374 Change-Id: Ie53e1997104cd2674821688b8a8247249f5e156f 2016-08-18 08:14 +0000 [60275359bc] Alexander Traud * sip_to_pjsip: Parse register even with transport. When using the migration script sip_to_pjsip.py and the register string started with a transport in sip.conf - like tls://... - register was not parsed correctly and therefore not migrated correctly to pjsip.conf. ASTERISK-22374 Change-Id: I44c12104eea2bd8558ada6d25d77edfecd92edd2 2016-08-18 08:13 +0000 [0d479232eb] Alexander Traud * sip_to_pjsip: Write local_net, contact_acl, contact_deny, and contact_permit. When using the migration script sip_to_pjsip.py, those keys got missing. These keys might appear several times and the function "merge_value" tried to collect those. However, because these keys have different names in sip.conf and pjsip.conf, "merge_value" was not able to find the new key name in sip.conf. This change lets "merge_value" search with the old key name in sip.conf and write with the new key name in pjsip.conf. ASTERISK-22374 Change-Id: Ie53c5278ae6f1cb8fa7e96c5289877d46981d9d2 2016-08-18 08:11 +0000 [cbc1b2d020] Alexander Traud * sip_to_pjsip: Map externhost/ip to Transports. When using the migration script sip_to_pjsip.py, the externhost or externip of sip.conf were erroneously written to Endpoints instead to Transports. ASTERISK-22374 Change-Id: I2c5873386cfc388899fa9cf2368639dd12f1b8e4 2016-08-18 08:04 +0000 [5f33e99534] Alexander Traud * sip_to_pjsip: Add defaultexpiry, maxexpiry, and minexpiry. When using the migration script sip_to_pjsip.py, defaultexpiry, maxexpiry, and minexpiry were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I007fbf543dcadc96fc3ed71c54da502bcb209b7b 2016-08-18 08:03 +0000 [231ea0350d] Alexander Traud * sip_to_pjsip: Write media_encryption. When using the migration script sip_to_pjsip.py, encryption=yes got missing and media_encryption=sdes was not written to pjsip.conf, because of a typo. ASTERISK-22374 Change-Id: I0fc3e55dc512a57603ae0fef41baacccf2a35c05 2016-08-18 08:02 +0000 [23eb065121] Alexander Traud * sip_to_pjsip: Write cos and tos. When using the migration script sip_to_pjsip.py, both tos_sip and cos_sip got missed, because of a typo. Therefore, cos and tos were not written to pjsip.conf. Furthermore, that revealed a misuse of an internal function, caused by a copy-and-paste error. ASTERISK-22374 Change-Id: Id245ebadf70ab9776eb280c026288540af3af5c2 2016-08-18 07:55 +0000 [0b675a208b] Alexander Traud * sip_to_pjsip: Add cert_file and ca_list_path. When using the migration script sip_to_pjsip.py, cert_file and ca_list_path were not migrated to pjsip.conf. ASTERISK-22374 Change-Id: I4612877d190b7f86a48698cefbf5c4db6c265825 2016-08-17 14:13 +0000 [1cd12d73a6] Richard Mudgett * res_pjsip_session.c: Fix unbound srv failover tests. Commit 1b666549f33d69dc080b212bf92126f3bc3a18b2 broke the srv failover functionality if a TCP connection gets disconnected. Under these conditions, session_inv_on_state_changed() gets a PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new transport. Unfortunately, session_inv_on_tsx_state_changed() also gets the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates the session. * Made session_inv_on_tsx_state_changed() complete terminating the session on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still PJSIP_INV_STATE_DISCONNECTED. ASTERISK-26305 #close Reported by: Richard Mudgett Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d 2016-08-16 15:36 +0000 [329507fe20] gtjoseph * res_pjsip: Add contact_user to endpoint contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 2016-08-17 08:10 +0000 [6f448f32fe] Torrey Searle * res_ari: Add http prefix to generated docs updated the uri handler to include the url prefix of the http server this enables res_ari to add it to the uris when generating docs Change-Id: I279335a2625261a8492206c37219698f42591c2e 2016-08-17 06:12 +0000 [56e0aed177] Alexander Traud * BuildSystem: Detect ca_list_path capabilities in external PJProject. Since Asterisk 13.8, pj_ssl_cert_load_from_files2 got detected only in the bundled PJProject but not in an external PJProject. Therefore, ca_list_path could not be used in pjsip.conf. With this change, pj_ssl_cert_load_from_files2 is detected again to enable ca_list_path again. ASTERISK-26303 #close Change-Id: I4a4a0cdc5cdff33730911fb4cfc0498c069043d0 2016-08-16 12:24 +0000 [2edcfcf1eb] gtjoseph * ari: Add documentation that path parameters are case-sensitive Added to api.wiki.mustache so that the generated object pages have the notation in the table header as well as under each method that has path parameters. ASTERISK-25492 #close Change-Id: I36c46c6dc0c9ac350470394a999a1b19ef3fcdaf 2016-08-15 15:29 +0000 [f4e28b3a09] Corey Farrell * Refactor usage pattern of xmldoc info tag. This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea 2016-08-04 20:00 +0000 [a8d9a53bae] Richard Mudgett * res_sorcery_config.c: Cleanup ao2 container usage idioms. Change-Id: Iad24b335fb121a2bc7f1d048ab7420569edcba5a 2016-08-04 15:57 +0000 [74a91b9ee5] Richard Mudgett * sorcery.c: Minor optimizations. * Remove some unused parameters from internal functions: sorcery_wizard_create() sorcery_wizard_update() sorcery_wizard_delete() * Created the struct sorcery_observer_invocation ao2 object without a lock since it is not needed in sorcery_observer_invocation_alloc(). * Cleanup generic ao2 container sorcery object id hash, sort, and cmp functions. Change-Id: Iff71d75f52bc1b8cee955456838c149faaa4f92e 2016-08-01 11:04 +0000 [29beb2890c] Richard Mudgett * sorcery.c: Tweak some container declaration formatting. * Tweak sorcery_object_type_alloc() formatting. * Tweak ast_sorcery_init() formatting. Change-Id: Ib02430023f15268cd7a2ea53f2c331213e4d3944 2016-08-11 23:30 +0000 [9b822293bd] Corey Farrell * pbx.c: Additional fixes to ast_context_remove_extension_callerid2. Do not check registrar of the first extension head. We should only check the registrar when we match the priority. Additionally fix a couple calls to strcmp which used the input callerid instead of the clean version ex.cidmatch. ASTERISK-26233 Change-Id: I17ea6881a18f40840ae9c1f5394aab1fbb3769f1 2016-08-10 14:41 +0000 [403c794684] Alexei Gradinari * core: Entity ID is not set or invalid The Exchanging Device and Mailbox States could not working if the Entity ID (EID) is not set manually and can't be obtained from ethernet interface. This patch replaces debug message to warning and addes missing description about option 'entityid' to asterisk.conf.sample. With this patch the asterisk also: (1) decline loading the modules which won't work without EID: res_corosync and res_pjsip_publish_asterisk. (2) warn if EID is empty on loading next modules: pbx_dundi, res_xmpp Starting with v197 systemd/udev will automatically assign "predictable" names for all local Ethernet interfaces. This patch also addes some new ethernet prefixes "eno" and "ens". ASTERISK-26164 #close Change-Id: I72d712f1ad5b6f64571bb179c5cb12461e7c58c6 2016-06-15 17:10 +0000 [93332cb1d0] Evgeniy Tsybra * chan_sip: Fix lastrtprx always updated Packets are read regulary, when there is no data in buffer fr->frametype is AST_FRAME_NULL. There was no check of frametype and lastrtprx always updated and, therefore, rtptimeout did not work at all. ASTERISK-25270 #close Change-Id: If3b5ca0dbb822582a86eb7d01dcae4e83448c41d 2016-08-15 07:17 +0000 [2735ec899a] Joshua Colp * manager: Clarify that dialplan manipulation actions are under system class. ASTERISK-26246 #close Change-Id: Id673b9786389f9d2a87f638ce1a25161f5f31657 2016-08-13 22:02 +0000 [f59bd47ed3] Matt Jordan * app_dial: Improve documentation * Add some helpful and other embedded paragraph tags * Document some of the lesser known channel variables set by Dial * Add examples for some common Dial uses, along with some more challenging but useful options Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1 2016-08-13 20:16 +0000 [4facaac408] Matt Jordan * manager: Add tags to relate interrelated events/actions together Change-Id: Idbac539205aa732bf786c4f765577d8e9ff28ba4 2016-08-13 20:15 +0000 [232d4fe24f] Matt Jordan * manager: Add tags to relate Bridge related events,actions, and apps Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85 2016-08-13 20:14 +0000 [63c0b2f7c9] Matt Jordan * manager: Add tags to relate AoC events and actions Change-Id: Iea89a36222712148c1775c05ed0ad1049d67a70e 2016-08-13 20:13 +0000 [0422667d6c] Matt Jordan * manager: Add tags to relate UserEvent actions/apps/events Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4 2016-08-12 15:53 +0000 [f9e734974b] Matt Jordan * res_agi: Improve documentation * Groups of AGI commands that have similar functionality now reference each other, and all reference the AGI application for ease of wiki reference. * The documentation for the AGI application has been improved, in particular noting the various AGI types and how they are invoked. * A warning message has been added to DeadAGI, noting that it is deprecated. Change-Id: I479ccdee8a7393f01b18692c3d4ab7e6bdd1875d 2016-08-12 13:53 +0000 [781bb410d0] Matt Jordan * manager: Add links between related events This patch adds some see-also references between related AMI events. It focuses primarily on those events that are guaranteed to come in pairs, such as DTMFBegin/DTMFEnd, as well as those that occur during the life cycle of an Asterisk channel, such as Newchannel/Hangup. Change-Id: Iaab600477052018d0f8c03d0c624c0856e9ff1f3 2016-08-12 11:15 +0000 [cfd6852d39] Matt Jordan * func_channel: Reorganize documentation * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b 2016-08-11 22:11 +0000 [cb043249b6] Corey Farrell * Run mandatory cleanup when startup fails. Errors during startup result in an exit. These error branches should be calling ast_run_atexit(0) to ensure mandatory cleanup is run. ASTERISK-26267 #close Change-Id: If226f2326ae2df7add20040696132214cf2bb680 2016-08-11 11:24 +0000 [4d5e96ab53] gtjoseph * res_pjsip_caller_id: Copy header name to short header name When compact_headers was set, we were sending a zero-length header name for PAI and RPID because we always forced the short header name length to 0. We did this because we cloned the header from "From" and wanted to clear "f" from the sname. By cloning however, we bypass pjproject's automatic logic that sets sname to name if there's no compact form of the header, which there isn't for PAI and RPID. So now we force sname to be the same as name right after we set name. res_pjsip_diversion needed the same treatment for the Diversion header. ASTERISK-26241 #close Change-Id: I633ec139630cd83809aae00336cee4a10077e467 2016-08-11 12:18 +0000 [143df33110] gtjoseph * res_pjsip: Fail global load if debug or default_from_user are empty If debug was specified in the global configuration but left blank, the logger would treat it as a wildcard and log all hosts. If default_from_user was empty, a crash would result. The global apply handler now checks for empty strings. ASTERISK-26239 #close ASTERISK-26238 #close Change-Id: Ie75727f5cd5808845d92cc81f5713842fb203336 2016-08-01 15:07 +0000 [1fc5c90014] Richard Mudgett * res_pjsip res_pjsip_mwi: Misc fixes and cleanups. * Eliminated RAII_VAR() usage in ast_sip_persistent_endpoint_update_state(). * Added a missing allocation failure check to persistent_endpoint_find_or_create(). * Made persistent_endpoint_find_or_create() create the new object without a lock as it isn't needed. * Cleaned up some ao2 container allocation idioms. * Reordered res_pjsip_mwi.c load_module() and unload_module() Change-Id: If8ce88fbd82a0c72a37a2388f74f77237a6a36a8 2016-08-04 18:03 +0000 [73052e5732] Richard Mudgett * location.c: Misc fixes and cleanups. * Eliminated most RAII_VAR() usage. * Added several missing allocation failure checks. * Made ast_sip_for_each_contact() allocate the wrapper ao2 object without a lock as it is not needed. Change-Id: Ie20913365156c95dd79e5d471cfd25e99ae880bc 2016-08-02 13:53 +0000 [9d4bd3d763] Richard Mudgett * taskprocessor.c: Tweak high water checks. * The high water check in ast_taskprocessor_alert_set_levels() would trigger immediately if the new high water level is zero and the queue was empty. * The high water check in taskprocessor_push() was off by one. Change-Id: I687729fb4efa6a0ba38ec9c1c133c4d407bc3d5d 2016-08-03 16:24 +0000 [e1248c3075] Richard Mudgett * res_pjsip: Make aor named lock a mutex. The named aor lock was always being locked for writes so a rwlock adds no benefit and may be slower because rwlocks are biased toward read locking. Change-Id: I8c5c2c780eb30ce5441832257beeb3506fd12b28 2016-07-29 17:41 +0000 [6e40334d89] Richard Mudgett * pjsip_distributor.c: Add missing allocation failure check. Change-Id: I932ab2cea845e534d9ff318035b6de39972d3b28 2016-08-11 11:13 +0000 [a3c5488ff4] Matt Jordan * app_queue: Prevent crash when a call is forwarded to an invalid location When a call forward attempt is made from a Queue member, the current code will hang up the forwarding channel in an off-nominal condition prior to raising the Stasis events informing the rest of Asterisk that the call was forwarded. This will result in a slew of dreaded FRACKs, most likely leading to a crash. This patch modifies the code such that we don't hang up the forwarding channel even in an off-nominal condition until we've safely raised the Stasis messages. ASTERISK-25797 #close Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38 2016-08-11 10:50 +0000 [5913929d31] Kevin Harwell * alembic: add auth_username to endpoint's identify_by enum A new identify_by option was added recently, auth_username. However, this setting was not added as an allowable choice in the database enumeration value. This patch updates the current enumeration, adding in the new setting. ASTERISK-26268 #close Change-Id: Ib4788e8485e4cd40172ec0abbf5810a147ab8bf8 2016-08-06 10:57 +0000 [1589452fdc] Alexei Gradinari * pjsip: Fix deadlock with suspend taskprocessor on masquerade If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b 2016-08-09 12:07 +0000 [f6ec94cca6] Kevin Harwell * alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b 2016-08-07 09:58 +0000 [5f815f9dba] Matt Jordan * channels/chan_pjsip: Add PJSIP_SEND_SESSION_REFRESH This patch adds a new PJSIP specific dialplan function, PJSIP_SEND_SESSION_REFRESH. When invoked on a PJSIP channel, the media session will be refreshed via either an UPDATE or re-INVITE request. When used in conjunction with the PJSIP_MEDIA_OFFER dialplan function, the formats in use on a PJSIP channel can be re-negotiated and changed dynamically after call setup. ASTERISK-26277 #close Change-Id: Ib98fe09ba889aafe26d58d32f0fd1323f8fd9b1b 2016-08-09 16:19 +0000 [a119bab6a6] Mark Michelson * res_rtp_asterisk: Cache local RTCP address. When an RTCP packet is sent or received, res_rtp_asterisk generates a Stasis event that contains the RTCP report as well as the local and remote addresses that the report pertains to. The addresses are determined using ast_find_ourip(). For the local address, this will typically result in a lookup of the hostname of the server, and then a DNS lookup of that hostname. If you do not have the host in /etc/hosts, then this results in a full DNS lookup, which can potentially block for some time. This is especially problematic when performing RTCP reads, since those are done on the same thread responsible for reading and writing media. This patch addresses the issue by performing a lookup of the local address when RTCP is allocated. We then use this cached local address for the Stasis events when necessary. ASTERISK-26280 #close Reported by Mark Michelson Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556 2016-08-08 12:53 +0000 [a06a1af0eb] Alexei Gradinari * res_pjsip_mwi: fix unsolicited mwi blocks PJSIP stack The PJSIP taskprocessors could be overflowed on startup if there are many (thousands) realtime endpoints configured with unsolicited mwi. The PJSIP stack could be totally unresponsive for a few minutes after boot completed. This patch creates a separate PJSIP serializers pool for mwi and makes unsolicited mwi use serializers from this pool. This patch also adds 2 new global options to tune taskprocessor alert levels: 'mwi_tps_queue_high' and 'mwi_tps_queue_low'. This patch also adds new global option 'mwi_disable_initial_unsolicited' to disable sending unsolicited mwi to all endpoints on startup. If disabled then unsolicited mwi will start processing on next endpoint's contact update. ASTERISK-26230 #close Change-Id: I4c8ecb82c249eb887930980a800c9f87f28f861a 2016-08-04 10:16 +0000 [485fd27f7c] Joshua Colp * res_pjsip_outbound_publish: Use a serializer shutdown group for unload. This change replaces the custom unload process for the outbound publish module with the common serializer shutdown group. ASTERISK-25217 #close Change-Id: I280a0384d860c486202d87d2d674394cca77ffb6 2016-08-03 15:39 +0000 [805f105f88] Corey Farrell * Add missing checks during startup. This ensures startup is canceled due to allocation failures from the following initializations. * channel.c: ast_channels_init * config_options.c: aco_init ASTERISK-26265 #close Change-Id: I911ed08fa2a3be35de55903e0225957bcdbe9611 2016-07-22 16:37 +0000 [ea71bd6e3e] Alexei Gradinari * app_voicemail: Add taskprocessor alert level options. On heavy loaded system with IMAP or DB storage, 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. It could happen when the IMAP or DB server dies or is unreachable. It could happen on startup when there are many (thousands) realtime endpoints configured with unsolicited mwi. If the taskprocessor queue reaches the high water level then the alert is triggered and pjsip stops processing new requests until the queue reaches the low water level to clear the alert. This patch adds 2 new 'general' configuration options to tune taskprocessor alert levels: 'tps_queue_high' - Taskprocessor high water alert trigger level. 'tps_queue_low' - Taskprocessor low water clear alert level ASTERISK-26229 #close Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8 2016-08-03 09:47 +0000 [9dc8cfabd5] Joshua Colp * astconfigparser: Really handle case where line is simply a comment. The regular expression would match causing the code that handled the line if it was merely a comment to never get executed. Change-Id: I3e4022481037ebcba9905587fe8c764b4ce21819 2016-07-23 08:51 +0000 [ad3e65433c] gtjoseph * asterisk.c: Add auto generation and persistence of UUID Upcoming features will require the generation and persistence of a UUID. Change-Id: I3ec0062427e133217db6ef496a4216f427c3b92d 2016-08-02 12:55 +0000 [efc4034d72] Kevin Harwell * rest-api: Code out of sync with the model Change-Id: Idccaa26fd4a423d47d013ee592b8fa6a0349c006 2016-07-29 13:13 +0000 [f6821fbaec] Mark Michelson * Remove SILK payload mappings from Asterisk core. SILK is a bit of a hog when it comes to using up our limited number of dynamic payload types in the RTP engine. By freeing up four slots, it allows for other codecs to potentially take the place. Now, codec_silk.so will dynamically use the payload slots in the RTP engine when it loads. A better fix would be make RTP dynamic payload types actually dynamic. However, at this stage of Asterisk 14 development, this is a risky move that would be imprudent. Change-Id: I5774e09408f9a203db189529eabdc0d3f4c1e612 (cherry picked from commit d50895c7b04036aeaad58990089399e46db4c817) 2016-08-01 11:08 +0000 [102d28c11a] Joshua Colp * sorcery: Use more compatible regex for local expressions. This changes the use of an empty regex for both res_sorcery_config and res_sorcery_memory to "." instead. This is a more compatible regular expression which also works on FreeBSD. ASTERISK-26206 #close Change-Id: Ia9166dd176f1597555ba22b6931180d0626c1388 2016-08-02 03:08 +0000 [b78d10a2df] Alexander Traud * res_pjsip: SIP/SDP origin (o=) contained square brackets on IP6 transports. ASTERISK-26256 #close Change-Id: I3fd68df561f81fdb8c6c497d465b50c12422f058 2016-08-01 16:13 +0000 [1f95c011c7] gtjoseph * menuselect: Add an opaque "member_data" string to the acceptable xml Change-Id: Id5ac43b95c8d7395f3be37f983632169db3d1afe 2016-07-27 09:56 +0000 [df42f64d62] David M. Lee * Replace strdupa with more portable ast_strdupa The strdupa function is a GNU extension, and not widely portable. We have an ast_strdupa function used within Asterisk which is preferred. I pulled the definition up from menuselect.c into the menuselect.h header file so it can be shared across menuselect. Change-Id: I9593c97f78386b47dc1e83201e80cb2f62b36c2e 2016-07-24 18:27 +0000 [56a07fbab9] gtjoseph * menuselect: Various menuselect enhancements * Add 'external' as a support level. * Add ability for module directories to add entries to the menu by adding members to the /.xml file. * Expand the description field to 3 lines in the ncurses implementation. * Allow the description field to wrap in the newt implementation. * Add description field to the gtk implementation. Change-Id: I7f9600a1984a42ce0696db574c1051bc9ad7c808 (cherry picked from commit 90f445729d5d86050d9d379485ff0a99f4a006c1) 2016-07-29 04:48 +0000 [7f9369c1b6] Joshua Colp * astconfigparser: Handle case where line is simply a comment. Change-Id: I2dea5815363f4d787d709228a04f33baee383ef5 2016-07-28 14:10 +0000 [57e9c66819] Corey Farrell * pbx.c: Fix handling of '-' in extension name and callerid This adds a two strings to ast_exten. name to go with exten and cidmatch_display to go with cidmatch. The new fields contain input used to add the extension in the first place. The existing fields now contain stripped input that excludes insignificant spaces and dashes. These stripped fields should always be used for comparisons. The unstripped fields should normally be used for display, but displaying stripped values will not cause runtime errors. Note the actual string is only stored twice if it contains dashes. If no dashes are found then both 'char *' fields point to the same memory. So this change has a minimum effect on memory usage. The existing functions ast_get_extension_name and ast_get_extension_cidmatch return unstripped values as they did before this change. Other similar bugs likely still exist where unstripped extensions are saved outside pbx.c then passed back in. ASTERISK-26233 #close Change-Id: I6cd61ce57acc1570ca6cc14960c4c3b0a9eb837f 2016-07-27 17:17 +0000 [873fc0fda5] Richard Mudgett * pbx.c: Allow dangerous functions when adding a hint to dialplan. We can allow dangerous functions when adding a hint since altering dialplan is itself a privileged activity. Otherwise, we could never execute dangerous functions. ASTERISK-25996 #close Reported by: Andrew Nagy Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba 2016-07-21 10:36 +0000 [f00525a6f6] Alexei Gradinari * pjproject: fixed a few bugs This patch fixes the issue in pjsip_tx_data_dec_ref() when tx_data_destroy can be called more than once, and checks if invalid value (e.g. NULL) is passed to. This patch updates array limit checks and docs in pjsip_evsub_register_pkg() and pjsip_endpt_add_capability(). Change-Id: I4c7a132b9664afaecbd6bf5ea4c951e43e273e40 2016-07-17 18:28 +0000 [972cee2e4c] gtjoseph * pjproject_bundled: Update for pjproject 2.5.5 Add more --disable-* switches to Makefile.rules including --disable-opus which was causing bundled pjproject to fail with "undefined reference" errors in libasteriskpj. Changed PJ_ENABLE_EXTRA_CHECK to 1. Removed 2 obsolete patches and added a new one. The new one was merged by Teluu on 6/27/2016. ASTERISK-26148 #close Change-Id: Ib8af6c6a9d31f7238ce65b336134c2efdc855063 2016-07-27 10:33 +0000 [8902a51d59] David M. Lee * Portably sscanf tv_usec In a timeval, tv_usec is defined as a suseconds_t, which could be different underlying types on different platforms. Instead of trying to scanf directly into the timeval, scanf into a long int, then copy that into the timeval. Change-Id: I29f22d049d3f7746b6c0cc23fbf4293bdaa5eb95 2016-07-27 12:36 +0000 [852e763571] Kevin Harwell * rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c 2016-07-21 22:44 +0000 [e8c34680ca] Richard Mudgett * dsp.c: Add fax and DTMF detection unit tests. * Add fax amplitude and frequency sweep tests. * Add DTMF amplitude and twist unit tests. Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7 2016-07-21 11:56 +0000 [c1f240b818] Richard Mudgett * dsp.c: Added descriptive comments to Goertzel calculations. * Added doxygen to describe some struct members and what is going on in the code. Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d 2016-07-13 13:48 +0000 [003a52fd62] Richard Mudgett * dsp.c: Fix incorrect format reference typo. Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896 2016-07-25 21:18 +0000 [4c0a0cbe02] Richard Mudgett * dsp.c: Correct DTMF twist dsp.conf documentation. Change-Id: Idf97e3a72f1edc5fca58f2fa7b20785922be0cae 2016-07-22 04:43 +0000 [87433c2566] Joshua Colp * astconfigparser.py: Update with realtime fixes. When configuring SIP URIs in the pjsip.conf file it is necessary to escape the semicolon so the parser does not treat it as a comment. This change allows this to work in the astconfigparser implementation. A secondary bug where some data was lost if a configuration option included a "=" in its value was also fixed. A bug where sections would be considered equal despite being different has also been fixed. Change-Id: If229f656ef22050b50e7b34e90c4bffe796431f8 2016-07-21 22:28 +0000 [159e437e5a] Richard Mudgett * dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc 2016-07-22 14:44 +0000 [eda95236d1] Mark Michelson * Fix sqlalchemy error regarding identifier length. sqlalchemy was complaining: sqlalchemy.exc.IdentifierError: Identifier 'ps_contacts_qualifyfreq_exptime' exceeds maximum length of 30 characters This fixes the problem by changing the index name to be "ps_contacts_qualifyfreq_exp" instead. ASTERISK-26227 #close Reported by Mark Michelson Change-Id: I0ed784f87504be2a59ee8d3242ef6f625d5ed1a9 2016-07-22 05:46 +0000 [66c9dfb272] Alexander Traud * chan_sip: Enable Session-Timers for SIP over TCP (and TLS). Asterisk defaults to timers=accept/refresher=uas. In that scenario, only in that scenario, Sessions-Timers (RFC 4028) had no effect via TCP. This change enables Session-Timers for SIP over TCP (and for SIP over TLS). However with longer international calls via TCP, the SIP channel might break, because all hops on the Internet route must stay online (have not a single power outage, for example). Therefore with Session-Timers enabled (which are enabled at default), you might see dropped calls. Consequently even with this change, you might be better-off going for session-timers=refuse in your sip.conf. ASTERISK-19968 #close Change-Id: I1cd33453c77c56c8e1394cd60a6f17bb61c1d957 2016-07-15 16:16 +0000 [33716106e0] Richard Mudgett * res_pjsip: Whitespace and comment cleanup. Change-Id: I11139a4a95df34e223ba622aa6227e33ab8f6c38 2016-07-21 09:05 +0000 [52ab0bf258] gtjoseph * chan_sip: Prevent deadlock when issuing "sip show channels" sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 2016-07-19 15:22 +0000 [5997ec7c9e] Alexei Gradinari * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a 2016-07-18 22:46 +0000 [7fdf7c3d4c] Corey Farrell * Add conditional support for noreturn functions. This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753 2016-07-19 13:18 +0000 [dcb8aa8c1c] Richard Mudgett * chan_dahdi.c: Fix deadlock potential in fax redirection. The dahdi_handle_dtmf() and my_handle_dtmf() have the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. ASTERISK-26216 #close Reported by: Richard Mudgett Change-Id: I9144b84ade5f96690996624ec8a2d40c56af40aa 2016-07-13 18:49 +0000 [fa91cf3eec] Richard Mudgett * chan_sip.c: Fix deadlock potential in fax redirection. The sip_read() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I6d3f5cccd4b77c3aa6ffc1a54c0f6bde61c9278e 2016-07-13 18:48 +0000 [2e1bdc3775] Richard Mudgett * chan_pjsip.c: Fix deadlock potential in fax redirection. The chan_pjsip_cng_tone_detected() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I32aecbb4818af646dc5a619f0dc040e9b1f222e5 2016-07-12 17:33 +0000 [628e8c91d5] Richard Mudgett * res_fax.c: Fix deadlock potential in FAXOPT(faxdetect) framehook. The fax_detect_framehook() has the potential to deadlock if an incoming fax happens during the Playback or similar application. * Fixed the potential deadlock by not calling ast_async_goto() with the channel lock held. * Made always eat the fax detection frame whether there is a fax extension or not. * Made only detach the framehook if we detected a fax and not on other possible frames. ASTERISK-26216 Reported by: Richard Mudgett Change-Id: I99da35c26d1cd802626ffb4c1b4eb5b015581b6d 2016-07-12 17:24 +0000 [676aeede36] Richard Mudgett * res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 2016-07-18 16:16 +0000 [652130feb2] Richard Mudgett * chan_dahdi: Add faxdetect_timeout option. The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 2016-07-15 20:44 +0000 [851b1c3a17] Richard Mudgett * res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d 2016-07-19 04:48 +0000 [021d4892cd] Alexander Traud * Makefile: Retain XML Declaration and DTD in docs. Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, the XML Declaration and DTD were overwritten by this. ASTERISK-26212 #close Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd 2016-07-18 18:39 +0000 [c8e41d14a1] Corey Farrell * Unit tests: Use AST_TEST_DEFINE in conditional code only. If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686 2016-07-18 05:13 +0000 [e404f51b42] Alexander Traud * res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets. With this change, the initial RTP sequence number is randomly chosen not between 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over counter (ROC) synchronization is not lost for sRTP, when the very first RTP packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 ASTERISK-26207 #close Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464 2016-07-18 04:14 +0000 [5f24874ebb] Alexander Traud * Makefile: Suppress echoing of target 'config' again. ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f 2016-07-14 03:25 +0000 [76d4983c15] Corey Farrell * features.c: Remove unneeded adsi.h include. adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59 2016-07-14 18:06 +0000 [cb58f853e1] Alexei Gradinari * res_pjsip_mwi: remove unneeded check on endpoint's contacts. The function create_mwi_subscriptions_for_endpoint checks if there is active contacts by retrieving aors and contacts. This function is used to create all unsolicited mwi subscriptions on startup and is used when contact added. In both cases it's not necessary to check if there are contacts. The contacts are needed when asterisk sends mwi. ASTERISK-26200 #close Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa 2016-06-30 15:58 +0000 [28501051b4] Mark Michelson * Update support for SILK format. This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e 2016-07-14 07:45 +0000 [43b5f8d57b] Richard Miller (license 5685) * app_queue: Only remove queue member from pending when state changes. It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0 2016-07-14 02:40 +0000 [a17b071e36] Corey Farrell * pbx: Fix leak of timezone for time based includes. Create include_free to run ast_destroy_timing and ast_free, use that in all places that freed an ast_include structure. This fixes a couple of paths that previously did not run ast_destroy_timing. ASTERISK-26196 #close Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838 2016-07-13 17:45 +0000 [8cef8f35e7] Kevin Harwell * translate: explicit format destination not properly set If the destination format's name differed from the codec name then the translator's explict_dst field would be improperly set. In some circumstances it would end up setting it to a newly created format that has the same name as the codec when it actually needed to be the given destination codec. This could cause the translation path to use the wrong format. For instance, if an endpoint had specified 'myulaw' as a format the translator could end up using a 'ulaw' format (with whatever/default settings) instead. If the format attribute settings differed between the two then there may unexpected results during processing. This patch removes the name check when building the translation path. This should make it always set the translator's explicit_dst to the given destination format as long as the sample rate and types match. Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5 2016-07-08 11:46 +0000 [afbd10b0c5] Richard Mudgett * stasis_endpoint.c: Fix contactstatus_to_json(). The roundtrip_usec json member is optional. If it isn't present then don't put it into the converted json structure where ast_json_pack() will choke on it. Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0 2016-07-13 13:45 +0000 [2be13d62fd] Corey Farrell * chan_sip: Fix reference leak in mwi_event_cb Cleanup the peer reference when stasis_subscription_final_message is true. Also free peer_name even if peer exists, after reload a new peer_name will be allocated. ASTERISK-26193 #close Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69 2016-06-22 07:13 +0000 [332beb27d8] Eugene Voityuk ,Alexander Traud * res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c 2016-07-13 11:30 +0000 [672a64bda3] Corey Farrell * threadpool: Fix leak in ast_threadpool_serializer_group error path. ast_threadpool_serializer_group leaks a reference to ser when listener is allocated but tps is not. Although listener takes the reference to ser cleanup functions are not run without tps. ASTERISK-26191 #close Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585 2016-07-11 10:22 +0000 [fea201f7e6] Richard Mudgett * pjsip_options.c: Fix container operation. aor_observer_deleted() needs to operate on all contacts found for the deleted AOR instead of only the first one found. This is really only a problem if there is more than one contact for the AOR. Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1 2016-07-11 10:21 +0000 [02877b4b4f] Richard Mudgett * pjsip_configuration.c: Misc cleanups. * Fix some whitespace in various routines. * Rename i to iter in persistent_endpoint_update_state(). * Fix off-nominal copy/paste message wording in persistent_endpoint_contact_deleted_observer() Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8 2016-07-13 08:57 +0000 [148cd1b319] Alexander Traud * BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf. Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. ASTERISK-26046 #close Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7 2016-07-11 10:25 +0000 [97b4c7a5b4] Richard Mudgett * res_pjsip: Fix statsd regression. The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1 2016-07-12 03:50 +0000 [3be6fa1e4b] Alexander Traud * BuildSystem: Allow own CFLAGS on ./configure. Before this change, make failed with the error Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH when CFLAGS were supplied to the configure script. This was introduced with which disabled BUILD_NATIVE when CFLAGS were supplied. Those who need different -march= values, please, go for ./configure make menuselect.makeopts or make menuselect ./menuselect/menuselect --disable BUILD_NATIVE ASTERISK-25289 #close Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc 2016-07-11 13:42 +0000 [5ee205d8bb] Richard Mudgett * ast_expr2: Fix off-nominal memory leak. Thanks to ibercom for pointing out a memory leak that was missed in the earlier patch for the issue. ASTERISK-26119 Reported by: Alexei Gradinari Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71 2016-07-11 10:17 +0000 [f5e9872016] Alexander Traud * install_prereq: Checkout of libSRTP 1.5.x. Since 5th November 2014, the master branch of libSRTP changed the prefix of several member names and is not compatible with the source code in Asterisk anymore. Therefore instead, this change checks out the latest version of the libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as backend. This makes AES-GCM and AES-IN possible. ASTERISK-22131 #close Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6 2016-07-10 19:08 +0000 [17efed6cf7] Joshua Colp * func_odbc: Fix connection deadlock. The func_odbc module was modified to ensure that the previous behavior of using a single database connection was maintained. This was done by getting a single database connection and holding on to it. With the new multiple connection support in res_odbc this will actually starve every other thread from getting access to the database as it also maintains the previous behavior of having only a single database connection. This change disables the func_odbc specific behavior if the res_odbc module is running with only a single database connection active. The connection is only kept for the duration of the request. ASTERISK-26177 #close Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f 2016-07-09 13:32 +0000 [06ba533bc7] Corey Farrell * chan_sip: Fix reference leaks in error paths. * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. * build_peer leaks peer on failure to allocate the endpoint. This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed with an unref in the appropriate place. ASTERISK-26184 #close Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12 2016-07-07 12:41 +0000 [9d4e664f62] Corey Farrell * REF_DEBUG: Prevent logging of container node objects. Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being recorded to the refs log for the node being replaced. This prevents logging of those unrefs since they would produce errors in refcounter.py. ASTERISK-26181 #close Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4 2016-07-07 10:55 +0000 [e26bd15e7a] Scott Griepentrog * PJSIP: provide valid tcp nodelay option for reuse When using TCP transport with chan_pjsip, the TCP_NODELAY option value was allocated on the stack, then passed as a pointer to the tcp transport configuration structure, and later re-used on subsequently created sockets when it was no longer valid. This patch changes the allocation to be a static. ASTERISK-26180 #close Reported by: Scott Griepentrog Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0 2016-07-07 10:38 +0000 [77b0145a25] Joshua Colp * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c 2016-07-04 16:38 +0000 [b4a9fa2c9e] Alexei Gradinari * res_sorcery_realtime: fix bug when successful UPDATE is treated as failed If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. This value should be treated as success. But the function sorcery_realtime_update treats it as failed. This bug was found using stress tests on PJSIP. If there are 2 consecutive SIP REGISTER requests with the same contact data during 1 second then res_pjsip_registrar adds contact location on 1st request and tries to update contact location on 2nd. The update fails and res_pjsip_registrar even removes correct contact location. The test "object_update_uncreated" was removed from test_sorcery_realtime.c because it's now a valid situation. This patch also adds missing debug of extra SQL parameter. ASTERISK-26172 #close Change-Id: I05a7f3051455336c9dda29efc229decf86071303 2016-06-24 19:55 +0000 [1dfd3fc995] Matt Jordan * res/res_pjsip_session: Check for presence of an active negotiator It is possible in a hypothetical situation for a session refresh to be invoked on a PJSIP when the negotiatior on the INVITE session has not yet been established. While this shouldn't occur with existing uses of ast_sip_session_refresh, the crashes that occur due to improperly calling PJSIP functions that expect a non-NULL negotiatior are avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this means that simply checking for the presence of the negotiator before passing it to other PJSIP functions that use it is allowable. As such, this patch adds checks for the presence of the negotiator before calling PJSIP functions that assume it is non-NULL. Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d 2015-10-19 18:55 +0000 [9dd0aeeb44] Matt Jordan * res/res_pjsip_pubsub: Add additional debug statements When something very sad and wrong occurs, it's challenging sometimes to figure out why. This patch adds some additional debug statements on off-nominal paths to try and make debugging easier. Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640 2015-10-19 18:55 +0000 [1ec4f8dd00] Matt Jordan * res/res_corosync: Raise a Stasis message on node join/leave events When res_corosync detects that a node leaves or joins, it currently is informed of this via Corosync callbacks. However, there are a few limitations with the information presented: (1) While we have information that Corosync is aware of - such as the Corosync nodeid - that information is really only useful inside of Corosync or res_corosync. There's no way to translate a Corosync nodeid to some other internally useful unique identifier for the Asterisk instance that just joined or left the cluster. (2) While res_corosync is notified of the instance joining or leaving the cluster, it has no mechanism to inform the Asterisk core or other modules of this event. This limits the usefulness of res_corosync as a heartbeat mechanism for other modules. This patch addresses both issues. First, it adds the notion of a cluster discovery message both within the Stasis message bus, as well as the binary event messages that res_corosync uses to transmit data back and forth within the cluster. When Asterisk joins the cluster, it sends a discovery message to the other nodes in the cluster, which correlates the Corosync nodeid along with the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids to Asterisk EIDs, such that it can map changes in cluster state with the Asterisk instance that has that nodeid. Likewise, when an Asterisk instance receives a discovery message from a node in the cluster, it now sends its own discovery message back to the originating node with the local Asterisk EID. This lets Asterisk instances within the cluster build a complete picture of the other Asterisk instances within the cluster. Second, it publishes the discovery messages onto the Stasis message bus. Said messages are published whenever a node joins or leaves the cluster. Interested modules can subscribe for the ast_cluster_discovery_type() message under the ast_system_topic() and be notified when changes in cluster state occur. Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465 2016-07-04 13:54 +0000 [2c16a81dd5] Alexei Gradinari * res_pjsip: Added "subscribe_context" to endpoint If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514 2016-07-04 05:58 +0000 [a1bd57884d] Alexander Traud * BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is using AS_HELP_STRING everywhere else already. ASTERISK-26046 Change-Id: I8299faf504ceaeee3e39930c59293809e116c631 2016-06-30 15:17 +0000 [640fbbbe28] Richard Mudgett * features: Fix channel datastore access. Found as a result of the testsuite tests/callparking test crashing. Several calls to ast_get_chan_featuremap_config() and ast_get_chan_features_xfer_config() did not lock the channel before calling so the channel's datastore list was accessed without the lock's protection. Apparently another thread deleted a datastore on the channel's list while the crashing thread was walking the list. Crash at 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. * Add missing channel locks to calls that were not already protected as the doxygen for those calls indicates. Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1 2016-06-22 17:26 +0000 [359134c8d3] Richard Mudgett * res_pjsip_session.c: Don't send extra BYE if SDP invalid. When an answer SDP is invalid we were disconnecting the outgoing call and sending two BYE requests. The first BYE was sent by PJPROJECT because of the invalid SDP answer. The second BYE was sent by Asterisk because it thought the canceled call was the result of the RFC5407 section 3.1.2 race condition. * Made not send the BYE on a canceled session if the SDP negotiation is incomplete because PJPROJECT has already sent a BYE for the failed negotiation. ASTERISK-25772 #close Reported by: Dmitriy Serov Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836 2016-06-27 17:19 +0000 [5fabcf2ca1] Richard Mudgett * res_pjsip_session.c: End call on initial invalid SDP negotiation. When an incoming call defers SDP negotiation and then sends us an invalid SDP in the ACK, we need to send a BYE to disconnect the call. In this case SDP negotiation has failed and we don't have valid media streams negotiated. ASTERISK-25772 Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8 2016-06-23 15:13 +0000 [38a4e983dc] Richard Mudgett * res_pjsip.c: Register PJMEDIA error code decoder. Registering the PJMEDIA error codes allows errors found when parsing an incoming SDP to be easier to figure out. "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" is much easier to understand than "Unknown error 220030". ASTERISK-25772 Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0 2016-06-27 16:56 +0000 [1952434df5] Richard Mudgett * res_pjsip_session.c: Remove unused parameter from handle_incoming(). Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa 2016-06-22 18:02 +0000 [28928ba5c4] Richard Mudgett * res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047 2016-06-30 08:25 +0000 [43a78100c0] gtjoseph * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK from getting set when using an external pjproject. ASTERISK-26099 #close Reported-by: Ross Beer Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae 2016-06-29 15:31 +0000 [99eff80e76] Matt Jordan * hep.conf.sample: Default 'enabled' to 'no' Following the principle of least surprise, we should not be sending massive numbers of PJSIP and RTCP HEP packets out into the ether to some only-slightly-random IP address. Having 'enabled' set to 'no' in the sample configuration file should prevent this from happening for those who run 'make samples'. ASTERISK-26159 #close Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1 2016-06-29 15:09 +0000 [78960975f2] Matt Jordan * pjproject/patches/config_site: Increase the max number of ICE candidates When negotiating ICE candidates with WebRTC capable endpoints, many networks will result in a browser offering ICE candidates that exceeds the default number of max candidates, 16. This patch bumps the max candidates to 32, with the max checks at twice the number of candidates. In practice, this has shown to be sufficient for browser/WebRTC negotiation. Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5 2016-06-28 09:00 +0000 [d07c8a0504] gtjoseph * codecs: Fix ABI incompatibility created by adding format_name to ast_codec Adding format_name even to the end of ast_codec caused issued with binary codec modules because the pointer would be garbage in asterisk when they registered. So, the ast_codec structure was reverted and an internal_ast_codec structure was created just for use in codec.c. A new internal-only API was also added (__ast_codec_register_with_format) so that codec_builtin could register codecs with the format_name in a separate parameter rather than in the ast_codec structure. ASTERISK-26144 #close Reported-by: Alexei Gradinari Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba 2016-06-28 08:22 +0000 [f3d236ca7f] gtjoseph * BuildSystem: Fix a few issues hightlighted by gcc 6.x gcc 6.1.1 caught a few more issues. Made sure the unit tests still pass for the func_env and stdtime issues. ASTERISK-26157 #close Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e 2016-06-28 10:33 +0000 [9d5b0934d9] Matt Jordan * configs/basic-pbx/modules.conf: Remove 'bad' modules This patch removes the following modules: - pbx_functions: It never existed. - res_pjsip_log_forwarder: It no longer exists. - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs aren't going to be installing HOMER - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't loaded, and we aren't configured to make use of the module Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5 2016-06-22 11:19 +0000 [1dfc286418] Joshua Colp * siren: Add format attribute modules for Siren7 and Siren14. This change removes hardcoded SDP parsing and generation for Siren7 and Siren14 from chan_sip and moves it to format attribute modules so it can also be used by chan_pjsip. With this the fmtp lines for both are added with the bitrate information. ASTERISK-26021 Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037 2016-06-23 04:33 +0000 [5f0a098243] Alexander Traud * BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf. Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C but requires ANSI C anyway. ASTERISK-26046 Change-Id: I914c014385e1862102d90fe7650621def78db02e 2016-06-22 15:04 +0000 [3d904659ec] Corey Farrell * res_fax: Fix reference leak in fax_v21_session_new. fax_v21_session_new created a session details object but only released the allocation reference during error conditions. fax_session_new adds it's own reference to details if needed so the caller is always responsible for cleaning it's own reference. ASTERISK-26141 #close Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88 2016-06-22 13:41 +0000 [48db4c2159] gtjoseph * res_rtp_asterisk: Fix a self-comparison identified by gcc 6 gcc 6 caught a previously unidentified self-comparison in ice_candidate_cmp. Fixed it and re-ordered the predicates for better short-circuiting. ASTERISK-26140 #close Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7 2016-06-22 10:37 +0000 [bc69b03316] gtjoseph * chan_unistim: Fix memcpy in get_to_address A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) was using a pointer to a pointer as the destination of a memcpy and a '&' instead of '*' in the sizeof. ASTERISK-26138 #close Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708 2016-06-20 13:18 +0000 [1b79e2deff] Mark Michelson * Fix Alembic upgrades. A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. This patch fixes another realtime problem as well. Our Alembic scripts store booleans as yes or no values. However, Sorcery tries to insert "true" or "false" instead. This patch updates Sorcery to use "yes" and "no" ASTERISK-26128 #close Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec 2016-06-22 10:55 +0000 [e30602587c] Alexander Traud * BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf. Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not support the platform SVR2 from the year 1987 anymore. ASTERISK-26046 Change-Id: I28161b037feb2d29ab46ed20e785928460226c22 2016-06-22 10:51 +0000 [77da168e58] gtjoseph * test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO Since the file was missing the depends on pjproject, it wasn't picking up the pjproject related include path. If there was no system installed pjproject and pjproject-bundled was used, a compile would fail because pjsip.h wasn't found. ASTERISK-26139 #close Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757 2016-06-02 17:26 +0000 [b3c787d1dd] Alexei Gradinari * res_pjsip: improve realtime performance #2 The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df 2016-06-21 06:52 +0000 [dfcd466bf0] Torrey Searle * res_rtp_asterisk: fix memory leak in dtls ensure that cert bios get freed after creating the fingerprint ASTERISK-26129 #close Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451 2016-06-21 17:42 +0000 [c982da0641] Richard Mudgett * res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c 2016-06-12 11:19 +0000 [6a568bcc66] gtjoseph * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription Occasionally under load we'll attempt to send a final NOTIFY on a subscription that's already been terminated and a SEGV will occur down in pjproject's evsub_destroy function. This is a result of a race condition between all the paths that can generate a notify and/or destroy the underlying pjproject evsub object: * The client can send a SUBSCRIBE with Expires: 0. * The client can send a SUBSCRIBE/refresh. * The subscription timer can expire. * An extension state can change. * An MWI event can be generated. * The pjproject transaction timer (timer_b) can expire. Normally when our pubsub_on_evsub_state is called with a terminate, we push a task to the serializer and return at which point the dialog is unlocked. This is usually not a problem because the task runs immediately and locks the dialog again. When the system is heavily loaded though, there may be a delay between the unlock and relock during which another event may occur such as the subscription timer or timer_b expiring, an extension state change, etc. These may also cause a terminate to be processed and if so, we could cause pjproject to try to destroy the evsub structure twice. There's no way for us to tell that the evsub was already destroyed and the evsub's group lock can't tolerate this and SEGVs. The remedy is twofold. * A patch has been submitted to Teluu and added to the bundled pjproject which adds add/decrement operations on evsub's group lock. * In res_pjsip_pubsub: * configure.ac and pjproject-bundled's configure.m4 were updated to check for the new evsub group lock APIs. * We now add a reference to the evsub group lock when we create the subscription and remove the reference when we clean up the subscription. This prevents evsub from being destroyed before we're done with it. * A state has been added to the subscription tree structure so termination progress can be tracked through the asyncronous tasks. * The pubsub_on_evsub_state callback has been split so it's not doing double duty. It now only handles the final cleanup of the subscription tree. pubsub_on_rx_refresh now handles both client refreshes and client terminates. It was always being called for both anyway. * The serialized_on_server_timeout task was removed since serialized_pubsub_on_rx_refresh was almost identical. * Missing state checks and ao2_cleanups were added. * Some debug levels were adjusted to make seeing only off-nominal things at level 1 and nominal or progress things at level 2+. ASTERISK-26099 #close Reported-by: Ross Beer. Change-Id: I779d11802cf672a51392e62a74a1216596075ba1 2016-06-21 07:05 +0000 [ef97911a1c] Alexander Traud * res_rtp_asterisk: Use latest DTLS version available by underlying platform. Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based cipher-suites. ASTERISK-26130 #close Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0 2016-06-21 10:53 +0000 [69d58a1e37] Scott Griepentrog * PJSIP: provide transport type with received messages The receipt of a SIP MESSAGE may occur over any transport including TCP and TLS. When the message is received, the original URI is added to the message in the field PJSIP_RECVADDR, but this is insufficient to ensure a reply message can reach the originating endpoint. This patch adds the PJSIP_TRANSPORT field populated with the transport type. ASTERISK-26132 #close Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e 2016-06-21 08:01 +0000 [cbfa9f771e] Alexander Traud * BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. Some configure scripts used both AC_HELP_STRING and its replacement AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were changed to AS_HELP_STRING. ASTERISK-26046 Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f 2016-06-20 10:29 +0000 [ba0d9e7f7a] Joshua Colp * res_pjsip_session: Handle race condition at shutdown with timer. When shutting down res_pjsip_session will get unloaded before res_pjsip. The act of unloading unregisters all the PJSIP services and sets their module IDs to -1. In some cases it is possible for a timer to occur after this happens which calls into res_pjsip_session. The res_pjsip_session module can then try to get the session from the INVITE session using the module ID. Since the module ID is now -1 this fails. This change stores a copy of the module ID and uses it for the timer callback scenario. If the module ID is -1 the callback immediately returns but if the module ID is valid then it continues as normal. This works as the original ID of the module is guaranteed to still be valid when used with the INVITE session. ASTERISK-26127 #close Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573 2016-06-20 12:13 +0000 [c1512f4108] Richard Mudgett * app_voicemail.c: Fix IMAP compile error. Fix compile error introduced by the patch for ASTERISK-26045 Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3 2016-06-16 15:56 +0000 [5134a8043a] Alexei Gradinari * fix: memory leaks, resource leaks, out of bounds and bugs ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c 2016-06-13 17:40 +0000 [cfebe3b94a] Mark Michelson * ARI: Ensure announcer channels are destroyed. Announcer channels were not being destroyed because the stasis_app_control structure that referenced them was not being destroyed. The control structure was not being destroyed because it was not being unlinked from its container. It was not being unlinked from its container because the after bridge callback for the announcer channel was not being run. The after bridge callback was not being run because the after bridge datastore was not being removed from the channel on destruction. The channel was not being destroyed because the hangup that used to destroy the channel was now only reducing the reference count to one. The reference count of the channel was only being reduced to one because the stasis_app_control structure was holding the final reference... The control structure used to not keep a reference to the channel, so that loop described above did not happen. The solution is to manually remove the control structure from its container when the playback on a bridge is complete. ASTERISK-26083 #close Reported by Joshua Colp Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4 2016-06-20 08:05 +0000 [76516bd79d] Alexander Traud * http: leverage 'bindaddr' for TLS in http.conf The internal HTTP/WebSocket server supports both TCP and TLS, which can be activated separately via the file http.conf. The source code intends to re-use the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified explicitly. This did not work because of a typo. This change resolves this typo. ASTERISK-26126 #close Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f 2016-05-31 09:10 +0000 [89cc86fc38] Vasil Kolev * chan_sip: bigger buffers for headers, better failure mode Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3 2016-05-18 17:37 +0000 [d53a36ff33] Richard Mudgett * res_pjsip_transport_management.c: Misc cleanups to survive shutdown. * In unload_module(), reordered destroying things to minimize the window that the global transports container could be used by other threads on shutdown. When shutting down you need to stop things in the opposite order of creation. * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to eliminate the crash potential by other threads using the container on shutdown. * Made struct monitored_transport.sip_received not use ast_atomic_fetchadd_int() since it is used as a boolean value that is only set TRUE. It was previously incremented for every received SIP message and could theoretically overflow. * In monitored_transport_state_callback(), allocated the monitored transport object without a lock since the lock was unused. * In keepalive_global_loaded(), removed releasing the transports container if the keepalive_thread could not be started. I set it up to be tried again if the user reloads the configuration. Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff 2016-01-05 19:08 +0000 [03953d8034] Richard Mudgett * res_pjsip.c: Add check that timer actually got scheduled. Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1 2016-06-13 13:33 +0000 [32ab98116e] Richard Mudgett * res_rtp_multicast.c: Fix warning message typo. Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3 2016-02-11 18:15 +0000 [0429c53368] Richard Mudgett * res_pjsip_session.c: Reorganize ast_sip_session_terminate(). Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b 2016-06-10 12:35 +0000 [5823f279f3] Richard Mudgett * chan_rtp: Backport changes from master. * Deprecate chan_multicast_rtp. Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e 2016-06-10 16:13 +0000 [dde58df318] Richard Mudgett * chan_rtp.c: Copy file from chan_multicast_rtp.c Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef 2016-06-08 06:15 +0000 [ca38a3cbb4] Alexander Traud * core: Not the configured but granted number of possible file descriptors. With CLI "core show settings", simply the parameter maxfiles of the file asterisk.conf was shown. If that parameter was not set, nothing was displayed although the environment might have set a default number itself. Or if maxfiles were not granted (completely), still maxfiles was shown. Now, the maximum number of possible file descriptors in the environment is shown. ASTERISK-26097 Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b 2016-06-07 18:45 +0000 [caf6cccc5c] Joshua Colp * cel: Ensure only one dial status per channel exists. CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe 2016-06-09 10:37 +0000 [715ef071a1] Mark Michelson * chan_pjsip: Lock channel when checking for RTP changes. bridge_native_rtp can call into an RTP-capable channel driver in order for the driver to update information about who the channel is communicating with. For SIP channel drivers, this means deactivating RTCP and sending a reinvite so that the endpoints can communicate directly. bridge_native_rtp does the right thing and has the channel locked when calling into the channel driver. chan_pjsip can't alter session properties in this thread, though. chan_pjsip queues a task on the session serializer in order to update properties there. The problem is that this queued task was not locking the channel. This meant that the queued task could attempt to deactivate RTCP at the same time that the channel thread was attempting to process an incoming RTCP packet. This could lead to a crash. This patch fixes the issue by locking the channel in the queued task when altering RTP properties. ASTERISK-26092 #close Reported by Niklas Larsson Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159 2016-06-09 09:20 +0000 [a99ddc6a0d] gtjoseph * build: Fix ast_sockaddr initialization to be more portable A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4 2016-06-08 12:26 +0000 [eabb398d71] Matt Jordan * res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea 2016-06-08 05:58 +0000 [0d84421f93] Alexander Traud * astfd: Not maximum size of a single file but maximum file descriptors. With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a single file was shown. Now, the maximum number of possible file descriptors is shown. ASTERISK-26097 Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3 2016-06-02 14:53 +0000 [9c5a0b814b] Timo Teräs * Fix #include poll.h and sys/cdefs.h POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be 2016-06-03 22:44 +0000 [9c35f34301] Richard Mudgett * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. This patch fixes a race condition processing received REGISTER requests and their retransmissions caused by REGISTER requests being processed by two threads. The "sip_transaction Unable to register REGISTER transaction (key exists)" message is a notable symptom of this issue. This issue was more likely to happen before the pjsip/distributor serializers were created. Instead of steps one and two below placing the REGISTER messages into the same pjsip/distributor they were placed in random pjsip/default serializers. 1) REGISTER requests come in and get placed on the pjsip/distributor serializer. 2) Before the first request is processed a retransmission comes in and is placed on the same pjsip/distributor serializer. 3) The first request goes up the pjsip stack and is then shunted off to the pjsip/aor/ serializer. 4) Before the first request is completed processing in the pjsip/aor/ serializer, the second request goes up the pjsip stack and is also shunted off to the pjsip/aor/ serializer. 5) The first request completes processing and sends out its response. 6) The second request completes processing and tries to send out its response but pjlib complains that the REGISTER transaction key already exists. 7) Sadness ensues. * The race is eliminated by removing the pjsip/aor/ serializer and continuing the processing in the pjsip/distributor serializer. Now any retransmissions queued in the pjsip/distributor serializer will be processed after the first message is completely processed. ASTERISK-26088 #close Reported by: Richard Mudgett Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a 2016-06-03 11:35 +0000 [557333ea4c] Richard Mudgett * stasis: Add setting subscription congestion levels. Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244 2016-06-02 18:19 +0000 [110d772467] Richard Mudgett * sorcery: Add setting object type congestion levels. Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 2016-06-02 16:08 +0000 [610eee2a36] Richard Mudgett * taskprocessors: Implement high/low water mark alerts. When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa 2016-05-27 17:31 +0000 [26e3492246] Richard Mudgett * res_pjsip_session: Use distributor serializer for incoming calls. We must continue using the serializer that the original INVITE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. Outgoing call legs create the pjsip/outsess/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc 2016-05-27 16:28 +0000 [ceb1007ed7] Richard Mudgett * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be 2016-05-27 12:50 +0000 [27bafc3a8b] Richard Mudgett * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0 2016-05-26 17:35 +0000 [16b08444da] Richard Mudgett * pjsip_distributor.c: Consistently pick a serializer for messages. Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407 2016-06-02 12:51 +0000 [993b769524] Richard Mudgett * pjsip_distributor.c: Ignore messages until fully booted. We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 2016-06-02 12:04 +0000 [321a9b128f] Joshua Colp * res_odbc: Implement a connection pool. Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff 2016-06-07 05:45 +0000 [c6ee4a0f44] Alexander Traud * res_srtp: Instead of libSRTP use OpenSSL as random source. Since libSRTP 1.5, its Random Number Generator (RNG) is not maintained anymore. Therefore, the symbol RAND_bytes is used instead of crypto_get_random. ASTERISK-24436 #close Change-Id: Iea0bae4d4e3c9aa0926ea442b6484b5159789d96 2016-06-07 02:16 +0000 [d38b8e6399] Alexander Traud * BuildSystem: Avoid 'ar cru' and use 'ar cr' instead. In several internal library projects, the files are archived with the help of 'ar cr'. Only the projects editline and the Objective Open H.323 stack implementation in C (ooh323c) use 'ar cru' instead. Recently, some platforms changed the default parameters of AR which creates "/usr/bin/ar: `u' modifier ignored since `D' is the default (see `U')". For consistency and to avoid this message all projects use 'ar cr' now. ASTERISK-26091 #close Change-Id: I710a9b1c01c1b5a1931a646098c044c8161ead40 2016-05-27 14:49 +0000 [c27c232057] gtjoseph * ari/resource_channels: Add 'formats' to channel create/originate If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b 2016-06-02 04:59 +0000 [cda3385409] Joshua Colp * alembic: Fix migration. The 81b01a191a46_pjsip_add_contact_reg_server.py script was attempting to use UniqueConstraint and failing. It was not imported and after importing it also continued to fail. I've changed the script to use the explicit name of the constraint instead. Change-Id: I2438b0be90b7ce583b47dd27983c0c1a02cea5b9 2016-06-01 13:57 +0000 [e2132dd358] Richard Mudgett * logging,cdr,cel: Fix stringfield memory leak. The stringfields refactor to allow adding stringfields to the end of a structure (f6f4cf459f43f072604927209b39646f84aaa2e2) exposed some incomplete cleanup code by some stringfield users. The most noticeable leaker is the logging system where there is a leak for every log message generated. ASTERISK-26078 #close Reported by: Etienne Lessard Patches: jira_asterisk_26078_v13.patch (license #5621) patch uploaded by Richard Mudgett Change-Id: If6a08b31336b492c3de6f9dfd07c447f8d5a8782 2016-05-25 10:34 +0000 [2de58c6d01] Alexei Gradinari * core/dial: New channel variable FORWARDERNAME Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2 2016-05-31 13:02 +0000 [b2ce0e354b] Richard Mudgett * pjsip_distributor.c: Use correct rdata info access method (Part 2). The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: I4d519385a577f3e9d9193a88125e493cf17fa799 2016-05-30 19:27 +0000 [fe305ccf01] gtjoseph * res_pjsip_mwi_body_generator: Re-order the body items Re-ordered the body items so Message-Account is second. Messages-Waiting: no Message-Account: sip:1571@:5060 Voice-Message: 0/0 (0/0) ASTERISK-26065 #close Reported-by: Ross Beer Change-Id: If5d35a64656eac98c2dd5e490cc0b2807bed80c3 2016-05-30 10:58 +0000 [e8abfdcdc5] gtjoseph * pjproject_bundled: Move to pjproject 2.5 Although all the patches we had against 2.4.5 were applied by Teluu, a new bug was introduced preventing re-use of tcp and tls transports This patch removes all the previous patches against 2.4.5, updates the version to 2.5, and adds a new patch to correct the transport re-use problem. Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068 2016-05-27 12:25 +0000 [37d039fdf3] Rusty Newton * res_pjsip: Add clarifying documentation to PJSIP_HEADER help text Added notes about when you can read or write headers. Specifically about being able to read on the inbound channel and write on an outbound channel. ASTERISK-26063 #close Reported by: Private Name Tested by: Rusty Newton Change-Id: Ibeb64af17d1f6451028b3c29855a3f151a01d8c5 2016-05-25 18:30 +0000 [03d5b3ce5c] Richard Mudgett * pjsip_distributor.c: Use correct rdata info access method. The pjproject doxygen for rdata->msg_info.info says to call pjsip_rx_data_get_info() instead of accessing the struct member directly. You need to call the function mostly because the function will generate the struct member value if it is not already setup. Change-Id: Iafe8b01242b7deb0ebfdc36685e21374a43936d2 2016-05-20 13:56 +0000 [859bbec09b] Alexei Gradinari * app_voicemail: fix bugs, imap mm_status log change to debug Fixed some bugs: - create dirpath when save downloading message from IMAP storage. - create IMAP folder if not exists when saving to IMAP storage - check if file successfully opened before write to it - some IMAP checks - remove non-standard flag 'Unseen' etc Change to debug IMAP mm_status log instead of verbose. Remove unused X-Asterisk-VM-Caller-channel message header for security reason. The clients should not know name of peer/endpoint. ASTERISK-26045 #close Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b 2016-05-19 14:56 +0000 [230686f4ec] Alexei Gradinari * res_pjsip: add "via_addr", "via_port", "call_id" to contact As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576 2016-05-24 16:56 +0000 [04c12561a7] Alexei Gradinari * res_pjsip: chatty verbose messages There are a lot of verbose messages about Endpoint and Contact status changes if there are many dynamic endpoints. The patch sets verbose level 2 for Endpoint status changes and verbose level 3 for Contact status changes. ASTERISK-26055 #close Change-Id: Ie64e261ddbbc41bfff0f0190241152cc123fe6d7 2016-02-12 09:59 +0000 [a42bea3314] Corey Farrell * threadpool: Fix potential data race. worker_start checked for ZOMBIE status without holding a lock. All other read/write of worker status are performed with a lock, so this check should do the same. ASTERISK-25777 #close Change-Id: I5e33685a5c26fdb300851989a3b82be8c4e03781 2016-05-18 10:58 +0000 [a32616d60c] Tzafrir Cohen * Makefile: remove OSARCH check for init install There are more specific checks for the platform. Specifically this allows installing OS/X init scripts. ASTERISK-26038 #close Change-Id: If08933621145b10362a0cfe73c079301d9c13f50 Signed-off-by: Tzafrir Cohen 2016-05-21 05:42 +0000 [9ddaab789e] Jesper (License 5518) * func_curl: Don't trim response text on non-ASCII characters The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of a signed comparison. ASTERISK-25669 #close Reported by: Jesper patches: strings.curl.trim.patch submitted by Jesper (License 5518) Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a 2016-05-20 16:59 +0000 [9453d1187a] Richard Mudgett * parking.h: Update ast_parking_park_call() doxygen to reality. ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260 2016-05-10 14:30 +0000 [cd89501d48] Alexei Gradinari * func_odbc: single database connection should be optional func_odbc was changed in Asterisk 13.9.0 to make func_odbc use a single database connection per DSN because of reported bug ASTERISK-25938 with MySQL/MariaDB LAST_INSERT_ID(). This is drawback in performance when func_odbc is used very often in dialplan. Single database connection should be optional. ASTERISK-26010 Change-Id: I57d990616c957dabf7597dea5d5c3148f459dfb6 2016-05-20 09:39 +0000 [c0b190dd9a] Mark Michelson * res_pjsip: Match dialogs on responses better. When receiving an incoming response to a dialog-starting INVITE, we were not matching the response to the INVITE dialog. Since we had not recorded the to-tag to the dialog structure, the PJSIP-provided method to find the dialog did not match. Most of the time, this was not a problem, because there is a fall-back that makes the response get routed to the same serializer that the request was sent on. However, in cases where an asynchronous DNS lookup occurs in the PJSIP core, the thread that sends the INVITE is not actually a threadpool serializer thread. This means we are unable to record a serializer to handle the incoming response. Now, imagine what happens when an INVITE is sent on a non-serialized thread, and an error response (such as a 486) arrives. The 486 ends up getting put on some random threadpool thread. Eventually, a hangup task gets queued on the INVITE dialog serializer. Since the 486 is being handled on a different thread, the hangup task can execute at the same time that the 486 is being handled. The hangup task assumes that it is the sole owner of the INVITE session and channel, so it ends up potentially freeing the channel and NULLing the session's channel pointer. The thread handling the 486 can crash as a result. This change has the incoming response match the INVITE transaction, and then get the dialog from that transaction. It's the same method we had been using for matching incoming CANCEL requests. By doing this, we get the INVITE dialog and can ensure that the 486 response ends up being handled by the same thread as the hangup, ensuring that the hangup runs after the 486 has been completely handled. ASTERISK-25941 #close Reported by Javier Riveros Change-Id: I0d4cc5d07e2a8d03e9db704d34bdef2ba60794a0 2016-05-19 11:41 +0000 [ddcf983e39] Joshua Colp * res_sorcery_astdb: Filter fields to only the registered ones. This change introduces the same filtering that is done in res_sorcery_realtime to the res_sorcery_astdb module. This allows persisted sorcery objects that may contain unknown fields to still be read in from the AstDB and used. This is particularly useful when switching between different versions of Asterisk that may have introduced additional fields. ASTERISK-26014 #close Change-Id: Ib655130485a3ccfd635b7ed5546010ca14690fb2 2016-05-09 21:40 +0000 [39fedfa423] snuffy * res_pjsip_empty_info: Respond to empty SIP INFO packets Some SBCs require responses to empty SIP INFO packets after establishing call via INVITE, if not responded to they may drop your call after unspecified timeout of X minutes. They are identified by having no Content-Type, check for this and respond with 200 - OK message. ASTERISK-24986 #close Reported-by: Ilya Trikoz, Federico Santulli Change-Id: Ib27e4f07151e5aef28fa587e4ead36c5b87c43e0 2016-05-18 07:54 +0000 [935e0496c4] gtjoseph * udptl: Don't eat sequence numbers until OK is received Scenario: Local fax -> Asterisk w/ firewall -> Provider -> Remote fax * Local fax starts rtp call to remote fax * Remote fax starts t38 call back to local fax. * Local fax sends t38 no-signal to Asterisk before sending an OK. * udptl processes the frame and increments the expected sequence number. * chan_sip drops the frame because the call isn't up so nothing goes out the external interface to open the port for incoming packets. * Local fax sends OK and Asterisk sends OK to the remote fax. * Remote fax sends t38 packets which are dropped by the firewall. * Local fax re-sends t38 no-signal with the same sequence number. * udptl drops the frame because it thinks it's a dup. * Still no outgoing packets to open the firewall. * t38 negotiation fails. The patch drops frames t38 received before udptl sequence processing when the call hasn't been answered yet. The second no-signal frame is then seen as new and is relayed out the external interface which opens the port and allows negotiation to continue. ASTERISK-26034 #close Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 2016-05-17 11:14 +0000 [77e8ec162b] gtjoseph * chan_sip: Prevent extra Session-Expires headers from being added When chan_sip does a re-INVITE to refresh a session and authentication is required, the INVITE with the Authorization header containes a second Session-Expires header without the ";refersher=" parameter. This is causing some proxies to return a 400. Also, when Asterisk is the uas and the refresher, it is including the Session-Expires and Min-SE headers in OPTIONS messages which is not allowed per RFC4028. This patch (based on the reporter's) Checks to see if a Session-Expires header is already in the message before adding another one. It also checks that the method is INVITE or UPDATE. ASTERISK-26030 #close Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 2016-05-16 15:29 +0000 [3f6ef63099] gtjoseph * res_pjsip_outbound_registration: Clean up state when registration is deleted Nothing was cleaning up the registration state object when ast_sorcery_delete was called on a registration. So, the registration was deleted from sorcery but the state object went right on refreshing the registration (or failing to refresh the registration) with the peer. * Added a 'deleted' observer on registration that removes the state object. ASTERISK-25964 #close Reported-by Matt Jordan Change-Id: I2db792145cdb1f72ebbf57dd9099596dbbf12c23 2016-05-15 19:05 +0000 [b6f9392a12] gtjoseph * res_pjsip: Set TCP_NODELAY on TCP transports Although it's perfectly legal to place multiple SIP messages in the same packet, it can cause problems because the Linux default is to enable Path MTU Discovery which sets the Don't Fragment bit on the packets. If adding a second message to the packet causes the MTU to be exceeded, and the destination isn't equipped to send a FRAGMENTATION NEEDED response to a large packet, the packet will just be dropped. We can't specifically tell the stack to send only 1 message per packet, but we can turn on TCP_NODELAY when we create the transport. This will at least tell the stack to send packets as soon as possible. ASTERISK-26005 #close Reported-by: Ross Beer Change-Id: I820f23227183f2416ca5e393bec510e8fe1c8fbd 2016-05-14 21:48 +0000 [361a16f316] Matt Jordan * configs/samples/pjsip.conf.sample: Fix typo A ':' is not a valid token for starting a comment. Change-Id: I123592d93a83d1bdde3e352822881eb9da85e5ad 2016-05-12 07:08 +0000 [f91a7dc993] Matt Jordan * res/res_hep_pjsip: Fix reported local IP address when bound to 'any' When bound to an 'any' address, e.g., 0.0.0.0, PJSIP reports as its local address the 'any' address, as opposed to the IP address we actually received the packet on. This can cause some confusion in Homer, as it will dutifully report what we send it. This patch uses the PJSIP inspection routines to determine which IP address we probably received the packet on based on the remote party's IP address. In the event that this fails, it falls back to the IP address natively reported by the transport. Change-Id: I076f835d2aef489e1ee1d01595b211eb2ce62da3 2016-05-14 12:29 +0000 [9de5cd209e] Sean Bright * res_ari: Correct Location headers returned by some ARI resources The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c 2016-05-13 11:38 +0000 [524a302974] Alexei Gradinari * res_pjsip: Endpoint IP Access Controls With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02 2016-05-11 20:17 +0000 [89ae4466ea] Matt Jordan * res_hep: Provide an option to pick the UUID type At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c 2016-05-10 02:56 +0000 [a73d79c22f] Tzafrir Cohen * basic-cfg: asterisk.conf: remove [directories] A minimal configuration does not need to explicitly spell out the directories. The built-in defaults will do just fine. In many cases they are wrong. Change-Id: Id1a671e5c5e9923765a4156b57f9f7e263fdd26c Signed-off-by: Tzafrir Cohen 2016-05-10 03:06 +0000 [1c56de9453] Tzafrir Cohen * basic-cfg: asterisk.conf: defaults of options Note the default of remmed-out options. To clarify that those values are not the defaults. Change-Id: I849c29b7a710f0abc37355fcb5bfee335ae30738 Signed-off-by: Tzafrir Cohen 2016-05-10 03:08 +0000 [d7af591c59] Tzafrir Cohen * basic-cfg: asterisk.conf: debug level 5 spams Don't suggest users to use debug level 5, which spews (usually non-useful) debug information. Reduce the suggestion to (an arbitrarily-selected) level 2. Change-Id: Ib53195f78945970956ff59ef13fa89b90e0fcd60 Signed-off-by: Tzafrir Cohen 2016-05-10 03:10 +0000 [9b7db18fc1] Tzafrir Cohen * basic-cfg: asterisk.conf: don't set languages * No need to set language in a miniml configuration. 'en' will do just fine. * It would be useful to have an example of setting it to a different language. * Setting the documentation language explicitly is likewise not required. Setting it to a different value is not common. At least until there is a set of translated documentation. Change-Id: I94d91ea34e129925f25af81ef8dc0906fb568cb7 Signed-off-by: Tzafrir Cohen 2016-05-10 08:17 +0000 [eec539a46e] Tzafrir Cohen * followme: delete the right recorded name file FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen 2016-05-12 14:36 +0000 [02d30e171e] Mark Michelson * Use doubles instead of floats for conversions when comparing strings. In 13.9.0, there was an issue where PJSIP contacts added to an AOR would be deleted at seemingly random times. One reason this was happening was because of an operation to retrieve the contacts whose expiration time was less than or equal to the current time. When retrieving existing contacts, the contact's expiration time and the current time were converted from a string to a float, and those two floats were compared. On some systems, including mine, this conversion was horribly off. For instance, I could regularly see the string "1463079214" get converted into 1463079168.000000. When switching from using a float to using a double, the conversion was as expected. Why was the conversion to float off? My best guess is that the conversion to float was attempting to store the entire value in the 23 bit significand of the IEEE-754 floating point number. In particular, if you take only the 23 most significant bits of 1463079214, you get the messed up 1463079168 that we were seeing in the conversion. It likely was possible to get a more precise value by composing the number using an exponent, but the conversion did not work that way. With a double, you have a 52 bit significand, allowing the entire value to fit there, and thereby allowing an accurate conversion. ASTERISK-26007 #close Reported by Greg Siemon Change-Id: I83ca7944aae8b7cd994b254c78ec02411d321070 2016-05-12 09:13 +0000 [e2df15bae9] gtjoseph * pjsip_distributor: Add missing newline to NOTICE There was a newline missing from the end of the "no matching endpoint" notice. Change-Id: Idc11fe5bc0354072291663dbffe648c471e39181 2016-05-10 10:19 +0000 [a94a12bbf7] Sebastian Damm * res_pjsip_outbound_registration: generate correct Contact URI for TLS There are two types of SIP URIs indicating a secure transport: * sips:user@example.org * sip:user@example.org;transport=tls When using a sips URI, Asterisk checks incoming INVITEs and answers from the other side for sips URIs, and rejects the packet if there are only sip URIs. So Asterisk should only generate a sips Contact URI if the other side supports it. This patch makes Asterisk generate either a sip or sips Contact URI depending on the format of the server URI. If you want a sip URI, use: server_uri=sip:example.org\;transport=tls If you want a sips URI, use: server_uri=sips:example.org ASTERISK-25990 #close Reported-by: Sebastian Damm Change-Id: I5ae57d6531ce940b5fc64d5cd2673e60db0f9ba2 2016-05-05 16:41 +0000 [36d66a23e0] Alexei Gradinari * logger: Add PID to syslog messages. During refactoring of this support the addition of the PID to messages was removed. This change adds it back in. ASTERISK-25538 #close Change-Id: Ie2d43b0652e59b7ac319a7dba94501540d70ba36 2016-05-11 14:07 +0000 [37214b0bdf] Matt Jordan * configure: Fix errors with AST_UNDEFINED_SANITIZER/AST_LEAK_SANITIZER When running on a system that does not support or use AST_UNDEFINED_SANITIZER or AST_LEAK_SANITIZER, the configure script would incorrectly set those constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would cause menuselect to error out, complaining that a blank value is not a valid option. This patch corrects the issue by setting the value to 0 if the options that those constants enable/disable is not found. Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba 2016-05-03 15:43 +0000 [49b25a0956] Kevin Harwell * res_pjsip_outbound_publish: state potential dropped on reloads/realtime fetches When reloading, or fetching realtime data, if the "apply" failed for any numerous reasons the current state object would not be maintained. This potentially resulted in publishes being stopped for some states/clients when they should not have been. This patch makes it so the current state object is kept upon any type of reload/ fetch failures. Change-Id: Iab6020c116d628ed2ae81183e987e2eaa3c90b30 2016-05-03 15:31 +0000 [1b5c91b7be] Kevin Harwell * res_pjsip_outbound_publish: Potential crash due to off nominal path It was possible for the explicit publish destroy function to be called without the pjsip client ever being initialized. This fix checks to make sure there is a client to destroy before attempting. Change-Id: I8eea1bfa3bd472149bfc255310be2a6248688f5c 2016-05-03 15:35 +0000 [10de553c9d] Kevin Harwell * res_pjsip_outbound_publishing: After unloading the library won't load again The same thing was happening in res_pjsip_publish_asterisk. When the library was unloaded it did not unregister the object type from sorcery. Subsequent loads resulted in a failed load due to the sorcery type already existing. Change-Id: Ifdc25e94e4cd40bc5a19eb4d0a00b86c2e9fedc9 2016-05-03 14:59 +0000 [1a833b9739] Kevin Harwell * res_pjsip_outbound_publish: Ref leak in off nominal callback paths There were a few spots where the client object's reference was being leaked in sip_outbound_publish_callback. This patch cleans up those leaks. Change-Id: I485d0bc9335090f373026f77c548042e258461df 2016-05-03 15:39 +0000 [4752ef02e0] Kevin Harwell * res_pjsip_outbound_publish: Won't unload if condition wait times out When res_pjsip_outbound_publish unloads it has to wait for all current publishing objects to get done. However if the wait condition times out then it does not fail the unload. This sometimes results in an infinite loop check while unloading. This patch now fails the unload operation if the condition times out. Change-Id: Id57b8cbed9d61222690fcba1e4f18e259df4c7ec 2016-05-05 11:37 +0000 [4d063814ba] Kevin Harwell * res_pjsip_authenticator_digest: Don't use source port in nonce verification From the issue reporter: "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of the timestamp, the source address, the source port, a server UUID that is calculated at startup, and the authentication realm. Rather than caching nonces that we create, we instead attempt to re-calculate the nonce when receiving an incoming request with authentication. We then compare the re-calculated nonce to the incoming nonce, and if they don't match, then authentication has failed early. The problem is that it is possible, especially when using TCP, to receive two requests from the same endpoint but have differing source ports for those requests. Asterisk itself commonly will use different source ports for outbound TCP requests." This patch removes the source port dependency when building the nonce. ASTERISK-25978 #close Change-Id: I871b5f4adce102df1c4988066283095ec509dffe 2016-05-07 14:39 +0000 [fb6227a372] gtjoseph * config_transport: Tell pjproject to allow all SSL/TLS protocols The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. SSL is not allowed. So, even if you specify "sslv3" for a transport method, it's silently ignored and one of the TLS protocols is used. This was a new behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that we never caught. Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). This tells pjproject to set the socket protocol to match the method. ASTERISK-26004 #close Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078 2016-05-04 02:40 +0000 [2db17a793c] Jaco Kroon * app_confbridge: Add a regcontext option for confbridge bridge profiles. This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f 2016-05-08 20:19 +0000 [2a7130b8b0] gtjoseph * pjproject_bundled: Check for python-dev and TEST_FRAMEWORK The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. The python bindings are now built only if TEST_FRAMEWORK is set and a python development package is installed. libresample was also disabled. ASTERISK-25993 #close Reported-by: Joshua Colp Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03 2016-05-04 15:16 +0000 [72eb7c8301] Alexei Gradinari * res_pjsip: module load priority The res_pjsip_authenticator_digest, res_pjsip_endpoint_identifier_* and res_pjsip_registrar modules should load ASAP to avoid "No matching endpoint found" for legitimate endpoint. ASTERISK-25994 Change-Id: Iac95d95ad031e0be104189d29e923a2ad7c24a1b 2016-05-04 03:17 +0000 [dd00c71aae] Chris Trobridge * config_options.c: Expand #ifdef to contain whole if statement. ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 2016-05-05 15:16 +0000 [e6eb17efd9] Alexei Gradinari * stasis_endpoints: Add new Status and Headers to ContactStatus ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. These additions should be also in stasis_endpoints to include in command "manager show event ContactStatus" Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a 2016-05-05 05:07 +0000 [fa11f4c920] Joshua Colp * file: Ensure nativeformats remains valid for lifetime of use. It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 2016-04-15 09:32 +0000 [9c2032240e] Alexei Gradinari * res_pjsip: improve realtime performance This patch modified pjsip_options to retrieve only permament contacts for aor if the qualify_frequency is > 0 and persisted contacts if the qualify_frequency is > 0. This patch also fixed a bug in res_sorcery_astdb. res_sorcery_astdb doesn't save object data retrived from astdb. ASTERISK-25826 Change-Id: I1831fa46c4578eae5a3e574ee3362fddf08a1f05 2016-05-04 16:11 +0000 [fe38d21c2a] Alexei Gradinari * pjsip: Added "reg_server" to contacts (fixed alembic) ASTERISK-25931 Change-Id: Icc4321a88f5c93ff809da3f372eebbf69c6a8549 2016-04-07 16:33 +0000 [7a14e669f0] Alexei Gradinari * res_pjsip/AMI: add contact.updated event With the old SIP module AMI sends PeerStatus event on every successfully REGISTER requests, ie, on start registration, update registration and stop registration. With PJSIP AMI sends ContactStatus only when status is changed. Regarding registration: on start registration - Created on stop registration - Removed but on update registration nothing This patch added contact.updated event. ASTERISK-25904 Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f 2016-05-02 16:08 +0000 [06d4ac0355] Alexei Gradinari * res_fax: add FAXMODE variable The app_fax set FAXMODE variable, but res_fax missing this feature. This patch add FAXMODE variable which is set to either "audio" or "T38". ASTERISK-25980 Change-Id: Ie3dcbfb72cc681e9e267a60202f7fb8723a51b6b 2016-05-02 16:52 +0000 [2d17fe06c5] Alexei Gradinari * res_fax/t38_gateway: Peer V.21 session is created on wrong channel The channel and peer V.21 sessions are created on the same channel now. The peer V.21 session should be created only on peer channel when one of channel can handle T.38. Also this patch enable debug for T.38 gateway session if global fax debug enabled. ASTERISK-25982 Change-Id: I78387156ea521a77eb0faf170179ddd37a50430e 2016-05-01 02:21 +0000 [a2f19d82a8] Diederik de Groot * configs/basic-pbx/asterisk.conf: contains incorrect path separator Note: When packagers use these files (as an example) the paths are never really used when they are split using '='. Note: Thirdparty applications will also have trouble parsing the file when expecting '=>'. Change-Id: I0ada647f588e81f023fb1333ca15a1a333fd6004 2016-04-30 17:52 +0000 [f39089f17c] gtjoseph * pjproject_bundled: Various fixes discovered during testing of OSes For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c 2016-03-17 14:29 +0000 [8028fc7585] Andrew Nagy * app_voicemail: always copy dynamic struct to avoid race condition Voicemail email addresses can be corrupt or voicemail emails can end up being sent to the wrong email address if asterisk is reading voicemail.conf during a reload and processing an email at the same time. This patch always copies the struct that would otherwise only be copied once. ASTERISK-24463 #close Reported by: John Campbell Tested by: Etienne Lessard Tested by: Andrew Nagy Change-Id: I3a0643813116da84e2617291903d0d489b7425fb 2016-04-15 14:26 +0000 [3cb8934de0] Alexei Gradinari * pjsip: Added "reg_server" to contacts. If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8 2016-04-28 11:35 +0000 [7992923c70] Richard Mudgett * res_pjsip: Start body generator users after suppliers. Change-Id: I8f0b57841feaab56c8a4e821b5ccb4e05e5fbadb 2016-04-28 16:06 +0000 [5dc0e082b2] Richard Mudgett * res_pjsip_pubsub.c: Add useful information to some messages. Change-Id: Ia0b2e15773894c599e5c5748bbc70e99f434192a 2016-04-26 15:58 +0000 [f9e416f053] Richard Mudgett * res_pjsip_pubsub.c: Fix body generator registration race. Change-Id: Id8752073ef06472a2fd96080f4009fac42843e67 2016-04-26 15:13 +0000 [b1b2019046] Richard Mudgett * res_pjsip_pubsub.h: Fix doxygen association. Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632 2016-04-25 16:00 +0000 [b7f07fdff5] Richard Mudgett * res_pjsip_outbound_publish.c: Remove redundant flag check. Change-Id: I0da80a3c3e0eae0c52ff27e7412ba027d6f52353 2016-04-28 16:54 +0000 [719ece5659] gtjoseph * pjproject_bundled: Disable PJSIP_UNESCAPE_IN_PLACE When pjsip_parse_uri is called with PJSIP_UNESCAPE_IN_PLACE enabled, the input uri string will become corrupted if it contains escape sequences. It's not possible to automatically strdup or strdupa the input string because the output uri pj_str_t's will have pointers to chunks of the input string. Getting around this would require more memory management code and wouldn't be worth the savings of doing the unescape in place. ASTERISK-25970 #close Reported-by: Dmitriy Serov Change-Id: I28dc0e599b5108f7959b9c46dc8278371b372f88 2016-03-07 18:34 +0000 [38bed4515d] gtjoseph * res_pjsip: Add ability to identify by Authorization username A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d 2016-04-27 13:23 +0000 [677d5b5151] Mark Michelson * func_odbc: Check connection status before executing queries. A recent change to func_odbc made it so that a single connection was maintained per DSN. The problem was that the code was optimistic about the health of the connection after initially opening it and did nothing to re-connect in case the connection had died. This change adds a check before executing a query to ensure that the connection to the database is still up and running. ASTERISK-25963 #close Reported by Ross Beer Change-Id: Id33c86eb04ff48ca088bb2e3086c27b3b683491d 2016-04-15 11:59 +0000 [df3639700a] Alexei Gradinari * res_pjsip: disable multi domain to improve realtime performace This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7 2016-04-26 11:13 +0000 [949bf6b282] Joshua Colp * chan_sip: Give more time for TCP/TLS threads to stop. The unload process currently tells each TCP/TLS to terminate but does not wait for them to do so. This introduces a race condition where the container holding the threads may be destroyed before the threads are able to remove themselves from it. When they finally do the container is invalid and can't be used causing a crash. A previous change existed which waited a bit to wait for any stranglers to finish. This change extends this and waits longer. ASTERISK-25961 #close Change-Id: Idc6262b670ca49ede32061159e323b7b63c6f3c6 2016-04-26 05:48 +0000 [6959f5484b] Joshua Colp * app_queue: Fix crash when unloading module. When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b 2016-04-24 22:51 +0000 [b38f1146e5] gtjoseph * config: Fix ast_config_text_file_save2 writability check for missing files A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 2016-04-21 23:53 +0000 [29bab0d1a4] Kirill Katsnelson * chan_sip: Make autocreated peers send PeerStatus events Since Stasis has been introduced, an attempt to send AMI messages by an autocreated peer caused a crash, and all events from autocreated peers were semi-inadvertently disabled altogether in 0b83761. This change restores the disabled functionality. ASTERISK-25950 Change-Id: Iecc350f23db603fadb2f302064643ebe9664e974 2016-04-21 14:23 +0000 [c345e530f4] Kevin Harwell * app_queue: queue members can receive multiple calls It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 2016-04-25 08:11 +0000 [c0688a6398] Javier Acosta * Fix case sensitive actions in AMI QueueSummary and QueueStatus ASTERISK-25954 #close Reported by: Javier Acosta Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 2016-04-22 17:53 +0000 [eb7c581806] gtjoseph * res_agi: Prevent run_agi from eating frames it shouldn't The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645 2016-04-22 15:25 +0000 [ebf0724a83] Richard Mudgett * test_message.c: Wait longer in case dialplan also processes the test message. Bumped the wait from 1 second to 5 seconds. The test message was hitting my default call handler and failing the test because it took longer. Change-Id: I3a03737f25e92983de00548fcc7bbc50dd7544ba 2016-04-12 15:29 +0000 [ba63aa7c9e] Richard Mudgett * Manager: Short circuit AMI message processing. Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 2016-04-13 17:54 +0000 [d5ee6acf28] Richard Mudgett * manager.c: Eliminate most RAII_VAR usage. * Made ast_manager_event_blob_create() not allocate the ao2 event object with a lock as it is not needed. Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c 2016-04-13 17:09 +0000 [7303e3dc96] Richard Mudgett * manager_channels.c: Fix allocation failure crash. An earlier allocation failure failed to create a channel snapshot for the AMI HangupRequest/SoftHangupRequest event which resulted in a crash in channel_hangup_request_cb(). Where the stasis message gets generated cannot tell if the NULL snapshot returned was because of an allocation failure or the channel was a dummy channel. * Made channel_hangup_request_cb() check if the channel blob has a snapshot and exit if it doesn't. * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 2016-04-13 13:50 +0000 [1e93f3d723] Richard Mudgett * Bridge system: Fix memory leaks and double frees on impart failure. You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 2016-04-13 13:20 +0000 [5e388d4188] Richard Mudgett * bridge_softmix.c: Fix crash if channel fails to join mixing tech. softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b 2016-04-22 13:49 +0000 [068ae54c76] Mark Michelson * func_odbc: Use one connection per DSN. res_odbc was changed in Asterisk 13.8.0 to remove connection management, opting instead to let unixodbc maintain open connections and return those to Asterisk as requested. This was a boon for realtime, since it meant that multiple threads could potentially run parallel queries since they could each be using their own database connections. However, on the user-facing side, func_odbc, there were some inherent behaviors being relied on that no longer hold true after the change. One such reported behavior was that MySQL's LAST_INSERTED_ID() works per-connection. This means that if Asterisk uses separate connections for every database operation, whereas before it used one connection for everything, we have broken expectations and functionality. The fix provided in this patch is to make func_odbc use a single database connection per DSN. This way, user-facing database usage will have the same behavior as it did pre-13.8.0. However, realtime, which is the real workhorse of database interaction, will continue to let unixodbc manage connections. ASTERISK-25938 #close Reported by Edwin Vandamme Change-Id: Iac961fe79154c6211569afcdfec843c0c24c46dc 2016-04-22 13:02 +0000 [6aeefa89bc] Leif Madsen * Remove reference to non-existent sip.conf option Option was removed in commit 7f883ef495b57ae9182e47213d01d5e8009dbf3f ASTERISK-25927 #close Change-Id: I92f9b0196d9fc41d1d58354c07340c465ef1fcf8 2016-04-21 08:26 +0000 [e750ea9b5b] Diederik de Groot * lock.c: Check *lt before dereferencing it *lt is NULL if t->tracking == 0 ASTERISK-25948 #close Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba 2016-04-15 14:36 +0000 [a036c35903] Richard Mudgett * res_stasis: Handle re-enter stasis bridge with swap channel. We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee 2016-04-19 16:58 +0000 [9942d50aa5] Richard Mudgett * bridge: Hold off more than one imparting channel at a time. An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 2016-04-19 17:52 +0000 [516c626a7d] gtjoseph * res_pjsip_callerid: Clear out display name if id->name is not valid When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning the From header, then it overwrites the display name and uri from the channel's connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was leaving the display name from the From header in the new RPID or PAI header. On an attended transfer where the originator had a caller id number set but not a display name, the re-INVITE to the final transferee had the number of the originator but the display name of the transferer. Added a check to clear out the display name in the new header if connected.id.name was invalid. ASTERISK-25942 #close Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b 2016-04-19 13:02 +0000 [ded3794fc6] Joshua Colp * app_talkdetect: Make the module core supported. This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 2016-04-18 12:12 +0000 [efae187217] Mark Michelson * PJSIP: Remove PJSIP parsing functions from uri length validation. The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 2016-04-18 17:00 +0000 [f436b9ab11] Mark Michelson * res_pjsip_registrar: Fix bad memory-ness with user_agent. Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 2016-04-18 13:41 +0000 [49bfdc9ac0] Joshua Colp * res_pjsip_transport_management: Allow unload to occur. At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a 2016-04-15 11:41 +0000 [f4693d1897] Richard Mudgett * bridge_channel.c: Ignore role setup failure in channel push. We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 2016-04-17 15:37 +0000 [22335fe18a] Jaco Kroon * chan_sip: Don't verify table if rtupdate=no If rtupdate=no do not verify sipregs/peers table has updatable fields. ASTERISK-25934 #close Change-Id: Iaa2c53037b93daccc7e7333c40d61861847b856d 2016-04-18 04:53 +0000 [3b9d8b60b2] ibercom * app_queue: Frequent segfaults in function can_ring_entry() ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117 2016-04-15 16:51 +0000 [724acb6ce7] Richard Mudgett * stasis_bridge.c: Update stasis bridge push diagnostic messages. Change-Id: I195b14994c9dcccb9452491ca20a885d2a54605a 2016-04-14 13:49 +0000 [5f78801859] Mark Michelson * transport management: Register thread with PJProject. The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 2016-03-17 12:28 +0000 [9740277713] gtjoseph * res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c) There are several places that do scheduled tasks or periodic housecleaning, each with its own implementation: * res_pjsip_keepalive has a thread that sends keepalives. * pjsip_distributor has a thread that cleans up expired unidentified requests. * res_pjsip_registrar_expire has a thread that cleans up expired contacts. * res_pjsip_pubsub uses ast_sched directly and then calls ast_sip_push_task. * res_pjsip_sdp_rtp also uses ast_sched to send keepalives. There are also places where we should be doing scheduled work but aren't. A good example are the places we have sorcery observers to start registration or qualify. These don't work when changes are made to a backend database without a pjsip reload. We need to check periodically. As a first step to solving these issues, a new ast_sip_sched facility has been created. ast_sip_sched wraps ast_sched but only uses ast_sched as a scheduled queue. When a task is ready to run, ast_sip_task_pusk is called for it. This ensures that the task is executed in a PJLIB registered thread and doesn't hold up the ast_sched thread so it can immediately continue processing the queue. The serializer used by ast_sip_sched is one of your choosing or a random one from the res_pjsip pool if you don't choose one. Another feature is the ability to automatically clean up the task_data when the task expires (if ever). If it's an ao2 object, it will be dereferenced, if it's a malloc'd object it will be freed. This is selectable when the task is scheduled. Even if you choose to not auto dereference an ao2 task data object, the scheduler itself maintains a reference to it while the task is under it's control. This prevents the data from disappearing out from under the task. There are two scheduling models. AST_SIP_SCHED_TASK_PERIODIC specifies that the invocations of the task occur at the specific interval. That is, every "interval" milliseconds, regardless of how long the task takes. If the task takes longer than the interval, it will be scheduled at the next available multiple of interval. For exmaple: If the task has an interval of 60 secs and the task takes 70 secs (it better not), the next invocation will happen at 120 seconds. AST_SIP_SCHED_TASK_DELAY specifies that the next invocation of the task should start "interval" milliseconds after the current invocation has finished. Also, the same ast_sched facility for fixed or variable intervals exists. The task's return code in conjunction with the AST_SIP_SCHED_TASK_FIXED or AST_SIP_SCHED_TASK_VARIABLE flags controls the next invocation start time. One res_pjsip.h housekeeping change was made. The pjsip header files were added to the top. There have been a few cases lately where I've needed res_pjsip.h just for ast_sip calls and had compiles fail spectacularly because I didn't add the pjsip header files to my source even though I never referenced any pjsip calls. Finally, a few new convenience APIs were added to astobj2 to make things a little easier in the scheduler. ao2_ref_and_lock() calls ao2_ref() and ao2_lock() in one go. ao2_unlock_and_unref() does the reverse. A few macros were also copied from res_phoneprov because I got tired of having to duplicate the same hash, sort and compare functions over and over again. The AO2_STRING_FIELD_(HASH|SORT|CMP)_FN macros will insert functions suitable for aor_container_alloc into your source. This facility can be used immediately for the situations where we already have a thread that wakes up periodically or do some scheduled work. For the registration and qualify issues, additional sorcery and schema changes would need to be made so that we can easily detect changed objects on a periodic basis without having to pull the entire database back to check. I'm thinking of a last-updated timestamp on the rows but more on this later. Change-Id: I7af6ad2b2d896ea68e478aa1ae201d6dd016ba1c 2016-03-08 12:12 +0000 [7fb3724a77] Mark Michelson * res_pjsip_transport_management: Kill idle TCP connections. "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 2016-03-08 10:52 +0000 [707fd4dcd0] Mark Michelson * Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433 2016-04-14 07:15 +0000 [0b4bb19e0b] Mark Michelson * AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55 2016-04-12 13:10 +0000 [f6e080c6a4] Richard Mudgett * bridge_softmix.c: Fix crash if could not allocate the dsp. Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372 2016-04-12 15:41 +0000 [cf15a2f2d3] gtjoseph * pjproject: Add patch for removing strip of '[]' from header params From the patch submitted to Teluu on 4/12/2016 <<<<<<<<< The wholesale stripping of '[]' from header parameters causes issues if something (like a port) occurs after the final ']'. '[2001:a::b]' will correctly parse to '2001:a::b' '[2001:a::b]:8080' will correctly parse to '2001:a::b' but the scanner is left with ':8080' and parsing stops with a syntax error. I can't even find a case where stripping the '[]' is a good thing anyway. Even if you continued to parse and resulted in a string that looks like this... '2001:a::b:8080', it's not valid. This came up in Asterisk because Kamailio sends us a Contact with an alias URI parameter that has an IPv6 address in it like this: Contact: which should be legal but causes a syntax error because of the characters after the final ']'. Even if it didn't, the '[]' should still not be stripped. I've run the Asterisk Test Suite for PJSIP (252 tests) many of which are IPv6 enabled. No issues were caused by removing the code that strips the '[]'. >>>>>>>>>>> ASTERISK-25123 #close Reported-by: Anthony Messina Change-Id: I5cb33f4ebf07ee1f2b26d07caae715e2ec65595a 2016-04-12 09:10 +0000 [daa086fae4] Joshua Colp * app_voicemail: Fix test_voicemail_notify_endl test. The test_voicemail_notify_endl test checks the end-of-line characters of an email message to confirm that they are consistent. The test wrongfully assumed that reading from the email message into a buffer will always result in more than 1 character being read. This is incorrect. If only 1 character was read the test would go outside of the buffer and access other memory causing a crash. The test now checks to ensure that 2 or more characters are read in ensuring the test stays within the buffer. ASTERISK-25874 #close Change-Id: Ic2c89cea6e90f2c0bc2d8138306ebbffd4f8b710 2016-04-07 12:02 +0000 [f896136460] Alexei Gradinari * app_voicemail/IMAP: function 'save_to_folder' creates wrong folder If try to move message to Cust1 (number 5) the function 'save_to_folder' tries to create Greeting folder instead of Cust1. This patch fixed it by setting GREETINGS_FOLDER = -1 ASTERISK-24927 #close Change-Id: I03d1a761894bcc2d130ec9b003bbcddc28e25c51 2016-04-07 16:18 +0000 [70b7673f09] Alexei Gradinari * res_pjsip: Add headers to AMI Event ContactStatusDetail * Added Useragent and RegExpire headers to AMI Event ContactStatusDetail with associated documentation. ASTERISK-25903 #close Change-Id: If3d121e943e588d016ba51d4eb9c6a421a562239 2016-04-11 16:20 +0000 [64ecd41c8f] Alexei Gradinari * Codecs: strip codec name while parsing allow/disallow options Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d 2016-04-11 14:26 +0000 [3f6c4667b8] Jaco Kroon * core_unreal: Fix hangupcauses not getting set on Local channels ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa 2016-04-01 13:30 +0000 [fe7e48db03] gtjoseph * res_pjsip contact: Lock expiration/addition of contacts Contact expiration can occur in several places: res_pjsip_registrar, res_pjsip_registrar_expire, and automatically when anyone calls ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar may also be attempting to renew or add a contact. Since none of this was locked it was possible for one thread to be renewing a contact and another thread to expire it immediately because it was working off of stale data. This was the casue of intermittent registration/inbound/nominal/multiple_contacts test failures. Now, the new named lock functionality is used to lock the aor during contact expire and add operations and res_pjsip_registrar_expire now checks the expiration with the lock held before deleting the contact. ASTERISK-25885 #close Reported-by: Josh Colp Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059 2016-04-10 14:16 +0000 [0c414eaf35] gtjoseph * pjproject: Add patch to fix Via IPv6 parsing There's a bug in pjproject's sip_parser where the ":" wasn't correctly interpreted. This is causing IPv6 addresses in the "received" parameter of the Via header to cause a syntax check failure. This patch was submitted to Teluu on 4/10/2016. ASTERISK-25910 #close Reported-by: Anthony Messina Change-Id: Ic7e4c4aa14ded61860401ec349f5177568c4d922 2016-03-31 20:04 +0000 [772ff3048f] gtjoseph * lock: Add named lock capability Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 2016-04-05 16:56 +0000 [fd601f26f7] Alexei Gradinari * res_pjsip_outbound_publish: Add transport for outbound PUBLISH The first available transport of the appropriate type is used now. This patch adds new config option 'transport' for outbound-publish. If transport is set then outbound PUBLISH requests will use this transport. ASTERISK-25901 #close Change-Id: Ib389130489b70e36795b0003fa5fd386e2680151 2016-04-07 16:39 +0000 [5f768d2a9c] Alexei Gradinari * res_pjsip_dialog_info: Add missing "direction" attribute in NOTIFY event BLF pickup isn't working on Cisco SPA and Snom phones if the direction="recipient" attribute is missing in 'dialog' tag. This patch adds direction="recipient" if extension state is Ringing. ASTERISK-24601 #close Change-Id: I5b2c097ca29fd59e92ba237ca5d397cb1b0bcd8c 2016-04-07 10:59 +0000 [82638fb0c7] Richard Mudgett * pbx.c: Minor code rearangements. * Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 2016-04-07 11:37 +0000 [bc320df173] Alexei Gradinari * app_voicemail/IMAP: IMAP access FATAL error: Out of memory Sometimes uw-imap function 'mail_fetchbody' returns huge len which then pass to uw-imap function 'rfc822_base64'. uw-imap tries to allocate huge memory and abort() on fail. This patch check the len. If the len more than max size (128 Mbytes) log error. This patch also set variables len, newlen to avoid uninizialezed len. This patch also check pointer returned by rfc822_base64. ASTERISK-25899 #close Change-Id: I4a0e7d655f11abef6a5224e2169df6d5c1f1caca 2016-04-07 12:26 +0000 [2ef8a954b3] Richard Mudgett * pbx: Update doxygen for extension state watchers. Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e 2016-04-07 11:49 +0000 [d312fdeb1b] gtjoseph * alembic: Remove batch operations (and sqlite support) Because SQLite doesn't support full ALTER capabilities, alembic scripts require batch operations. However, that capability wasn't available until 0.7.0 which some distributions haven't reached yet. Therefore, the batch operations introduced in commit 86d6e44cc (review 2319) have been reverted and SQLite is unsupported again, for now anyway. Tested the full upgrade and downgrade on MySQL/Mariadb and Postgresql. ASTERISK-25890 #close Reported-by: Harley Peters Change-Id: I82eba5456736320256f6775f5b0b40133f4d1c80 2016-04-07 11:05 +0000 [901e8d78c4] Joshua Colp * res_pjsip_registrar_expire: Fix race condition at shutdown. When shutting down, the PJSIP sorcery is destroyed. The registrar expiration module queries the PJSIP sorcery to determine what to expire. As there was no synchronization between termination of the expiration thread and the unloading of the module it was possible for the thread to try to access the PJSIP sorcery after it had been destroyed. This change ensures that the thread is shut down before allowing the module to be considered unloaded. Change-Id: I69fd239edbaaf160c2d37ae00d3ac06e5596fe8b 2016-04-06 16:28 +0000 [8207372e66] Joshua Colp * res_pjsip: Fix configuration setting of "regcontext". Due to a merge problem two options were swapped causing the regcontext setting to not get set. Change-Id: Icb33edc668e7357bacbaec2861a6b5ac64edaff1 2016-04-06 08:01 +0000 [0735a4d6d7] Jacek Konieczny * frame.c: Copy the whole subclass in ast_frdup(). The problem is ast_frdup() does not copy whole frame.subclass for voice, video and image frames, only the format is copied. For video frames, the subclass structure contains the .frame_ending flag used to put the RTP marker where it needs to be. ASTERISK-25894 #close Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33 2016-04-05 14:23 +0000 [c61dca6419] Mark Michelson * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. At this point, PJMedia's SDP negotiator saves Asterisk's local state as being recvonly. Now, when the device attempts to unhold, it again uses a deferred SDP reinvite, so we end up doing the following: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating recvonly on streams * Device sends ACK with SDP indicating sendonly on streams The problem here is that Asterisk offered recvonly, and by RFC 3264's rules, if an offer is recvonly, the answer has to be sendonly. The result is that the device is not taken off hold. What is supposed to happen is that Asterisk should indicate sendrecv in the 200 OK that it sends. This way, the device has the freedom to indicate sendrecv if it wants the stream taken off hold, or it can continue to respond with sendonly if the purpose of the reinvite was something else (like a session timer refresher). The fix here is to alter the SDP negotiator's state when we receive a reinvite with no SDP. If the negotiator's state is currently in the recvonly or inactive state, then we alter our local state to be sendrecv. This way, we allow the device to indicate the stream state as desired. ASTERISK-25854 #close Reported by Robert McGilvray Change-Id: I7615737276165eef3a593038413d936247dcc6ed 2016-03-27 23:33 +0000 [50b0922a22] gtjoseph * config: Allow filters when appending to a category In sorcery based config files where there are multiple categories with the same name, you can't use the (+) operator to reliably append to a category because config.c stops looking when it finds the first one with the same name. Example: [1000] type = endpoint [1000] type = aor [1000](+) authenticate_qualify = yes This config will fail because config.c appends authenticate_qualify to the first category it finds, the endpoint, and that's not valid for endpoint. Solution: The capability to find a category that contains a certain variable already exists so the only real change was to parse anything after the '+' that's not a comma, as a filter string. [1000] type = endpoint [1000] type = aor [1000](+type=aor) authenticate_qualify = yes This now works as expected. Although the following example doesn't make any sense for pjsip, you can even specify multiple filters: [1000](+type=aor&qualify_frequency=10) ASTERISK-25868 #close Reported-by: Nick Repin Change-Id: I10773da4c79db36fbf1993961992af63d3441580 2016-04-05 10:21 +0000 [cb56ef8069] Joshua Colp * res_http_websocket: Make core supported. Websockets are a core part of ARI support and as such this module should also be core supported. Change-Id: I8f9283c6a167152761b92984779bb39e3db51a9c 2016-03-25 23:22 +0000 [f6f4cf459f] gtjoseph * stringfields: Refactor to allow fields to be added to the end of structures String fields are great, except that you can't add new ones without breaking ABI compatibility because it shifts down everything else in the structure. The only alternative is to add your own char * field to the end of the structure and manage the memory yourself which isn't ideal, especially since you then can't use the OPT_STRINGFIELD_T type. Background: The reason string fields had to be declared inside the AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS declared the pool, then the fields, then the manager, you could use the offsets of the pool and manager and iterate over the sequential addresses in between to access the fields. The actual pool, field allocation and field set operations don't actually care where the field is. It's just iteration over the fields that was the problem. Solution: Extended String Fields An extended string field is one that is declared outside the AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent structure. Other than using AST_STRING_FIELD_EXTENDED instead of AST_STRING_FIELD, it looks the same as other string fields. It's storage comes from the pool and it participates in string field compare and copy operations peformed on the parent structure. It's also a valid target for the OPT_STRINGFIELD_T aco option type. Implementation: To keep track of the extended fields and make sure that ABI isn't broken, the existing embedded_pool pointer in the manager structure was repurposed to be a pointer to a separate header structure that contains the embedded_pool pointer plus a vector of fields. The length of the manager structure didn't change and the embedded_pool pointer isn't used in the macros, only the stringfields C code. A side benefit of this is that changing the header structure in the future won't break ABI. ast_string_fields_init initializes the normal string fields and appends them to the vector, and subsequent calls to ast_string_field_init_extended initialize and append the extended fields. Cleanup, ast_string_fields_cmp, and ast_string_fields_copy can now work on the vector instead of sequentially traversing the addresses between the pool and manager. The total size of a structure using string fields didn't change, whether using extended fields or not, nor have the offsets of any structure members, either inside the original block or outside. Adding an extended field to the end of a structure is the same as adding a char *. Details: The stringfield C code was pulled out from utils.c and into stringfields.c. It just made sense. Additional work was done in ast_string_field_init and ast_calloc_with_stringfields to handle the allocation of the new header structure and the vector, and the associated cleanup. In the process some additional NULL pointer checking was added. A lot of work was done in stringfields.h since the logic for compare and copy is there. Documentation was added as well as somne additional NULL checking. The ability to call ast_calloc_with_stringfields with a number of structures greater than 1 never really worked. Well, the calloc worked but there was no way to access the additional structures or clean them up. It was agreed that there was no use case for requesting more than 1 structure so an ast_assert was added to prevent it and the iteration code removed. Testing: The stringfield unit tests were updated to test both normal and extended fields. Tests for ast_string_field_ptr_set_by_fields and ast_calloc_with_stringfields were also added. As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except res_pjsip itself, saved off. The patch was then added and a full compile and install was performed. Then the older res_pjsip_* moduled were copied over the installed versions so res_pjsip was new and the rest were old. No issues. contact->aor, which is a char * at the end of contact, was then changed to an extended string field and a recompile and reinstall was performed, again leaving stock versions of the the res_pjsip_* modules. Again, no issues with the res_pjsip_* modules using the old stringfield implementation and with contact->aor as a char *, and res_pjsip itself using the new stringfield implementation and contact->aor being an extended string field. Finally, several existing string fields were converted to extended string fields to test OPT_STRINGFIELD_T. Again, no issues. Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61 2016-04-04 18:02 +0000 [fe448ac8a7] gtjoseph * res_pjsip_mwi: Fix segv caused by 16c7d8e74a9af13f98c3c22aa9c43ce39965f6b7 I forgot the new voicemail_extension wasn't a stringfield and didn't check for NULL where I should have. Change-Id: I029482d5c2ab72474838750461bd46b0809c90fb 2016-04-03 11:47 +0000 [9d4318f798] gtjoseph * install_prereq: Fix check_installed_debs remove subversion check_installed_debs wasn't handling virtual packages like libsrtp-dev and libresample-dev and on multiarch systems it was accidentally filtering out all packages if any :i386 packages were found instead of just filtering out the :i386 packages themselves. Change-Id: Ifd68da0d1ee30cc84df14de3f9b9079d7c3cecda 2016-04-01 13:09 +0000 [566601837e] gtjoseph * utils.c: Fix typo in handle_show_locks ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e). Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf 2016-03-30 18:34 +0000 [964f54bd5d] gtjoseph * pjproject_bundled: Fix use of LDCONFIG for shared library link creation LDCONFIG apparently isn't set to something sane on all systems so the creation of the shared library links fails. Instead of just testing for non-blank, main/Makefile now checks that LDCONFIG is actually executable and reverts to LN if it isn't. This applies to both libasteriskpj and libasteriskssl. Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG. ASTERISK-25873 #close Reported-by: Hans van Eijsden Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729 2016-03-30 13:31 +0000 [5f73c2ef0a] Richard Mudgett * res_stasis.c: Protect channel datastore list from stasis end. Change-Id: Ifadc469590bd4d5368e19d3763db3bd1f80fdb95 2016-03-29 18:06 +0000 [74d63f56ee] Richard Mudgett * res_ari: Cannot get control also means channel is unavailable. The only caller of ari_bridges_play_found() has this note: If ari_bridges_play_found fails because the channel is unavailable for playback, The channel will be removed from the playback list soon. We can keep trying to get channels from the list until we either get one that will work or else there isn't a channel for this bridge anymore, in which case we'll revert to ari_bridges_play_new. Change-Id: Ib068141b367ccaa17be0dab4181c98e26c5127d6 2016-03-29 14:29 +0000 [cf49b44090] Richard Mudgett * res_stasis_recording.c: Cleanup stasis_app_recording_find_by_name(). Change-Id: Ic7d93c402c498677a122505558859c853d4e5ac7 2016-03-28 14:23 +0000 [7f53f1d89e] Richard Mudgett * core_unreal.c: Add clarification comment about channel ref. Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3 2016-03-29 13:47 +0000 [ecf4102d02] Richard Mudgett * res_stasis: Add control ref to playback and recording structs. The stasis_app_playback and stasis_app_recording structs need to have a struct stasis_app_control ref. Other threads can get a reference to the playback and recording structs from their respective global container. These other threads can then use the control pointer they contain after the control struct has gone. * Add control ref to stasis_app_playback and stasis_app_recording structs. With the refs added, the control command queue can now have a circular control reference which will cause the control struct to never get released if the control's command queue is not flushed when the channel leaves the Stasis application. Also the command queue needs better protection from adding commands if the control->is_done flag is set. * Flush the control command queue on exit. ASTERISK-25882 #close Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d 2016-03-28 18:10 +0000 [a179aba65e] Richard Mudgett * res_stasis: Fix crash on a hanging up channel. * Give the struct stasis_app_control ao2 object a ref to the channel held in the object. Now the channel will still be around if a thread needs to post a stasis message instead of crash because the topic was destroyed. * Moved stopping any lingering silence generator out of the struct stasis_app_control destructor and made it a part of exiting the Stasis application. Who knows which thread the destructor will be called under so it cannot affect the channel's silence generator. Not only was the channel unprotected when the silence generator was stopped, stasis may no longer even control the channel. ASTERISK-25882 Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4 2016-03-30 12:38 +0000 [16c7d8e74a] gtjoseph * res_pjsip_mwi: Allow subscribe to vm access extension as an alias Background: If your extension is 1000 and the voicemail access extension is 1571 and you dial 1571, usually a dialplan rule calls voicemailmain with your extension and you are placed directly in your mailbox. Therefore most admins program the voicemail (or other speed dial) button on their phones to the access extension. Some phones (Snom at least) use whatever is programmed there to also subscribe for MWI and so can't dial one number and subscribe to another. This works fine in chan_sip because chan_sip completely ignores the user portion of the SUBSCRIBE message request URI. If it can match the peer, is subscribes to the peer's mailbox. The user could be set to anything or nothing and you'd still get subscribed to your mailbox. Issue: chan_pjsip actually uses the user portion of the URI to find an aor and its mailboxes. Therefore a subscribe to 1571 results in a 404. Sure, you can create an aor for 1571 but you certainly can't add your entire voicemail system's mailboxes to it and everyone would get notified of every MWI. Solution: When an MWI subscribe comes in and an aor can't be found that matches the resource directly, check the resource against the endpoint's aors. If an aor is found that has a voicemail_extension that matches the resource, use it. ASTERISK-25865 Reported-by: Ross Beer Change-Id: I770ea185f751f1ada888fafb4b452115f1c06e9e 2016-03-24 22:55 +0000 [d8f0bc3572] gtjoseph * res_pjsip_mwi: Add voicemail extension and mwi_subscribe_replaces_unsolicited res_pjsip_mwi was missing the chan_sip "vmexten" functionality which adds the Message-Account header to the MWI NOTIFY. Also, specifying mailboxes on endpoints for unsolicited mwi and on aors for subscriptions required that the admin know in advance which the client wanted. If you specified mailboxes on the endpoint, subscriptions were rejected even if you also specified mailboxes on the aor. Voicemail extension: * Added a global default_voicemail_extension which defaults to "". * Added voicemail_extension to both endpoint and aor. * Added ast_sip_subscription_get_dialog for support. * Added ast_sip_subscription_get_sip_uri for support. When an unsolicited NOTIFY is constructed, the From header is parsed, the voicemail extension from the endpoint is substituted for the user, and the result placed in the Message-Account field in the body. When a subscribed NOTIFY is constructed, the subscription dialog local uri is parsed, the voicemail_extension from the aor (looked up from the subscription resource name) is substituted for the user, and the result placed in the Message-Account field in the body. If no voicemail extension was defined, the Message-Account field is not added to the NOTIFY body. mwi_subscribe_replaces_unsolicited: * Added mwi_subscribe_replaces_unsolicited to endpoint. The previous behavior was to reject a subscribe if a previous internal subscription for unsolicited MWI was found for the mailbox. That remains the default. However, if there are mailboxes also set on the aor and the client subscribes and mwi_subscribe_replaces_unsolicited is set, the existing internal subscription is removed and replaced with the external subscription. This allows an admin to configure mailboxes on both the endpoint and aor and allows the client to select which to use. ASTERISK-25865 #close Reported-by: Ross Beer Change-Id: Ic15a9415091760539c7134a5ba3dc4a6a1217cea 2016-03-30 09:46 +0000 [8dc8d6ceb8] gtjoseph * res_rtp_asterisk: Fix placement of txcount increment Commit 1bce690ccb36a4744a327c07af23a9a3a0fa20cd was incrementing txcount for rtcp packets as well as rtp packets and that was causing sender reports to be generated instead of receiver reports in cases where no rtp was actually being sent. Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, to rtp_sento which only handles rtp packets. Discovered by the hep/rtcp-receiver test. Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5 2016-03-26 22:33 +0000 [c7eb18d865] gtjoseph * chan_pjsip: Add 'pjsip show channelstats' Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869 2016-03-10 19:52 +0000 [1583559a06] gtjoseph * res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS No one seemed to notice but every time an OPTIONS goes out, it goes out with a From of "asterisk" (or whatever the default from_user is set to), even if you specify an endpoint. The issue had several causes... qualify_contact is only called with an endpoint if called from the CLI. If the endpoint is NULL, qualify_contact only looks up the endpoint if authenticate_qualify=yes. Even then, it never passes it on to ast_sip_create_request where the From header is set. Therefore From is always "asterisk" (or whatever the default from_user is set to). Even if ast_sip_create_request were to get an endpoint, it only sets the From if endpoint->from_user is set. The fix is 4 parts... First, create_out_of_dialog_request was modified to use the endpoint id if endpoint was specified and from_user is not set. Second, qualify_contact was modified to always look up an endpoint if one wasn't specified regardless of authenticate_qualify. It then passes the endpoint on to create_out_of_dialog_request. Third (and most importantly), find_an_endpoint was modified to find an endpoint by using an "aors LIKE %contact->aor%" predicate with ast_sorcery_retrieve_by_fields. As such, this patch will only work if the sorcery realtime optimizations patch goes in. Otherwise we'd be pulling the entire endpoints database every time we send an OPTIONS. Since we already know the contact's aor, the on_endpoint callback was also modified to just check if the contact->aor is an exact match to one of the endpoint's. Finally, since we now have an endpoint for every OPTIONS request, res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was updated to get the transport from the endpoint and set it on tdata. Now the correct transport is used. Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af 2016-03-25 10:59 +0000 [0cfab30b28] Jacek Konieczny * res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 explicitly states: There MUST be a separate DTLS-SRTP session for each distinct pair of source and destination ports used by a media session This means RTP keying material cannot be used for DTLS RTCP, which was the reason why RTCP encryption would fail. ASTERISK-25642 Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a 2016-03-25 10:42 +0000 [6a9c18fb59] Jacek Konieczny * app_echo: forward and generate VIDUPDATE frames When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e 2016-03-27 12:53 +0000 [1bce690ccb] gtjoseph * res_rtp_asterisk: Fix packet stats on bridged connection rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated for bridged streams because the calulations were being done after the bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. Moved the calculations so they occur for all valid received packets and all transmitted packets. Also added rxoctetcount and txoctetcount to ast_rtp_instance_stat. Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb 2016-03-25 23:19 +0000 [50f90d4099] Philip Correia * res_parking: Fix blind transfer dynamic lots creation. Blind transfers to a recognized parking extension need to use the parker's channel variable values to create the dynamic parking lot. This is because there is always only one parker while the parkee may actually be a multi-party bridge. A multi-party bridge can never supply the needed channel variables to create the dynamic parking lot. In the multi-party bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and channel variables are inherited by the local channel used to park the bridge. * In park_common_setup(), make use the parker instead of the parkee to supply the dynamic parking lot channel variable values. In all but one case, the parkee is the same as the parker. However, in the recognized parking extension blind transfer scenario for a two party bridge they are different channels. For consistency, we need to use the parker channel. * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the local channel when blind transferring a multi-party bridge to a recognized parking extension. * When a local channel starts a call, the Local;2 side needs to inherit the CHANNEL(parkinglot) value from Local;1. The DTMF one-touch parking case wasn't even trying to create dynamic parking lots before it aborted the attempt. * In parking_park_call(), add missing code to create a dynamic parking lot. A DTMF bridge hook is documented as returning -1 to remove the hook. Though the hook caller is really coded to accept non-zero. See the ast_bridge_hook_callback typedef. * In feature_park_call(), don't remove the DTMF one-touch parking hook because of an error. ASTERISK-24605 #close Reported by: Philip Correia Patches: call_park.patch (license #6672) patch uploaded by Philip Correia Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9 2016-03-08 15:55 +0000 [5aa5c49413] gtjoseph * sorcery/res_pjsip: Refactor for realtime performance There were a number of places in the res_pjsip stack that were getting all endpoints or all aors, and then filtering them locally. A good example is pjsip_options which, on startup, retrieves all endpoints, then the aors for those endpoints, then tests the aors to see if the qualify_frequency is > 0. One issue was that it never did anything with the endpoints other than retrieve the aors so we probably could have skipped a step and just retrieved all aors. But nevermind. This worked reasonably well with local config files but with a realtime backend and thousands of objects, this was a nightmare. The issue really boiled down to the fact that while realtime supports predicates that are passed to the database engine, the non-realtime sorcery backends didn't. They do now. The realtime engines have a scheme for doing simple comparisons. They take in an ast_variable (or list) for matching, and the name of each variable can contain an operator. For instance, a name of "qualify_frequency >" and a value of "0" would create a SQL predicate that looks like "where qualify_frequency > '0'". If there's no operator after the name, the engines add an '=' so a simple name of "qualify_frequency" and a value of "10" would return exact matches. The non-realtime backends decide whether to include an object in a result set by calling ast_sorcery_changeset_create on every object in the internal container. However, ast_sorcery_changeset_create only does exact string matches though so a name of "qualify_frequency >" and a value of "0" returns nothing because the literal "qualify_frequency >" doesn't match any name in the objset set. So, the real task was to create a generic string matcher that can take a left value, operator and a right value and perform the match. To that end, strings.c has a new ast_strings_match(left, operator, right) function. Left and right are the strings to operate on and the operator can be a string containing any of the following: = (or NULL or ""), !=, >, >=, <, <=, like or regex. If the operator is like or regex, the right string should be a %-pattern or a regex expression. If both left and right can be converted to float, then a numeric comparison is performed, otherwise a string comparison is performed. To use this new function on ast_variables, 2 new functions were added to config.c. One that compares 2 ast_variables, and one that compares 2 ast_variable lists. The former is useful when you want to compare 2 ast_variables that happen to be in a list but don't want to traverse the list. The latter will traverse the right list and return true if all the variables in it match the left list. Now, the backends' fields_cmp functions call ast_variable_lists_match instead of ast_sorcery_changeset_create and they can now process the same syntax as the realtime engines. The realtime backend just passes the variable list unaltered to the engine. The only gotcha is that there's no common realtime engine support for regex so that's been noted in the api docs for ast_sorcery_retrieve_by_fields. Only one more change to sorcery was done... A new config flag "allow_unqualified_fetch" was added to reg_sorcery_realtime. "no": ignore fetches if no predicate fields were supplied. "error": same as no but emit an error. (good for testing) "yes": allow (the default); "warn": allow but emit a warning. (good for testing) Now on to res_pjsip... pjsip_options was modified to retrieve aors with qualify_frequency > 0 rather than all endpoints then all aors. Not only was this a big improvement in realtime retrieval but even for config files there's an improvement because we're not going through endpoints anymore. res_pjsip_mwi was modified to retieve only endpoints with something in the mailboxes field instead of all endpoints then testing mailboxes. res_pjsip_registrar_expire was completely refactored. It was retrieving all contacts then setting up scheduler entries to check for expiration. Now, it's a single thread (like keepalive) that periodically retrieves only contacts whose expiration time is < now and deletes them. A new contact_expiration_check_interval was added to global with a default of 30 seconds. Ross Beer reports that with this patch, his Asterisk startup time dropped from around an hour to under 30 seconds. There are still objects that can't be filtered at the database like identifies, transports, and registrations. These are not going to be anywhere near as numerous as endpoints, aors, auths, contacts however. Back to allow_unqualified_fetch. If this is set to yes and you have a very large number of objects in the database, the pjsip CLI commands will attempt to retrive ALL of them if not qualified with a LIKE. Worse, if you type "pjsip show endpoint " guess what's going to happen? :) Having a cache helps but all the objects will have to be retrieved at least once to fill the cache. Setting allow_unqualified_fetch=no prevents the mass retrieve and should be used on endpoints, auths, aors, and contacts. It should NOT be used for identifies, registrations and transports since these MUST be retrieved in bulk. Example sorcery.conf: [res_pjsip] endpoint=config,pjsip.conf,criteria=type=endpoint endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error ASTERISK-25826 #close Reported-by: Ross Beer Tested-by: Ross Beer Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67 2016-03-23 14:24 +0000 [05fc3a96d1] Richard Mudgett * res_parking: Cleanup find_channel_parking_lot_name() usage. Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02 2016-03-18 14:01 +0000 [a4189763ab] Richard Mudgett * res_parking: Misc fixes. res/parking/parking_applications.c: * Add malloc fail checks in setup_park_common_datastore(). * Fix playing parking failed announcement to only happen on non-blind transfers in park_app_exec(). It could never go out before because a test was provedly always false. res/parking/parking_bridge.c: * Fix NULL tolerance in generate_parked_user() because bridge_parking_push() can theoretically pass a NULL parker channel if the parker channel went away for some reason. * Clarify some weird code dealing with blind_transfer in bridge_parking_push(). res/parking/parking_bridge_features.c: * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel which will be bulk copied to the Local;2 channel on the subsequent ast_call(). The additional advantage is if the parker channel has the BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed to be overridden. res/parking/parking_manager.c: * Fix AMI Park action input range checking of the Timeout header in manager_park(). * Reduced locking scope to where needed in manager_park(). res/res_parking.c: * Fix some off nominal missing unlocks by eliminating the returns. Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca 2014-12-15 05:23 +0000 [6f95b5eda1] Philip Correia * res_parking: Update parking documentation for dynamic parking lots. * Remove duplicate res_parking.conf courtesytone config option documentation. ASTERISK-24596 #close Reported by: Philip Correia ASTERISK-24605 Reported by: Philip Correia Patches: call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5 2016-03-24 14:08 +0000 [81ce60f6d4] Alexander Traud * chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66 2016-03-19 07:34 +0000 [c6e4c48e67] Gianluca Merlo * config: fix flags in uint option handler The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e 2016-03-10 16:58 +0000 [59c8e189fd] Mark Michelson * Restrict CLI/AMI commands on shutdown. During stress testing, we have frequently seen crashes occur because a CLI or AMI command attempts to access information that is in the process of being destroyed. When addressing how to fix this issue, we initially considered fixing individual crashes we observed. However, the changes required to fix those problems would introduce considerable overhead to the nominal case. This is not reasonable in order to prevent a crash from occurring while Asterisk is already shutting down. Instead, this change makes it so AMI and CLI commands cannot be executed if Asterisk is being shut down. For AMI, this is absolute. For CLI, though, certain commands can be registered so that they may be run during Asterisk shutdown. ASTERISK-25825 #close Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990 2016-03-24 07:45 +0000 [ff3eebf454] Walter Doekes * musiconhold: Only warn if music class is not found in memory and database. The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf 2016-03-24 05:38 +0000 [82e55e4883] Walter Doekes * core/logging: Fix broken syslog levels on older glibc. The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However this macro is broken in older glibc (< 2.17); it would left-shift the facility a second time, causing the resultant priority to become invalid. The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: The priority argument is formed by ORing the facility and the level values [...]. ASTERISK-25510 #close Reported by: Michael Newton Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03 2016-03-23 08:59 +0000 [d963a33749] gtjoseph * pjproject-bundled: Cleanups for reported issues PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd 2015-11-20 08:02 +0000 [c5170677e7] Francesco Castellano * chan_sip.c: Space after port causes unnecessary resolution attempt check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06 2016-03-19 17:49 +0000 [51deadee38] gtjoseph * progdocs: Exclude ./third-party from documentation generation We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17 2016-03-18 20:32 +0000 [aa2fcd244e] Gianluca Merlo * func_aes: fix misuse of strlen on binary data The encryption code for AES_ENCRYPT evaluates the length of the data to be encoded in base64 using strlen. The data is binary, thus the length of it can be underestimated at the first NULL character. Reuse the write pointer offset to evaluate it, instead. ASTERISK-25857 #close Change-Id: If686b5d570473eb926693c73461177b35b13b186 2016-12-08 17:42 +0000 Asterisk Development Team * asterisk certified/13.8-cert4 Released. 2016-12-08 11:40 +0000 [2e0239c28c] Kevin Harwell * Update for certified/13.8-cert4 2016-11-30 09:31 +0000 [4fece22836] Walter Doekes * 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] 1*WSP ; 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 2016-08-06 10:57 +0000 [016d20ce12] Alexei Gradinari * pjsip: Fix deadlock with suspend taskprocessor on masquerade If both channels which should be masqueraded are in the same serializer: 1st channel will be locked waiting condition 'complete' 2nd channel will be locked waiting condition 'suspended' On heavy load system a chance that both channels will be in the same serializer 'pjsip/distibutor' is very high. To reproduce compile res_pjsip/pjsip_distributor.c with DISTRIBUTOR_POOL_SIZE=1 Steps to reproduce: 1. Party A calls Party B (bridged call 'AB') 2. Party B places Party A on hold 3. Party B calls Voicemail app (non-bridged call 'BV') 4. Party B attended transfers Party A to voicemail using REFER. 5. When asterisk masquerades calls 'AB' and 'BV', a deadlock is happened. This patch adds a suspension indicator to the taskprocessor. When a session suspends/unsuspends the serializer it sets the indicator to the appropriate state. The session checks the suspension indicator before suspend the serializer. ASTERISK-26145 #close Change-Id: Iaaebee60013a58c942ba47b1b4930a63e686663b 2016-11-08 10:48 +0000 [87e1ebc91a] Mark Michelson * res_pjsip_session: Do not call session supplements when it's too late. res_pjsip_sesssion was hooking into transaction and invite state changes. One of the reasons for doing so was due to the PJSIP_EVENT_TX_MSG event. The idea was that we were hooking into the message sending process, and so we should call session supplements to alter the outgoing message. In reality, this event was meant to indicate that the message either a) had already been sent, or b) required a DNS lookup and would be sent when the DNS query completed. In case (a), this meant we were altering an already-sent request/response for no reason. In case (b), this potentially meant we could be trying to alter a request/response at the same time that the DNS resolution completed. In this case, it meant we might be stomping on memory being used by the thread actually sending the message. This caused potential crashes and memory corruption. This patch removes the calls to session supplements from the case where the PJSIP_EVENT_TX_MSG event occurs. In all of these cases, trying to alter the message at this point is too late, and it can cause nothing but harm to try to do it. Because there were no longer any calls to the handle_outgoing() function, it has been removed. Change-Id: Ibcc223fb1c3a237927f38754e0429e80ee301e92 (cherry picked from commit e043d1a55cf356066b3b39ebac8b4bbb612ce807) 2016-11-02 09:15 +0000 [c8df727502] Joshua Colp * app_dial: Fix incorrect device state when channel is picked up. Given the scenario where multiple channels are dialed using Dial() but the caller is picked up using PickupChan() all outgoing channels except the channel specified to PickupChan() would be marked as ringing until the call had been hung up. When using the PickupChan application the channel executing the application is swapped into place of another channel. As part of this process the channel is answered. The Dial application has explicit logic which checks if the channel is answered, cancels all other outgoing channels, and bridges. This logic is different than the normal logic that is executed when an outgoing channel is answered. This different logic failed to publish dial events stating that the other outgoing channels had been canceled. As a result references to the outgoing channels were held onto by the dial masquerade process until the call had been ended and the channels had gone away. This would result in the channels appearing in the "core show channels" list despite not being present anymore and would also result in incorrect device state. This change makes it so that this logic also publishes dial events stating that the other outgoing channels have been canceled. ASTERISK-26549 Change-Id: Iea7168e6e82f7d4609ec0366153804e4f55ea64f 2016-10-18 16:50 +0000 [4c50689d09] Joshua Colp * Revert "pjproject_bundled: Add patch to address SSL crash" This reverts commit 28cc8a9dff2fb9210726cfa6274ae683fbfa4a01. Change-Id: I777cf8173f7a88273090bed72bfe57fb0e72b84f 2016-10-17 11:39 +0000 [28cc8a9dff] gtjoseph * pjproject_bundled: Add patch to address SSL crash Addresses crashes when an attempt is made to operate on an SSL socket after the socket has been closed. ASTERISK-26477 #close Change-Id: I421305b357558b4f9e690210dc0f4831ef4b3002 (cherry picked from commit 546ec4b038ac3d750c5138d7fbb8e3ce93f482df) 2016-10-12 16:24 +0000 [7c2bd702fd] Richard Mudgett * Audit ast_json_pack() calls for needed UTF-8 checks. Added needed UTF-8 checks before constructing json objects in various files for strings obtained outside the system. In this case string values from a channel driver's peer and not from the user setting channel variables. * aoc.c: Fixed type mismatch in s_to_json() for time and granularity json object construction. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: Iac2d867fa598daba5c5dbc619b5464625a7f2096 2016-10-12 16:20 +0000 [3b1b2fc1d5] Richard Mudgett * json: Check party id name, number, subaddresses for UTF-8. * Updated unit test as ast_json_name_number() is now NULL tolerant. ASTERISK-26466 #close Reported by: Richard Mudgett Change-Id: I7d4e14194f8f81f24a1dc34d1b8602c0950265a6 2016-10-11 18:14 +0000 [a835adfa92] Richard Mudgett * json: Add UTF-8 check call. Since the json library does not make the check function public we recreate/copy the function in our interface module. ASTERISK-26466 Reported by: Richard Mudgett Change-Id: I36d3d750b6f5f1a110bc69ea92b435ecdeeb2a99 2016-07-19 15:22 +0000 [7baedd9ecd] Alexei Gradinari * res_pjsip_pubsub: fixed a bug when pjsip_tx_data_dec_ref is called twice. This patch removed call of pjsip_tx_data_dec_ref in send_notify if send_request failed. The pjsip_dlg_send_request deletes the message on error by itself. It seems this patch fixes next issues: ASTERISK-26199 ASTERISK-26166 ASTERISK-26174 Change-Id: I8b05917c93d993f95d604c042ace5f1a5500f59a 2016-10-05 14:53 +0000 [a8e37c3d06] gtjoseph * pjproject_bundled: Add MALLOC_DEBUG capability pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. NOTE: The certified/13.8 version of this patch also builds libresample which is needed by pjsua. Later versions do not need libresample. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0 2016-08-29 18:08 +0000 [adcdecd47f] Richard Mudgett * res_pjsip: Add ignore_uri_user_options option. This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62 2016-09-08 16:34 +0000 Asterisk Development Team * asterisk certified/13.8-cert3 Released. 2016-09-08 11:34 +0000 [6cec20200b] gtjoseph * Release summaries: Remove previous versions 2016-09-08 11:34 +0000 [9c0f0eef6e] gtjoseph * .version: Update for certified/13.8-cert3 2016-09-08 11:34 +0000 [3923b114b9] gtjoseph * .lastclean: Update for certified/13.8-cert3 2016-09-08 11:34 +0000 [83362b5590] gtjoseph * realtime: Add database scripts for certified/13.8-cert3 2016-08-23 06:35 +0000 [d947baa255] Corey Farrell (license 5909) * chan_sip: Don't allocate new RTP instances on top of old ones. In some scenarios dialog_initialize_rtp can be called multiple times on the same dialog. This can cause RTP instances to be leaked along with multiple file descriptors for each instance. This change makes it so the existing RTP instances are destroyed and not overwritten, stopping the memory leak. ASTERISK-26272 #close patches: ASTERISK-26272-13.patch submitted by Corey Farrell (license 5909) Change-Id: Id529de1184c68f2f4d254ab41a1f458dafdb5f73 2016-08-10 15:14 +0000 [df9aa402a5] Mark Michelson * ConfBridge: Make some announcements asynchronous. Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a 2016-08-24 14:42 +0000 [b1e827d259] gtjoseph * res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362 2016-08-10 15:14 +0000 [c218e038d7] Mark Michelson * ConfBridge: Rework announcer channel methodology NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0 2016-08-23 05:53 +0000 [b82f0b7722] Joshua Colp * Revert "ConfBridge: Rework announcer channel methodology" This reverts commit 4ca730127ccdc895e4d9e32cb0828c27bf74817b. Change-Id: I8886feb69ae2dbf521a8c0937792349b70db52b2 2016-08-10 15:14 +0000 [4ca730127c] Mark Michelson * ConfBridge: Rework announcer channel methodology One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5 2016-08-16 15:36 +0000 [f40c6874c6] gtjoseph * res_pjsip: Add contact_user to endpoint contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4 2016-08-09 12:07 +0000 [78a6da0885] Kevin Harwell * alembic/sqlalchemy: auto increment only allowed on a single column The extensions table defined two columns (id and priority) as primary key autoincrement columns. However only one is allowed when defining the primary key. This patch removes the autoincrement attribute from the priority column since it does not need to be as such and really should not have been on there in the first place. This patch also removes 'context', 'exten', and 'priority' from the primary key index and creates a new combined unique contraint index on them. ASTERISK-26183 #close Change-Id: Ib9c712c612a4d7ec1edb0dcb77f1bae0905a470b (cherry picked from commit f6ec94cca66addac71d566d6fa48188b407f26ba) 2016-08-15 13:27 +0000 Asterisk Development Team * asterisk certified/13.8-cert2 Released. 2016-08-15 08:27 +0000 [d7eec92332] Joshua Colp * Release summaries: Remove previous versions 2016-08-15 08:27 +0000 [847b0d007d] Joshua Colp * .version: Update for certified/13.8-cert2 2016-08-15 08:27 +0000 [bd8581cd52] Joshua Colp * .lastclean: Update for certified/13.8-cert2 2016-08-15 08:27 +0000 [628620c5ef] Joshua Colp * realtime: Add database scripts for certified/13.8-cert2 2016-08-09 16:19 +0000 [13b950f4be] Mark Michelson * res_rtp_asterisk: Cache local RTCP address. When an RTCP packet is sent or received, res_rtp_asterisk generates a Stasis event that contains the RTCP report as well as the local and remote addresses that the report pertains to. The addresses are determined using ast_find_ourip(). For the local address, this will typically result in a lookup of the hostname of the server, and then a DNS lookup of that hostname. If you do not have the host in /etc/hosts, then this results in a full DNS lookup, which can potentially block for some time. This is especially problematic when performing RTCP reads, since those are done on the same thread responsible for reading and writing media. This patch addresses the issue by performing a lookup of the local address when RTCP is allocated. We then use this cached local address for the Stasis events when necessary. ASTERISK-26280 #close Reported by Mark Michelson Change-Id: I3dd61882c2e57036f09f0c390cf38f7c87e9b556 2016-08-01 06:57 +0000 [acf021cdec] Joshua Colp * ChangeLog: Updated for certified/13.8-cert2-rc1 2016-08-01 06:57 +0000 [fba5cf4a00] Joshua Colp * Release summaries: Add summaries for certified/13.8-cert2-rc1 2016-08-01 06:57 +0000 [b2cc9b4879] Joshua Colp * Release summaries: Remove previous versions 2016-08-01 06:57 +0000 [20e25657fa] Joshua Colp * .version: Update for certified/13.8-cert2-rc1 2016-08-01 06:57 +0000 [08c26fba06] Joshua Colp * .lastclean: Update for certified/13.8-cert2-rc1 2016-08-01 06:57 +0000 [b539479f10] Joshua Colp * realtime: Add database scripts for certified/13.8-cert2-rc1 2016-06-21 10:53 +0000 [164bfc8574] Scott Griepentrog * PJSIP: provide transport type with received messages The receipt of a SIP MESSAGE may occur over any transport including TCP and TLS. When the message is received, the original URI is added to the message in the field PJSIP_RECVADDR, but this is insufficient to ensure a reply message can reach the originating endpoint. This patch adds the PJSIP_TRANSPORT field populated with the transport type. ASTERISK-26132 #close Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e (cherry picked from commit 69d58a1e377938e5236f51200e222eb219739441) 2016-07-21 22:28 +0000 [7809034c0d] Richard Mudgett * dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc 2016-07-21 09:05 +0000 [5bc48a290b] gtjoseph * chan_sip: Prevent deadlock when issuing "sip show channels" sip_show_channels locks the dialogs container first then locks each sip_pvt so it can spit out the details. The rest of sip dialog processing locks the sip_pvt first then locks the dialogs container if it needs to. Both lock in the order they need but deadlocks can result. To fix, sip_show_channels and sip_show_channelstats have been converted to use an iterator rather than ao2_callback. This way the container is locked only while getting the next entry and is unlocked when the callback is called. ASTERISK-23013 #close Change-Id: Id9980419909e811f89484950ed46ef117b9eb990 2016-07-12 17:24 +0000 [49defa5578] Richard Mudgett * res_fax: Fix FAXOPT(faxdetect) timeout option. The fax detection timeout option did not work because basically the wrong variable was checked in fax_detect_framehook(). As a result, the timer would timeout immediately and disable fax detection. * Fixed ignoring negative timeout values. We'd complain and then go right on using the negative value. * Fixed destroy_faxdetect() in the off-nominal case of an incomplete object creation. * Added more range checking to FAXOPT(gateway) timeout parameter. ASTERISK-26214 #close Reported by: Richard Mudgett Change-Id: Idc5e698dfe33572de9840bc68cd9fc043cbad976 2016-07-18 16:16 +0000 [a0485fe851] Richard Mudgett * chan_dahdi: Add faxdetect_timeout option. The new option allows the channel driver's faxdetect option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. * Don't clear dsp_features after passing them to the dsp code in my_pri_ss7_open_media(). We should still remember them especially for the new faxdetect_timeout option. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Ieffd3fe788788d56282844774365546dce8ac810 2016-07-15 20:44 +0000 [d172104e12] Richard Mudgett * res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d 2016-07-13 14:09 +0000 Asterisk Development Team * asterisk certified/13.8-cert1 Released. 2016-07-13 08:34 +0000 [482561f1e3] Joshua Colp * Release summaries: Remove previous versions 2016-07-13 08:34 +0000 [3cb116d75a] Joshua Colp * .version: Update for certified/13.8-cert1 2016-07-13 08:34 +0000 [797d39c81c] Joshua Colp * .lastclean: Update for certified/13.8-cert1 2016-07-13 08:34 +0000 [f5fbfe9a6a] Joshua Colp * realtime: Add database scripts for certified/13.8-cert1 2016-07-07 10:38 +0000 [22a36e5b10] Joshua Colp * chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c 2016-06-22 13:41 +0000 [d0c04c8986] gtjoseph * res_rtp_asterisk: Fix a self-comparison identified by gcc 6 gcc 6 caught a previously unidentified self-comparison in ice_candidate_cmp. Fixed it and re-ordered the predicates for better short-circuiting. ASTERISK-26140 #close Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7 2016-06-30 08:25 +0000 [0d694ce9b8] gtjoseph * configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK from getting set when using an external pjproject. ASTERISK-26099 #close Reported-by: Ross Beer Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae 2016-06-28 08:22 +0000 [5f444b1f5b] gtjoseph * BuildSystem: Fix a few issues hightlighted by gcc 6.x gcc 6.1.1 caught a few more issues. Made sure the unit tests still pass for the func_env and stdtime issues. ASTERISK-26157 #close Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e 2016-06-22 16:15 +0000 [f282a88ee4] Mark Michelson * ChangeLog: Updated for certified/13.8-cert1-rc3 2016-06-22 16:15 +0000 [bd6da93116] Mark Michelson * Release summaries: Add summaries for certified/13.8-cert1-rc3 2016-06-22 16:14 +0000 [4df81def29] Mark Michelson * Release summaries: Remove previous versions 2016-06-22 16:14 +0000 [286d58affc] Mark Michelson * .version: Update for certified/13.8-cert1-rc3 2016-06-22 16:14 +0000 [8b7fe94df7] Mark Michelson * .lastclean: Update for certified/13.8-cert1-rc3 2016-06-22 16:14 +0000 [0449fd2e1e] Mark Michelson * realtime: Add database scripts for certified/13.8-cert1-rc3 2016-06-09 09:20 +0000 [a6610fbe2f] gtjoseph * build: Fix ast_sockaddr initialization to be more portable A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4 2016-06-12 11:19 +0000 [102d88e791] gtjoseph * res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription Occasionally under load we'll attempt to send a final NOTIFY on a subscription that's already been terminated and a SEGV will occur down in pjproject's evsub_destroy function. This is a result of a race condition between all the paths that can generate a notify and/or destroy the underlying pjproject evsub object: * The client can send a SUBSCRIBE with Expires: 0. * The client can send a SUBSCRIBE/refresh. * The subscription timer can expire. * An extension state can change. * An MWI event can be generated. * The pjproject transaction timer (timer_b) can expire. Normally when our pubsub_on_evsub_state is called with a terminate, we push a task to the serializer and return at which point the dialog is unlocked. This is usually not a problem because the task runs immediately and locks the dialog again. When the system is heavily loaded though, there may be a delay between the unlock and relock during which another event may occur such as the subscription timer or timer_b expiring, an extension state change, etc. These may also cause a terminate to be processed and if so, we could cause pjproject to try to destroy the evsub structure twice. There's no way for us to tell that the evsub was already destroyed and the evsub's group lock can't tolerate this and SEGVs. The remedy is twofold. * A patch has been submitted to Teluu and added to the bundled pjproject which adds add/decrement operations on evsub's group lock. * In res_pjsip_pubsub: * configure.ac and pjproject-bundled's configure.m4 were updated to check for the new evsub group lock APIs. * We now add a reference to the evsub group lock when we create the subscription and remove the reference when we clean up the subscription. This prevents evsub from being destroyed before we're done with it. * A state has been added to the subscription tree structure so termination progress can be tracked through the asyncronous tasks. * The pubsub_on_evsub_state callback has been split so it's not doing double duty. It now only handles the final cleanup of the subscription tree. pubsub_on_rx_refresh now handles both client refreshes and client terminates. It was always being called for both anyway. * The serialized_on_server_timeout task was removed since serialized_pubsub_on_rx_refresh was almost identical. * Missing state checks and ao2_cleanups were added. * Some debug levels were adjusted to make seeing only off-nominal things at level 1 and nominal or progress things at level 2+. ASTERISK-26099 #close Reported-by: Ross Beer. Change-Id: I779d11802cf672a51392e62a74a1216596075ba1 2016-06-13 13:33 +0000 [d9ab222edc] Richard Mudgett * res_rtp_multicast.c: Fix warning message typo. Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3 2016-06-10 12:35 +0000 [39329a9e66] Richard Mudgett * chan_rtp: Backport changes from master. * Deprecate chan_multicast_rtp. Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e 2016-06-10 16:13 +0000 [6d45341963] Richard Mudgett * chan_rtp.c: Copy file from chan_multicast_rtp.c Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef 2016-06-03 22:44 +0000 [0322479ff7] Richard Mudgett * res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. This patch fixes a race condition processing received REGISTER requests and their retransmissions caused by REGISTER requests being processed by two threads. The "sip_transaction Unable to register REGISTER transaction (key exists)" message is a notable symptom of this issue. This issue was more likely to happen before the pjsip/distributor serializers were created. Instead of steps one and two below placing the REGISTER messages into the same pjsip/distributor they were placed in random pjsip/default serializers. 1) REGISTER requests come in and get placed on the pjsip/distributor serializer. 2) Before the first request is processed a retransmission comes in and is placed on the same pjsip/distributor serializer. 3) The first request goes up the pjsip stack and is then shunted off to the pjsip/aor/ serializer. 4) Before the first request is completed processing in the pjsip/aor/ serializer, the second request goes up the pjsip stack and is also shunted off to the pjsip/aor/ serializer. 5) The first request completes processing and sends out its response. 6) The second request completes processing and tries to send out its response but pjlib complains that the REGISTER transaction key already exists. 7) Sadness ensues. * The race is eliminated by removing the pjsip/aor/ serializer and continuing the processing in the pjsip/distributor serializer. Now any retransmissions queued in the pjsip/distributor serializer will be processed after the first message is completely processed. ASTERISK-26088 #close Reported by: Richard Mudgett Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a 2016-06-03 11:35 +0000 [942fa0c95b] Richard Mudgett * stasis: Add setting subscription congestion levels. Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244 2016-06-02 18:19 +0000 [b046fa1907] Richard Mudgett * sorcery: Add setting object type congestion levels. Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6 2016-06-02 16:08 +0000 [237f9ef7af] Richard Mudgett * taskprocessors: Implement high/low water mark alerts. When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa 2016-05-27 17:31 +0000 [ff70f04a37] Richard Mudgett * res_pjsip_session: Use distributor serializer for incoming calls. We must continue using the serializer that the original INVITE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. Outgoing call legs create the pjsip/outsess/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc 2016-05-27 16:28 +0000 [4b26c9ead8] Richard Mudgett * res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be 2016-05-27 12:50 +0000 [a137d1822e] Richard Mudgett * res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. We must continue using the serializer that the original SUBSCRIBE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/ serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0 2016-05-26 17:35 +0000 [9a7a5aec18] Richard Mudgett * pjsip_distributor.c: Consistently pick a serializer for messages. Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407 2016-06-02 12:51 +0000 [f2a76c4292] Richard Mudgett * pjsip_distributor.c: Ignore messages until fully booted. We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264 2016-04-01 13:30 +0000 [51e45e5ca5] gtjoseph * res_pjsip contact: Lock expiration/addition of contacts Contact expiration can occur in several places: res_pjsip_registrar, res_pjsip_registrar_expire, and automatically when anyone calls ast_sip_location_retrieve_aor_contact. At the same time, res_pjsip_registrar may also be attempting to renew or add a contact. Since none of this was locked it was possible for one thread to be renewing a contact and another thread to expire it immediately because it was working off of stale data. This was the casue of intermittent registration/inbound/nominal/multiple_contacts test failures. Now, the new named lock functionality is used to lock the aor during contact expire and add operations and res_pjsip_registrar_expire now checks the expiration with the lock held before deleting the contact. ASTERISK-25885 #close Reported-by: Josh Colp Change-Id: I83d413c46a47796f3ab052ca3b349f21cca47059 2016-03-31 20:04 +0000 [880d502141] gtjoseph * lock: Add named lock capability Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 2016-06-02 12:04 +0000 [a81feefde9] Joshua Colp * res_odbc: Implement a connection pool. Testing has shown that our usage of UnixODBC is problematic due to bugs within UnixODBC itself as well as the heavy weight cost of connecting and disconnecting database connections, even when pooling is enabled. For users of UnixODBC 2.3.1 and earlier crashes would occur due to insufficient protection of the disconnect operation. This was fixed in UnixODBC 2.3.2 and above. For users of UnixODBC 2.3.3 and higher a slow-down would occur under heavy database use due to repeated connection establishment. A regression is present where on each connection the database configuration is cached again, with the cache growing out of control. The connection pool implementation present in this change helps to mitigate these issues by reducing how much we connect and disconnect database connections. We also solve the issue of crashes under UnixODBC 2.3.1 by defaulting the maximum number of connections to 1, returning us to the previous working behavior. For users who may have a fixed version the maximum concurrent connection limit can be increased helping with performance. The connection pool works by keeping a list of active connections. If the connection limit has not been reached a new connection is established. If the connection limit has been reached then the request waits until a connection becomes available before continuing. ASTERISK-26074 #close ASTERISK-26054 #close Change-Id: I6774bf4bac49a0b30242c76a09c403d2e856ecff 2016-05-30 10:58 +0000 [aab8bc5d31] gtjoseph * pjproject_bundled: Move to pjproject 2.5 Although all the patches we had against 2.4.5 were applied by Teluu, a new bug was introduced preventing re-use of tcp and tls transports This patch removes all the previous patches against 2.4.5, updates the version to 2.5, and adds a new patch to correct the transport re-use problem. Change-Id: I0dc6c438c3910f7887418a5832ca186aea23d068 (cherry picked from commit e8abfdcdc5ce4d32d1fe281e75b13fd652f9e5f7) 2016-05-18 07:54 +0000 [b9a28ccbd4] gtjoseph * udptl: Don't eat sequence numbers until OK is received Scenario: Local fax -> Asterisk w/ firewall -> Provider -> Remote fax * Local fax starts rtp call to remote fax * Remote fax starts t38 call back to local fax. * Local fax sends t38 no-signal to Asterisk before sending an OK. * udptl processes the frame and increments the expected sequence number. * chan_sip drops the frame because the call isn't up so nothing goes out the external interface to open the port for incoming packets. * Local fax sends OK and Asterisk sends OK to the remote fax. * Remote fax sends t38 packets which are dropped by the firewall. * Local fax re-sends t38 no-signal with the same sequence number. * udptl drops the frame because it thinks it's a dup. * Still no outgoing packets to open the firewall. * t38 negotiation fails. The patch drops frames t38 received before udptl sequence processing when the call hasn't been answered yet. The second no-signal frame is then seen as new and is relayed out the external interface which opens the port and allows negotiation to continue. ASTERISK-26034 #close Change-Id: I11744b39748bd2ecbbe8ea84cdb4f3c5943c5af9 2016-05-17 11:14 +0000 [f85c77a9e1] gtjoseph * chan_sip: Prevent extra Session-Expires headers from being added When chan_sip does a re-INVITE to refresh a session and authentication is required, the INVITE with the Authorization header containes a second Session-Expires header without the ";refersher=" parameter. This is causing some proxies to return a 400. Also, when Asterisk is the uas and the refresher, it is including the Session-Expires and Min-SE headers in OPTIONS messages which is not allowed per RFC4028. This patch (based on the reporter's) Checks to see if a Session-Expires header is already in the message before adding another one. It also checks that the method is INVITE or UPDATE. ASTERISK-26030 #close Change-Id: I58a7b07bab5a3177748d8a7034fb8ad8e11ce1d9 2016-05-07 14:39 +0000 [8bf050b853] gtjoseph * config_transport: Tell pjproject to allow all SSL/TLS protocols The default tls settings for pjproject only allow TLS 1, TLS 1.1 and TLS 1.2. SSL is not allowed. So, even if you specify "sslv3" for a transport method, it's silently ignored and one of the TLS protocols is used. This was a new behavior of pjsip_tls_setting_default() in 2.4 (when tls.proto was added) that we never caught. Now we need to set tls.proto = 0 after we call pjsip_tls_setting_default(). This tells pjproject to set the socket protocol to match the method. ASTERISK-26004 #close Change-Id: Icfb55c1ebe921298dedb4b1a1d3bdc3ca41dd078 2016-05-05 11:37 +0000 [4fc2c98369] Kevin Harwell * res_pjsip_authenticator_digest: Don't use source port in nonce verification From the issue reporter: "res_pjsip_outbound_authenticator_digest builds a nonce that is a hash of the timestamp, the source address, the source port, a server UUID that is calculated at startup, and the authentication realm. Rather than caching nonces that we create, we instead attempt to re-calculate the nonce when receiving an incoming request with authentication. We then compare the re-calculated nonce to the incoming nonce, and if they don't match, then authentication has failed early. The problem is that it is possible, especially when using TCP, to receive two requests from the same endpoint but have differing source ports for those requests. Asterisk itself commonly will use different source ports for outbound TCP requests." This patch removes the source port dependency when building the nonce. ASTERISK-25978 #close Change-Id: I871b5f4adce102df1c4988066283095ec509dffe 2016-05-05 05:07 +0000 [4e7791d483] Joshua Colp * file: Ensure nativeformats remains valid for lifetime of use. It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 2016-05-03 07:55 +0000 [601602f44b] Joshua Colp * ChangeLog: Updated for certified/13.8-cert1-rc2 2016-05-03 07:55 +0000 [13461bb9a6] Joshua Colp * Release summaries: Add summaries for certified/13.8-cert1-rc2 2016-05-03 07:54 +0000 [cadb5c4e64] Joshua Colp * Release summaries: Remove previous versions 2016-05-03 07:54 +0000 [d4d5548ef8] Joshua Colp * .version: Update for certified/13.8-cert1-rc2 2016-05-03 07:54 +0000 [a5bc40ae51] Joshua Colp * .lastclean: Update for certified/13.8-cert1-rc2 2016-05-03 07:54 +0000 [2b6df52c66] Joshua Colp * realtime: Add database scripts for certified/13.8-cert1-rc2 2016-04-15 11:59 +0000 [c4426f1035] Alexei Gradinari * res_pjsip: disable multi domain to improve realtime performace This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7 2016-04-26 05:48 +0000 [c69e0f1813] Joshua Colp * app_queue: Fix crash when unloading module. When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b 2016-04-21 14:23 +0000 [eebe8b3dd3] Kevin Harwell * app_queue: queue members can receive multiple calls It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48 2016-04-22 17:53 +0000 [5cbd4b9799] gtjoseph * res_agi: Prevent run_agi from eating frames it shouldn't The run_agi function is eating control frames when it shouldn't be. This is causing issues when an AGI is run from CONNECTED_LINE_SEND_SUB in a blond transfer. Alice calls Bob. Bob attended transfers to Charlie but hangs up before Charlie answers. Alice gets the COLP UPDATE indicating Charlie but Charlie never gets an UPDATE and is left thinking he's connected to Bob. In this case, when CONNECTED_LINE_SEND_SUB runs on Alice's channel and it calls an AGI, the extra eaten frames prevent CONNECTED_LINE_SEND_SUB from running on Charlie's channel. The fix was to accumulate deferrable frames in the "forever" loop instead of dropping them, and re-queue them just before running the actual agi command or exiting. ASTERISK-25951 #close Change-Id: I0f4bbfd72fc1126c2aaba41da3233a33d0433645 2016-04-15 14:36 +0000 [bc51227ef8] Richard Mudgett * res_stasis: Handle re-enter stasis bridge with swap channel. We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee 2016-04-19 16:58 +0000 [8dd79720e6] Richard Mudgett * bridge: Hold off more than one imparting channel at a time. An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 2016-04-19 17:52 +0000 [2a2e754d15] gtjoseph * res_pjsip_callerid: Clear out display name if id->name is not valid When create_new_id_hdr creates a new RPID or PAI header, it starts by cloning the From header, then it overwrites the display name and uri from the channel's connected.id. If the connected.id.name wasn't valid, create_new_id_hdr was leaving the display name from the From header in the new RPID or PAI header. On an attended transfer where the originator had a caller id number set but not a display name, the re-INVITE to the final transferee had the number of the originator but the display name of the transferer. Added a check to clear out the display name in the new header if connected.id.name was invalid. ASTERISK-25942 #close Change-Id: I60b4bf7a7ece9b7425eba74151c0b4969cd2738b 2016-04-19 13:02 +0000 [188ce34aff] Joshua Colp * app_talkdetect: Make the module core supported. This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88 2016-04-19 13:00 +0000 [da80f40014] Joshua Colp * app_talkdetect: Enable for testsuite tests. Change-Id: I9acf2e2210f7a15cdd2c63c4c8dcb92de6b47d43 2016-04-18 12:12 +0000 [9f3ecf0a8d] Mark Michelson * PJSIP: Remove PJSIP parsing functions from uri length validation. The PJSIP parsing functions provide a nice concise way to check the length of a hostname in a SIP URI. The problem is that in order to use those parsing functions, it's required to use them from a thread that has registered with PJLib. On startup, when parsing AOR configuration, the permanent URI handler may not be run from a PJLib-registered thread. Specifically, this could happen when Asterisk was started in daemon mode rather than console-mode. If PJProject were compiled with assertions enabled, then this would cause Asterisk to crash on startup. The solution presented here is to do our own parsing of the contact URI in order to ensure that the hostname in the URI is not too long. The parsing does not attempt to perform a full SIP URI parse/validation, since the hostname in the URI is what is important. ASTERISK-25928 #close Reported by Joshua Colp Change-Id: Ic3d6c20ff3502507c17244a8b7e2ca761dc7fb60 2016-04-18 17:00 +0000 [39b4742db1] Mark Michelson * res_pjsip_registrar: Fix bad memory-ness with user_agent. Recent changes to the PJSIP registrar resulted in tests failing due to missing AOR_CONTACT_ADDED test events. The reason for this was that the user_agent string had junk values in it, resulting in being unable to generate the event. I'm going to be honest here, I have no idea why this was happening. Here are the steps needed for the user_agent variable to get messed up: * REGISTER is received * First contact in the REGISTER results in a contact being removed * Second contact in the REGISTER results in a contact being added * The contact, AOR, expiration, and user agent all have to be passed as format parameters to the creation of a string. Any subset of those parameters would not be enough to cause the problem. Looking into what was happening, the thing that struck me as odd was that the user_agent variable was meant to be set to the value of the User-Agent SIP header in the incoming REGISTER. However, when removing a contact, the user_agent variable would be set (via ast_strdupa inside a loop) to the stored contact's user_agent. This means that the user_agent's value would be incorrect when attempting to process further contacts in the incoming REGISTER. The fix here is to use a different variable for the stored user agent when removing a contact. Correcting the behavior to be correct also means the memory usage is less weird, and the issue no longer occurs. ASTERISK-25929 #close Reported by Joshua Colp Change-Id: I7cd24c86a38dec69ebcc94150614bc25f46b8c08 2016-04-18 13:41 +0000 [4caa57f6b3] Joshua Colp * res_pjsip_transport_management: Allow unload to occur. At shutdown it is possible for modules to be unloaded that wouldn't normally be unloaded. This allows the environment to be cleaned up. The res_pjsip_transport_management module did not have the unload logic in it to clean itself up causing the res_pjsip module to not get unloaded. As a result the res_pjsip monitor thread kept going processing traffic and timers when it shouldn't. Change-Id: Ic8cadee131e3b2c436a81d3ae8bb5775999ae00a 2016-04-14 13:49 +0000 [0b35582bbb] Mark Michelson * transport management: Register thread with PJProject. The scheduler thread that kills idle TCP connections was not registering with PJProject properly and causing assertions if PJProject was built in debug mode. This change registers the thread with PJProject the first time that the scheduler callback executes. AST-2016-005 Change-Id: I5f7a37e2c80726a99afe9dc2a4a69bdedf661283 2016-03-08 12:12 +0000 [9f8b803a29] Mark Michelson * res_pjsip_transport_management: Kill idle TCP connections. "Idle" here means that someone connects to us and does not send a SIP request. PJProject will not automatically time out such connections, so it's up to Asterisk to do it instead. When we receive an incoming TCP connection, we will start a timer (equivalent to transaction timer D) waiting to receive an incoming request. If we do not receive a request in that timeframe, then we will shut down the TCP connection. ASTERISK-25796 #close Reported by George Joseph AST-2016-005 Change-Id: I7b0d303e5d140d0ccaf2f7af562071e3d1130ac6 2016-03-08 10:52 +0000 [a35d3eb73b] Mark Michelson * Rename res_pjsip_keepalive res_pjsip_transport_management ASTERISK-25796 Reported by George Joseph AST-2016-005 Change-Id: Id322a05f927392293570599730050bc677d99433 2016-04-14 07:15 +0000 [3de37dee68] Mark Michelson * AST-2016-004: Fix crash on REGISTER with long URI. Due to some ignored return values, Asterisk could crash if processing an incoming REGISTER whose contact URI was above a certain length. ASTERISK-25707 #close Reported by George Joseph Patches: 0001-res_pjsip-Validate-that-URIs-don-t-exceed-pjproject-.patch AST-2016-004 Change-Id: I0ed3898fe7ab10121b76c8c79046692de3a1be55 2016-03-23 08:59 +0000 [e378c18815] gtjoseph * pjproject-bundled: Cleanups for reported issues PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd (cherry picked from commit d963a3374991c64594cf196e90a5c74964c8ba7c) 2016-04-06 11:02 +0000 [dd93204a84] Joshua Colp * ChangeLog: Updated for certified/13.8-cert1-rc1 2016-04-06 11:01 +0000 [6d29a919d4] Joshua Colp * Release summaries: Add summaries for certified/13.8-cert1-rc1 2016-04-06 10:27 +0000 [4fa3428247] Joshua Colp * Release summaries: Remove previous versions 2016-04-06 10:27 +0000 [b418e14998] Joshua Colp * .version: Update for certified/13.8-cert1-rc1 2016-04-06 10:27 +0000 [69b6cf2368] Joshua Colp * .lastclean: Update for certified/13.8-cert1-rc1 2016-04-06 10:27 +0000 [847dc5c7d7] Joshua Colp * realtime: Add database scripts for certified/13.8-cert1-rc1 2016-04-06 09:20 +0000 [c23bf7c8df] Joshua Colp * ChangeLog: Updated for certified/13.8-cert1-rc1 2016-04-06 09:19 +0000 [4f94668022] Joshua Colp * Release summaries: Add summaries for certified/13.8-cert1-rc1 2016-04-06 08:47 +0000 [454daec0e1] Joshua Colp * Release summaries: Remove previous versions 2016-04-06 08:47 +0000 [4ba2b5e92c] Joshua Colp * .version: Update for certified/13.8-cert1-rc1 2016-04-06 08:47 +0000 [e6f27ca09c] Joshua Colp * .lastclean: Update for certified/13.8-cert1-rc1 2016-04-06 08:47 +0000 [08dbdd5996] Joshua Colp * realtime: Add database scripts for certified/13.8-cert1-rc1 2016-04-06 08:26 +0000 [ec7a89771d] Joshua Colp * ChangeLog: Updated for certified/13.8-cert1-rc1 2016-04-06 08:25 +0000 [ffcb651205] Joshua Colp * Release summaries: Add summaries for certified/13.8-cert1-rc1 2016-04-06 07:52 +0000 [97499f717a] Joshua Colp * Release summaries: Remove previous versions 2016-04-06 07:52 +0000 [99d52771b5] Joshua Colp * .version: Update for certified/13.8-cert1-rc1 2016-04-06 07:52 +0000 [eb9e193c65] Joshua Colp * .lastclean: Update for certified/13.8-cert1-rc1 2016-04-06 07:52 +0000 [8ec588b8b1] Joshua Colp * realtime: Add database scripts for certified/13.8-cert1-rc1 2016-04-05 14:23 +0000 [4b87a773dc] Mark Michelson * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating sendrecv on streams. * Device sends ACK with SDP indicating sendonly on streams. At this point, PJMedia's SDP negotiator saves Asterisk's local state as being recvonly. Now, when the device attempts to unhold, it again uses a deferred SDP reinvite, so we end up doing the following: * Device sends reinvite with no SDP * Asterisk sends 200 OK with SDP indicating recvonly on streams * Device sends ACK with SDP indicating sendonly on streams The problem here is that Asterisk offered recvonly, and by RFC 3264's rules, if an offer is recvonly, the answer has to be sendonly. The result is that the device is not taken off hold. What is supposed to happen is that Asterisk should indicate sendrecv in the 200 OK that it sends. This way, the device has the freedom to indicate sendrecv if it wants the stream taken off hold, or it can continue to respond with sendonly if the purpose of the reinvite was something else (like a session timer refresher). The fix here is to alter the SDP negotiator's state when we receive a reinvite with no SDP. If the negotiator's state is currently in the recvonly or inactive state, then we alter our local state to be sendrecv. This way, we allow the device to indicate the stream state as desired. ASTERISK-25854 #close Reported by Robert McGilvray Change-Id: I7615737276165eef3a593038413d936247dcc6ed 2016-04-05 09:06 +0000 [c29e2e3fb7] Joshua Colp * .version: Update for certified/13.8 Change-Id: I37e5a8e36c2f4f9137f8f230c99220005424e514 2015-01-06 21:29 +0000 [3c796e694e] Matt Jordan * Disable extended support modules Change-Id: Ia2e359021b3eccecce20028098c5b6d1099c3f9e 2016-03-28 18:10 +0000 [7b6c4decd3] Richard Mudgett * res_stasis: Fix crash on a hanging up channel. * Give the struct stasis_app_control ao2 object a ref to the channel held in the object. Now the channel will still be around if a thread needs to post a stasis message instead of crash because the topic was destroyed. * Moved stopping any lingering silence generator out of the struct stasis_app_control destructor and made it a part of exiting the Stasis application. Who knows which thread the destructor will be called under so it cannot affect the channel's silence generator. Not only was the channel unprotected when the silence generator was stopped, stasis may no longer even control the channel. ASTERISK-25882 Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4 2016-03-29 14:39 +0000 [fad0410486] Mark Michelson * ChangeLog: Updated for 13.8.0 2016-03-29 14:39 +0000 [0f885f0076] Mark Michelson * Release summaries: Add summaries for 13.8.0 2016-03-29 14:34 +0000 [a1fa37aebd] Mark Michelson * Release summaries: Remove previous versions 2016-03-29 14:34 +0000 [e7de5fd439] Mark Michelson * .version: Update for 13.8.0 2016-03-29 14:34 +0000 [8baf813848] Mark Michelson * .lastclean: Update for 13.8.0 2016-03-29 14:34 +0000 [42469df205] Mark Michelson * realtime: Add database scripts for 13.8.0 2016-03-22 13:32 +0000 [06f5ace1fa] Mark Michelson * ChangeLog: Updated for 13.8.0-rc1 2016-03-22 13:26 +0000 [a698424678] Mark Michelson * Release summaries: Add summaries for 13.8.0-rc1 2016-03-22 13:21 +0000 [e395a0b973] Mark Michelson * .version: Update for 13.8.0-rc1 2016-03-22 13:21 +0000 [38a86b2dbf] Mark Michelson * .lastclean: Update for 13.8.0-rc1 2016-03-22 13:21 +0000 [e0c8c8bf4a] Mark Michelson * realtime: Add database scripts for 13.8.0-rc1 2016-03-18 14:31 +0000 [6a40520fe9] Kevin Harwell * chan_pjsip: ref leak when checking direct_media_glare Fix the reference leak introduced in the following commit: 9444ddadf8525d1ce66a1faf1db97f9f6c265ca4 ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85 2016-03-16 12:37 +0000 [9444ddadf8] Kevin Harwell * chan_pjsip: transfers with direct media reinvite has wrong address/port During a transfer involving direct media a race occurs between when the transferer channel is swapped out, initiating rtp changes/updates, and the subsequent reinvites. When Alice, after speaking with Charlie (Bob is on hold), connects Bob and Charlie invites are sent to each in order to establish the call between them. Bob is taken off hold and Charlie is told to have his media flow through Asterisk. However, if before those invites go out the bridge updates Bob's and/or Charlie's rtp information with direct media data (i.e. address, port) then the invite(s) will contain the remote data in the SDP instead of the Asterisk data. The race occurs in the native bridge glue code when updating the peer. The direct_media_address can get set twice before sending out the first invite during call connection. This can happen because the checking/setting of the direct_media_address happened in one thread while the sending of the invite(s) happened in another thread. This fix removes the race condition by moving the checking/setting of the direct_media_address to be in the same thread as the sending of the invites(s). This serializes the checking/setting and sending so they can no longer happen out of order. ASTERISK-25849 #close Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873 2015-10-19 07:11 +0000 [88240f98d9] Rodrigo Ramírez Norambuena * install_prereq: Update repositories before install on Debian systems When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa (cherry picked from commit 88f3dbaec9509bfba8bc1de7799aa0dc65304bb5) 2015-06-03 20:12 +0000 [efcf9a96db] Rodrigo Ramírez Norambuena * install_prereq: Check if is installed aptitude otherwise to install. If in Debian or system based, dont have aptitude installed the script do nothing. This patch checked if aptitude installed, if not installed. Also, if execute script with all packages installed yet, the script not show nothing and return exit 1 because the command 'grep' get nothing from pipe from 'awk'. ASTERISK-25113 #close Reported By: Rodrigo Ramírez Norambuena Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f (cherry picked from commit 6737ded0581a9e1256bdfe30c1d747e7ca93f8b3) 2016-03-03 04:43 +0000 [2b1b8e382a] Sergio Medina Toledo * res_pjsip_refer.c: Fix seg fault in process of Refer-to header. The "Refer-to" header of an incoming REFER request is parsed by pjsip_parse_uri(). That function requires the URI parameter to be NULL terminated. Unfortunately, the previous code added the NULL terminator by overwriting memory that may not be safe. The overwritten memory results could be benign, memory corruption, or a segmentation fault. Now the URI is NULL terminated safely by copying the URI to a new chunk of memory with the correct size to be NULL terminated. ASTERISK-25814 #close Change-Id: I32565496684a5a49c3278fce06474b8c94b37342 2016-03-11 12:22 +0000 [de04308ae4] Richard Mudgett * chan_sip.c: Fix mwi resub deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 #close Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6 2016-03-10 17:01 +0000 [5f6627a8a4] Richard Mudgett * chan_sip.c: Fix registration timeout and expire deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508 2016-03-10 12:17 +0000 [32bd7a64f9] Richard Mudgett * chan_sip.c: Fix t38id deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f 2016-03-09 16:34 +0000 [43556b800b] Richard Mudgett * chan_sip.c: Fix reinviteid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1 2016-03-09 16:32 +0000 [38c1cdab2c] Richard Mudgett * chan_sip.c: Fix packet retransid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix retrans_pkt() to call check_pendings() with both the owner channel and the private objects locked as required. * Refactor dialog retransmission packet list to safely remove packet nodes. The list nodes are now ao2 objects. The list has a ref and the scheduled entry has a ref. ASTERISK-25023 Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641 2016-03-09 16:26 +0000 [e4ad55c888] Richard Mudgett * chan_sip.c: Fix waitid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Made always run check_pendings() under the scheduler thread so scheduler ids can be checked safely. ASTERISK-25023 Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52 2016-03-08 15:08 +0000 [98d5669c28] Richard Mudgett * chan_sip.c: Fix session timers deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900 2016-03-07 13:21 +0000 [9cb8f73226] Richard Mudgett * chan_sip.c: Fix autokillid deadlock potential. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f 2016-03-07 18:28 +0000 [c5c7f48a15] Richard Mudgett * chan_sip.c: Fix provisional_keepalive_sched_id deadlock. This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48 2016-03-09 11:22 +0000 [f959d84dfd] Richard Mudgett * chan_sip.c: Adjust how dialog_unlink_all() stops scheduled events. This patch is part of a series to resolve deadlocks in chan_sip.c. * Make dialog_unlink_all() unschedule all items at once in the sched thread. ASTERISK-25023 Change-Id: I7743072fb228836e8228b72f6dc46c8cc50b3fb4 2016-03-10 21:54 +0000 [5f3225ddcc] Richard Mudgett * chan_sip.c: Clear scheduled immediate events on unload. This patch is part of a series to resolve deadlocks in chan_sip.c. The reordering of chan_sip's shutdown is to handle any immediate events that get put onto the scheduler so resources aren't leaked. The typical immediate events at this time are going to be concerned with stopping other scheduled events. ASTERISK-25023 Change-Id: I3f6540717634f6f2e84d8531a054976f2bbb9d20 2016-03-15 14:51 +0000 [7a74971771] Richard Mudgett * sip/dialplan_functions.c: Fix /channels/chan_sip/test_sip_rtpqos crash. This patch is part of a series to resolve deadlocks in chan_sip.c. Delaying destruction of the chan_sip sip_pvt structures caused the /channels/chan_sip/test_sip_rtpqos unit test to crash. That test registers a special test ast_rtp_engine with the rtp engine module. When the unit test completes it cleans up by unregistering the test ast_rtp_engine and exits. Since the delayed destruction of the sip_pvt happens after the unit test returns, the destructor tries to call the rtp engine destroy callback of the test ast_rtp_engine auto variable which no longer exists on the stack. * Change the test ast_rtp_engine auto variable to a static variable. Now the variable can still exist after the unit test exits so the delayed sip_pvt destruction can complete successfully. ASTERISK-25023 Change-Id: I61e34a12d425189ef7e96fc69ae14993f82f3f13 2016-03-15 13:31 +0000 [d2c09ed73b] Andrew Nagy * app_stasis: Don't hang up if app is not registered This prevents pbx_core from hanging up the channel if the app isn't registered. ASTERISK-25846 #close Change-Id: I63216a61f30706d5362bc0906b50b6f0544aebce 2016-03-07 15:50 +0000 [b2d2906445] Richard Mudgett * sched.c: Ensure oldest expiring entry runs first. This patch is part of a series to resolve deadlocks in chan_sip.c. * Updated sched unit test to check new behavior. ASTERISK-25023 Change-Id: Ib69437327b3cda5e14c4238d9ff91b2531b34ef3 2016-03-04 18:25 +0000 [9ae21b510f] Richard Mudgett * chan_sip.c: Made sip_reinvite_retry() call sip_pvt_lock_full(). Change-Id: I90f04208a089f95488a2460185a8dbc3f6acca12 2016-03-07 18:56 +0000 [56bcb97a3c] Richard Mudgett * chan_sip.c: Simplify sip_pvt destructor call levels. Remove destructor calling destroy_it calling really_destroy_it for no benefit. Just make the destructor the really_destroy_it function. Change-Id: Idea0d47b27dd74f2488db75bcc7f353d8fdc614a 2016-03-14 08:59 +0000 [677a65fcbb] Joshua Colp * build: Add configure check for proto field of PJSIP TLS transport setting. Older versions of PJSIP do not have the proto field on the TLS transport setting structure. This change adds a configure check so even if it is not present we will still be able to build. Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9 2016-03-12 16:02 +0000 [32f0a3d52a] gtjoseph * build_system: Split COMPILE_DOUBLE from DONT_OPTIMIZE I can't ever recall actually needing the intermediate files or the checking that a double compile produces. What I CAN remember is every DONT_OPTIMIZE build needing 3 invocations of gcc instead of 1 just to do the checks and produce those intermediate files. Having said that, Richard pointed out that the reason for the double compile was that there were cases in the past where a submitted patch failed to compile because the submitter never tried it with the optimizations turned on. To get the best of both worlds, COMPILE_DOUBLE has been split into its own option. If DONT_OPTIMIZE is turned on, COMPILE_DOUBLE will also be selected BUT you can then turn it off if all you need are the debugging symbols. This way you have to make an informed decision about disabling COMPILE_DOUBLE. To allow COMPILE_DOUBLE to be both auto-selected and turned off, a new feature was added to menuselect. The element can now contain an "autoselect" attribute which will turn the used member on but not create a hard dependency. The cflags.xml implementation for COMPILE_DOUBLE looks like this... COMPILE_DOUBLE core * app_chanspy: Fix occasional deadlock with ChanSpy and Local channels. Channel masquerading had a conflict with autochannel locking. When locking autochannel->channel, the channel is fetched from the autochannel and then locked. During the fetch, the autochannel -- which has no locks itself -- can be modified by someone who owns the channel lock. That means that the value of autochan->channel cannot be trusted until you hold the lock. In practice, this caused problems with Local channels getting masqueraded away while the ChanSpy attempted to get info from that channel. The old channel which was about to get removed got locked, but the new (replaced) channel got unlocked (no-op). Because the replaced channel was now locked (and would never get unlocked), it couldn't get removed from the channel list in a timely manner, and would now cause deadlocks when iterating over the channel list. This change checks the autochannel after locking the channel for changes to the autochannel. If the channel had been changed, the lock is reobtained on the new channel. In theory it seems possible that after this fix, the lock attempt on the old (wrong) channel can be on an already destroyed lock, maybe causing a crash. But that hasn't been observed in the wild and is harder induce than the current deadlock. Thanks go to Filip Frank for suggesting a fix similar to this and especially to IRC user hexanol for pointing out why this deadlock was possible and testing this fix. And to Richard for catching my rookie while loop mistake ;) ASTERISK-25321 #close Change-Id: I293ae0014e531cd0e675c3f02d1d118a98683def 2016-03-07 21:34 +0000 [875d5e9872] gtjoseph * pjproject_bundled: Remove --with-external-pa from configure options. Not sure why it was there in the first place as we already specify --disable-sound. Change-Id: Ia80a40e8b1e1acc287955ab11ba1fbd0c7d4cff9 2016-03-06 14:38 +0000 [530cff5f5f] gtjoseph * res_pjsip: Strip spaces from items parsed from comma-separated lists Configurations like "aors = a, b, c" were either ignoring everything after "a" or trying to look up " b". Same for mailboxes, ciphers, contacts and a few others. To fix, all the strsep(©, ",") calls have been wrapped in ast_strip. To facilitate this, ast_strip, ast_skip_blanks and ast_skip_nonblanks were updated to handle null pointers. In some cases, an ast_strlen_zero() test was added to skip consecutive commas. There was also an attempt to ast_free an ast_strdupa'd string in ast_sip_for_each_aor which was causing a SEGV. I removed it. Although this issue was reported for realtime, the issue was in the res_pjsip modules so all config mechanisms were affected. ASTERISK-25829 #close Reported-by: Mateusz Kowalski Change-Id: I0b22a2cf22a7c1c50d4ecacbfa540155bec0e7a2 2016-03-04 20:37 +0000 [3c8076a83b] gtjoseph * install_prereq: Add packages for bundled pjproject RedHat/CentOS needs python-devel Debian/Ubuntu needs automake, libsrtp-dev and python-dev Ubuntu also needed libncurses5-dev for cmenuselect so while not needed for pjproject, I adedd it anyway. Change-Id: Idf5fa16e2d87c687439621507e122cb9461d7089 2016-02-24 17:25 +0000 [27f32cd0a6] gtjoseph * res_pjsip_caller_id: Anonymize 'From' when caller id presentation is prohibited Per RFC3325, the 'From' header is now anonymized on outgoing calls when caller id presentation is prohibited. TID = trust_id_outbound PRO = Set(CALLERID(pres)=prohib) USR = endpoint/from_user DOM = endpoint/from_domain PAI = YES(privacy=off), NO(not sent), PRI(privacy=full) (assumes send_pai=yes) Conditions |Result --------------------|---------------------------------------------------- TID PRO USR DOM |PAI FROM --------------------|---------------------------------------------------- Y Y abc def.ghi |PRI "Anonymous" Y Y abc |PRI "Anonymous" Y Y def.ghi |PRI "Anonymous" Y Y |PRI "Anonymous" Y N abc def.ghi |YES Y N abc |YES > Y N def.ghi |YES "Caller Name" @def.ghi> Y N |YES "Caller Name" @> N Y abc def.ghi |NO "Anonymous" N Y abc |NO "Anonymous" N Y def.ghi |NO "Anonymous" N Y |NO "Anonymous" N N abc def.ghi |YES N N abc |YES > N N def.ghi |YES "Caller Name" @def.ghi> N N |YES "Caller Name" @> ASTERISK-25791 #close Reported-by: Anthony Messina Change-Id: I2c82a5ca1413c2c00fb62ea95b0ae8e97af54dc9 2016-03-03 17:34 +0000 [7cf7b0a4f9] gtjoseph * third_party/Makefile.rules: Replace unsupported != operator with $(shell ...) Apparently the != operator is fairly new so I've replaced it with the old $(shell ...) syntax. Change-Id: I16b2e1878a4f91e7e9740abd427f9639f933c479 Reported-by: Richard Mudgett 2016-01-23 15:50 +0000 [53f57001f2] gtjoseph * loader: Retry dlopen when loading fails Although we use the RTLD_LAZY flag when calling dlopen the first time on a module, this only defers resolution for function calls. Pointer references to functions are determined at link time so dlopen expects them to be there. Since we don't cross-module link, pointers to functions in other modules won't be available and dlopen will fail. Doing a "hardened" build also causes problems because it typically sets "-z now" on the ld command line which overrides RTLD_LAZY at run time. If the failing module isn't a GLOBAL_SYMBOLS module, then dlopen will be called again after all the GLOBAL_SYMBOLS modules have been loaded and they'll eventually resolve. If the calling module IS a GLOBAL_SYMBOLS module itself and a third module depends on it, then there's an issue because the second time through the dlopen loop, GLOBAL_SYMBOLS modules aren't given any special treatment and since the order in which dlopen is called isn't deterministic, the dependent may again be tried before the module it needs is loaded. Simple solution: Save modules that fail load_resource because of a dlopen error in a list and retry them immediately after the first pass. Keep retrying until the failed list is empty or we reach a #defined max retries. Error messages are suppressed until the final pass which also gets rid of those confusing error messages about module failures that are later corrected. Change-Id: Iddae1d97cd2f00b94e61662447432765755f64bb 2016-03-01 16:18 +0000 [40d9e9e238] Kevin Harwell * bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880 2016-03-01 18:08 +0000 [ff3da61c35] Kevin Harwell * res_pjsip_refer.c: Delay sending the initial SIP Notify with frag 100 During the transfer process, some phones (okay it was the Jitsi softphone, but maybe others are out there) send a "bye" immediately after receiving a SIP Notify. When a "bye" is received early for some types of transfers the transferer channel may no longer be available during late stage transfer processing. For instance, during an attended transfer involving stasis bridging at one point the created local channel looks for an associated swap channel in order to retrieve the stasis application name. If the transferer has hung up then the local channel will fail to find it. The local channel then has no way to know which stasis app to enter, so it fails and hangs up as well. Thus the transfer does not complete as expected. This patch delays the sending of the initial notify in order to give the transfer process enough time to gather the necessary data for a successful transfer. ASTERISK-25771 Change-Id: I09cfc9a5d6ed4c007bc70625e0972b470393bf16 2016-03-03 08:26 +0000 [26b8f2692e] Joshua Colp * res_pjsip_dtmf_info: NULL terminate the message body. PJSIP does not ensure that when printing the message body the buffer will be NULL terminated. This is problematic when searching for the signal and duration values of the DTMF. This change ensures the buffer is always NULL terminated. Change-Id: I52653a1a60c93092d06af31a27408d569cc98968 2016-03-01 20:03 +0000 [86d6e44cc1] gtjoseph * alembic: Fix downgrade and tweak for sqlite Downgrade had a few issues. First there was an errant 'update' statement in add_auto_dtmf_mode that looks like it was a copy/paste error. Second, we weren't cleaning up the ENUMs so subsequent upgrades on postgres failed because the types already existed. For sqlite... sqlite doesn't support ALTER or DROP COLUMN directly. Fortunately alembic batch_operations takes care of this for us if we use it so the alter and drops were converted to use batch operations. Here's an example downgrade: with op.batch_alter_table('ps_endpoints') as batch_op: batch_op.drop_column('tos_audio') batch_op.drop_column('tos_video') batch_op.add_column(sa.Column('tos_audio', yesno_values)) batch_op.add_column(sa.Column('tos_video', yesno_values)) batch_op.drop_column('cos_audio') batch_op.drop_column('cos_video') batch_op.add_column(sa.Column('cos_audio', yesno_values)) batch_op.add_column(sa.Column('cos_video', yesno_values)) with op.batch_alter_table('ps_transports') as batch_op: batch_op.drop_column('tos') batch_op.add_column(sa.Column('tos', yesno_values)) # Can't cast integers to YESNO_VALUES, so dropping and adding is required batch_op.drop_column('cos') batch_op.add_column(sa.Column('cos', yesno_values)) Upgrades from base to head and downgrades from head to base were tested repeatedly for postgresql, mysql/mariadb, and sqlite3. Change-Id: I862b0739eb3fd45ec3412dcc13c2340e1b7baef8 2016-03-02 15:55 +0000 [6f0d7ce9db] gtjoseph * config_transport: Fix objects returned by ast_sip_get_transport_states ast_sip_get_transport_states was returning a container of internal_state objects instead of ast_sip_transport_state objects. This was causing transport lookups to fail, most noticably in res_pjsip_nat, which couldn't find the correct external addresses. This was causing contacts to go out with internal ip addresses. ASTERISK-25830 #close Reported-by: Sean Bright Change-Id: I1aee6a2fd46c42e8dd0af72498d17de459ac750e 2016-03-02 11:17 +0000 [1ea7a5a774] Scott Griepentrog * CHAOS: cleanup possible null vars on msg alloc failure In message.c, if msg_alloc fails to init the string field, vars may be null, so use a null tolerant cleanup. In res_pjsip_messaging.c, if msg_data_create fails, mdata will be null, so use a null tolerant cleanup. ASTERISK-25323 Change-Id: Ic2d55c2c3750d5616e2a05ea92a19c717507ff56 2016-03-02 09:34 +0000 [3c37c7071f] Scott Griepentrog * CHAOS: prevent crash on failed strdup This patch avoids crashing on a null pointer if the strdup() allocation fails. ASTERISK-25323 Change-Id: I3f67434820ba53b53663efd6cbb42749f4f6c0f5 2016-02-29 18:11 +0000 [9633be9d25] Richard Mudgett * func_callerid.c: Update REDIRECTING reason documentation. Change-Id: I6e8d39b0711110a4bceafa652e58b30465e28386 2016-02-26 18:57 +0000 [4165ea7778] Richard Mudgett * SIP diversion: Fix REDIRECTING(reason) value inconsistencies. Previous chan_sip behavior: Before this patch chan_sip would always strip any quotes from an incoming reason and pass that value up as the REDIRECTING(reason). For an outgoing reason value, chan_sip would check the value against known values and quote any it didn't recognize. Incoming 480 response message reason text was just assigned to the REDIRECTING(reason). Previous chan_pjsip behavior: Before this patch chan_pjsip would always pass the incoming reason value up as the REDIRECTING(reason). For an outgoing reason value, chan_pjsip would send the reason value as passed down. With this patch: Both channel drivers match incoming reason values with values documented by REDIRECTING(reason) and values documented by RFC5806 regardless of whether they are quoted or not. RFC5806 values are mapped to the equivalent REDIRECTING(reason) documented value and is set in REDIRECTING(reason). e.g., an incoming RFC5806 'unconditional' value or a quoted string version ('"unconditional"') is converted to REDIRECTING(reason)'s 'cfu' value. The user's dialplan only needs to deal with 'cfu' instead of any of the aliases. The incoming 480 response reason text supported by chan_sip checks for known reason values and if not matched then puts quotes around the reason string and assigns that to REDIRECTING(reason). Both channel drivers send outgoing known REDIRECTING(reason) values as the unquoted RFC5806 equivalent. User custom values are either sent as is or with added quotes if SIP doesn't allow a character within the value as part of a RFC3261 Section 25.1 token. Note that there are still limitations on what characters can be put in a custom user value. e.g., embedding quotes in the middle of the reason string is silly and just going to cause you grief. * Setting a REDIRECTING(reason) value now recognizes RFC5806 aliases. e.g., Setting REDIRECTING(reason) to 'unconditional' is converted to the 'cfu' value. * Added missing malloc() NULL return check in res_pjsip_diversion.c set_redirecting_reason(). * Fixed potential read from a stale pointer in res_pjsip_diversion.c add_diversion_header(). The reason string needed to be copied into the tdata memory pool to ensure that the string would always be available. Otherwise, if the reason string returned by reason_code_to_str() was a user's reason string then the string could be freed later by another thread. Change-Id: Ifba83d23a195a9f64d55b9c681d2e62476b68a87 2016-02-26 18:54 +0000 [41f4af4ce5] Richard Mudgett * res_pjsip_send_to_voicemail.c: Allow either quoted or not send_to_vm reason. Change-Id: Id6350b3c7d4ec8df7ec89863566645e2b0f441fd 2016-02-29 20:41 +0000 [4c5998ff55] Richard Mudgett * res_pjsip_send_to_voicemail.c: Fix off-nominal double channel unref. * Fix double unref of other_party channel in off nominal path. * This is unlikely to be a real problem. However, for safety, in handle_incoming_request() keep the datastore ref with the other_party channel ref until we are finished with the other_party channel. Change-Id: I78f22547bf0bb99fb20814ceab75952bd857f821 2016-01-18 21:54 +0000 [b59956a875] gtjoseph * build-system: Allow building with static pjproject Background here: http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html From CHANGES: * To help insure that Asterisk is compiled and run with the same known version of pjproject, a new option (--with-pjproject-bundled) has been added to ./configure. When specified, the version of pjproject specified in third-party/versions.mak will be downloaded and configured. When you make Asterisk, the build process will also automatically build pjproject and Asterisk will be statically linked to it. Once a particular version of pjproject is configured and built, it won't be configured or built again unless you run a 'make distclean'. To facilitate testing, when 'make install' is run, the pjsua and pjsystest utilities and the pjproject python bindings will be installed in ASTDATADIR/third-party/pjproject. The default behavior remains building with the shared pjproject installation, if any. Building: All you have to do is include the --with-pjproject-bundled option on the ./configure command line (and remove any existing --with-pjproject option if specified). Everything else is automatic. Behind the scenes: The top-level Makefile was modified to include 'third-party' in the list of MOD_SUBDIRS. The third-party directory was created to contain any third party packages that may be needed in the future. Its Makefile automatically iterates over any subdirectories passing on targets. The third-party/pjproject directory was created to house the pjproject source distribution. Its Makefile contains targets to download, patch configure, generate dependencies, compile libs, apps and python bindings, sanitized build.mak and generate a symbols list. When bootstrap.sh is run, it automatically includes the configure.m4 file in third-party/pjproject. This file has a macro to download and conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR and PJPROJECT_BUNDLED. It also tests for the capabilities like PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to trying to compile. Of course, bootstrap.sh is only run once and the configure file is incldued in the patch. When configure is run with the new options, the macro in configure.m4 triggers the download, patch, conifgure and tests. No compilation is performed at this time. The downloaded tarball is cached in /tmp so it doesn't get downloaded again on a distclean. When make is run in the top-level Asterisk source directory, it will automatically descend all the subdirectories in third_party just as it does for addons, apps, etc. The top-level Makefile makes sure that the 'third-party' is built before 'main' so that dependencies from the other directories are built first. When main does build, a new shared library (libasteriskpj) is created that links statically to the pjproject .a files and exports all their symbols. The asterisk binary links to that, just as it does with libasteriskssl. When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject python bindings are installed in ASTDATADIR/third-party/pjproject. This will facilitate testing, including running the testsuite which will be updated to check that directory for the pjsua module ahead of the system python library. Modules should continue to depend on pjproject if they use pjproject APIs directly. They should not care about the implementation. No changes to any res_pjsip modules were made. Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103 2016-02-22 16:59 +0000 [18a323e542] Richard Mudgett * chan_sip.c: Fix T.38 issues caused by leaving a bridge. chan_sip could not handle AST_T38_TERMINATED frames being sent to it when the channel left the bridge. The action resulted in overlapping outgoing reINVITEs. The testsuite tests/fax/sip/directmedia_reinvite_t38 was not happy. * Force T.38 to be remembered as locally bridged. Now when the channel leaves the native RTP bridge after T.38, the channel remembers that it has already reINVITEed the media back to Asterisk. It just needs to terminate T.38 when the AST_T38_TERMINATED arrives. * Prevent redundant AST_T38_TERMINATED from causing problems. Redundant AST_T38_TERMINATED frames could cause overlapping outgoing reINVITEs if they happen before the T.38 state changes to disabled. Now the T.38 state is set to disabled before the reINVITE is sent. ASTERISK-25582 #close Change-Id: I53f5c6ce7d90b3f322a942af1a9bcab6d967b7ce 2016-02-18 18:27 +0000 [263a39f2cc] Richard Mudgett * res_pjsip_t38.c: Back out part of an earlier fix attempt. This backs out item 4 of the 4875e5ac32f5ccad51add6a4216947bfb385245d commit. Item 4 added the t38_bye_supplement. Unfortunately, the frame that it puts into the bridge may or may not be processed by the time the bridged peer is kicked out of the bridge. If it is processed then all is well. However, if it is not processed then that channel is stuck in fax mode until it hangs up or maybe if it joins another bridge for T.38 faxing. ASTERISK-25582 Change-Id: Ib20a03ecadf1bf8a0dcadfadf6c2f2e60919a9f7 2016-02-22 13:54 +0000 [221422be50] Richard Mudgett * bridge core: Add owed T.38 terminate when channel leaves a bridge. The channel is now going to get T.38 terminated when it leaves the bridging system and the bridged peers are going to get T.38 terminated as well. ASTERISK-25582 Change-Id: I77a9205979910210e3068e1ddff400dbf35c4ca7 2016-02-19 16:01 +0000 [0a5bc64491] Richard Mudgett * channel api: Create is_t38_active accessor functions. ASTERISK-25582 Change-Id: I69451920b122de7ee18d15bb231c80ea7067a22b 2016-02-19 19:06 +0000 [513638a5f4] Richard Mudgett * bridge_channel: Don't settle owed events on an optimization. Local channel optimization could cause DTMF digits to be duplicated. Pending DTMF end events would be posted to a bridge when the local channel optimizes out and is replaced by the channel further down the chain. When the real digit ends, the channel would get another DTMF end posted to the bridge. A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B 1) LocalA has the /n flag to prevent optimization. 2) B is sending DTMF to A through the local channel chain. 3) When LocalB optimizes out it can move B to the position of LocalB;1 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would settle an owed DTMF end to the bridge toward LocalA;2. 5) When B finally ends its DTMF it sends the DTMF end down the chain. 6) Without this patch, A would hear the DTMF digit end when LocalB optimizes out and when B ends the original digit. ASTERISK-25582 Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251 2016-02-22 12:15 +0000 [7c4495cb70] Richard Mudgett * channel.c: Route all control frames to a channel through the same code. Frame hooks can conceivably return a control frame in exchange for an audio frame inside ast_write(). Those returned control frames were not handled quite the same as if they were sent to ast_indicate(). Now it doesn't matter if you use ast_write() to send an AST_FRAME_CONTROL to a channel or ast_indicate(). ASTERISK-25582 Change-Id: I5775f41421aca2b510128198e9b827bf9169629b 2016-02-25 15:13 +0000 [48d713a832] gtjoseph * sorcery: Refactor create, update and delete to better deal with caches The ast_sorcery_create, update and delete function have been refactored to better deal with caches and errors. The action is now called on all non-caching wizards first. If ANY succeed, the action is called on all caching wizards and the observers are notified. This way we don't put something in the cache (or update or delete) before knowing the action was performed in at least 1 backend and we only call the observers once even if there were multiple writable backends. ast_sorcery_create was never adding to caches in the first place which was preventing contacts from getting added to a memory_cache when they were created. In turn this was causing memory_cache to emit errors if the contact was deleted before being retrieved (which would have populated the cache). ASTERISK-25811 #close Reported-by: Ross Beer Change-Id: Id5596ce691685a79886e57b0865888458d6e7b46 2016-02-25 15:39 +0000 [ee947d4a7a] gtjoseph * res_pjsip_mwi: Turn some NOTICEs and WARNINGs into debug 1s. There are a few cases where we're emitting notices or warnings for things that really need neither, like a client retrying to subscribe to mwi when they're not conifgured for it. They get a 404 so there's no need for non-debug messages. Change-Id: I05e38a7ff6c2f2521146f4be6a79731b9864e61f 2016-02-25 14:17 +0000 [6e70e8ccdb] gtjoseph * res_sorcery_memory_cache: Fix SEGV in some CLI commands A few of the CLI commands weren't checking for enough arguments and were SEGVing. Change-Id: Ie6494132ad2fe54b4f014bcdc112a37c36a9b413 2016-02-25 10:29 +0000 [4417f64d83] Leif Madsen * Add initial support to build Docker images This work-in-progress is the first step to being able to reliably build Asterisk containers from the Asterisk source. I'm submitting this based on feedback gained at AstriDevCon 2015. Information about how to use this is provided in contrib/docker/README.md and will result in a local Asterisk container being built right from your source. I believe this can eventually be automated via hub.docker.com. Change-Id: Ifa070706d40e56755797097b6ed72c1e243bd0d1 2016-02-22 19:31 +0000 [e7a6abbbd3] Richard Mudgett * rtp_engine.h: Remove extraneous semicolons. Change-Id: Ib462633d396fa941379dfef648dcd2245e350084 2016-02-23 14:57 +0000 [6656afffa0] Richard Mudgett * chan_sip.c: Suppress T.38 SDP c= line if addr is the same. Use the correct comparison function since we only care if the address without the port is the same. Change-Id: Ibf6c485f843a1be6dee58a47b33d81a7a8cbe3b0 2016-02-16 08:14 +0000 [ea9deff996] Christof Lauber * res_config_sqlite3: Fix crashes when reading peers from sqlite3 tables Introduced realloaction of ast_str buf in sqlite3_escape functions in case the returned buffer from threadstorage was actually too small. Change-Id: I3c5eb43aaade93ee457943daddc651781954c445 2016-02-11 11:01 +0000 [d2a1457e0b] gtjoseph * res_pjsip/config_transport: Allow reloading transports. The 'reload' mechanism actually involves closing the underlying socket and calling the appropriate udp, tcp or tls start functions again. Only outbound_registration, pubsub and session needed work to reset the transport before sending requests to insure that the pjsip transport didn't get pulled out from under them. In my testing, no calls were dropped when a transport was changed for any of the 3 transport types even if ip addresses or ports were changed. To be on the safe side however, a new transport option was added (allow_reload) which defaults to 'no'. Unless it's explicitly set to 'yes' for a transport, changes to that transport will be ignored on a reload of res_pjsip. This should preserve the current behavior. Change-Id: I5e759850e25958117d4c02f62ceb7244d7ec9edf 2016-02-07 17:34 +0000 [6b921f706d] gtjoseph * res_pjproject: Add ability to map pjproject log levels to Asterisk log levels Warnings and errors in the pjproject libraries are generally handled by Asterisk. In many cases, Asterisk wouldn't even consider them to be warnings or errors so the messages emitted by pjproject directly are either superfluous or misleading. A good exampe of this are the level-0 errors pjproject emits when it can't open a TCP/TLS socket to a client to send an OPTIONS. We don't consider a failure to qualify a UDP client an "ERROR", why should a TCP/TLS client be treated any differently? A config file for res_pjproject has bene added (pjproject.conf) and a new log_mappings object allows mapping pjproject levels to Asterisk levels (or nothing). The defaults if no pjproject.conf file is found are the same as those that were hard-coded into res_pjproject initially: 0,1 = LOG_ERROR, 2 = LOG_WARNING, 3,4,5 = LOG_DEBUG Change-Id: Iba7bb349c70397586889b8f45b8c3d6c6c8c3898 2016-02-18 10:55 +0000 [f295088764] Alexei Gradinari * res_pjsip_outbound_publish: Fix processing 412 response When Asterisk receives a 412 (Conditional Request Failed) response it has to recreate publish session. There is bug in res_pjsip_outbound_publish.c The function sip_outbound_publish_client_alloc is called with wrong object while processing 412 (Conditional Request Failed) response. This patch fixes it. ASTERISK-25229 #close Change-Id: I3b62f2debf6bb1e5817cde7b13ea39ef2bf14359 2016-02-18 11:15 +0000 [f1f79812c1] Mark Michelson * Fix failing threadpool_auto_increment test. The threadpool_auto_increment test fails infrequently for a couple of reasons * The threadpool listener was notified of fewer tasks being pushed than were actually pushed * The "was_empty" flag was set to an unexpected value. The problem is that the test pushes three tasks into the threadpool. Test expects the threadpool to essentially gather those three tasks, and then distribute those to the threadpool threads. It also expects that as the tasks are pushed in, the threadpool listener is alerted immediately that the tasks have been pushed. In reality, a task can be distributed to the threadpool threads quicker than expected, meaning that the threadpool has already emptied by the time each subsequent task is pushed. In addition, the internal threadpool queue can be delayed so that the threadpool listener is not alerted that a task has been pushed even after the task has been executed. From the test's point of view, there's no way to be able to predict exactly the order that task execution/listener notifications will occur, and there is no way to know which listener notifications will indicate that the threadpool was previously empty. For this reason, the test has been updated to only check the things it can check. It ensures that all tasks get executed, that the threads go idle after the tasks are executed, and that the listener is told the proper number of tasks that were pushed. Change-Id: I7673120d74adad64ae6894594a606e102d9a1f2c 2016-02-16 23:37 +0000 [79dc5e2f00] Rodrigo Ramírez Norambuena * app_queue: fix Calculate talktime when is first call answered Fix calculate of average time for talktime is wrong when is completed the first call beacuse the time for talked would be that call. ASTERISK-25800 #close Change-Id: I94f79028935913cd9174b090b52bb300b91b9492 2016-02-17 13:30 +0000 [5a3a857dd6] Richard Mudgett * cel.c: Fix mismatch in ast_cel_track_event() return type. The return type of ast_cel_track_event() is not large enough to return all 64 potential bits of the event enable mask. Fortunately, the defined CEL events do not really need all 64 bits and the return value is only used to determine if the requested CEL event is enabled. * Made the ast_cel_track_event() return 0 or 1 only so the return value can fit inside an int type instead of zero or a truncated 64 bit non-zero value. Change-Id: I783d932320db11a95c7bf7636a72b6fe2566904c 2016-02-16 16:37 +0000 [87ab65c557] gtjoseph * res_odbc: Fix exports.in for missing symbols res_odbc.exports.in was missing a few symbols. Changed to wildcards. Change-Id: Ieadd76df24e43ea92577f651d478a0f7b742c30c 2016-02-16 12:20 +0000 [c0f3062031] gtjoseph * res_statsd: Fix exports.in for missing symbols res_statsd.export.in was missing the _va variations of the log functions causing Asterisk to crash in res_pjsip if OPTIONAL_API wasn't enabled. ASTERISK-25727 #close Reported-by: Gergely Dömsödi Change-Id: I395729f9f51bdd33c5ca757f5f96ebedad74077b 2016-02-15 21:31 +0000 [5e848dae7b] gtjoseph * res_pjsip_config_wizard: Add command to export primitive objects A new command (pjsip export config_wizard primitives) has been added that will export all the pjsip objects it created to the console or a file suitable for reuse in a pjsip.conf file. ASTERISK-24919 #close Reported-by: Ray Crumrine Change-Id: Ica2a5f494244b4f8345b0437b16d06aa0484452b 2016-02-15 15:37 +0000 [34c64707d1] gtjoseph * res_pjsip_caller_id: Fix segfault when replacing rpid or pai header If the PJSIP_HEADER dialplan function adds a PAI or RPID header and send_rpid or send_pai is set, res_pjsip_caller_id attemps to retrieve, parse and modify the header added by the dialplan function. Since the header added by the dialplan function is generic string, there are no virtual functions to parse the uri and we get a segfault when we try. Since the modify, was really only an overwrite, we now just delete the old header if it was type PJSIP_H_OTHER and recreate it. This raises a question for another time though: What should happen with duplicate headers? Right now res_pjsip_header_funcs doesn't check for dups so if it's session supplement is loaded after res_pjsip_caller_id's (or any other module that adds headers), there'll be dups in the message. ASTERISK-25337 #close Change-Id: I5e296b52d30f106b822c0eb27c4c2b0e0f71c7fa 2016-02-15 13:08 +0000 [ebe167f792] Mark Michelson * Fix creation race of contact_status structures. It is possible when processing a SIP REGISTER request to have two threads end up creating contact_status structures in sorcery. contact_status is created using a "find or create" function. If two threads call into this at the same time, each thread will fail to find an existing contact_status, and so both will end up creating a new contact status. During testing, we would see sporadic failures because the PJSIP_CONTACT() dialplan function would operate on a different contact_status than what had been updated by res_pjsip/pjsip_options. The fix here is two-fold: 1) The "find or create" function for contact_status now has a lock around the entire operation. This way, if two threads attempt the operation simultaneously, the first to get there will create the object, and the second will find the object created by the first thread. 2) res_sorcery_memory has had its create callback updated so that it will not allow for objects with duplicate IDs to be created. Change-Id: I55b1460ff1eb0af0a3697b82d7c2bac9f6af5b97 2016-02-15 12:52 +0000 [1c4f2a920d] Joshua Colp * res_pjsip_pubsub: Move where the subscription is stored to after initialized. A problem arose when testing the AMI subscription listing actions where it was possible for a subscription that had not been fully initialized to be listed. This was problematic as the underlying listing code would crash. This change makes it so the subscription tree is fully set up before it is added to the list of subscriptions. This ensures that when the listing actions get the subscription it is valid. ASTERISK-25738 #close Change-Id: Iace2b13641c31bbcc0d43a39f99aba1f340c0f48 2015-02-20 20:51 +0000 [ac00c6bc2d] Corey Farrell * main/asterisk.c: Reverse #if statement in listener() to fix code folding. listener() opens the same code block in two places (#if and #else). This confuses some folding editors causing it to think that an extra code block was opened. Folding in 'geany' causes all code after listener() to be folded as if it were part of that procedure. ASTERISK-24813 #close Change-Id: I4b8c766e6c91e327dd445e8c18f8a6f268acd961 2016-02-09 17:34 +0000 [b1b797e0e7] gtjoseph * res_pjsip: Refactor load_module/unload_module load_module was just too hairy with every step having to clean up all previous steps on failure. Some of the pjproject init calls have now been moved to a separate load_pjsip function and the unload_pjsip function was enhanced to clean up everything if an error happened at any stage of the load process. In the process, a bunch of missing pj_shutdowns, serializer_pool_shutdowns and ast_threadpool_shutdowns were also corrected. Change-Id: I5eec711b437c35b56605ed99537ebbb30463b302 2016-02-09 22:42 +0000 [20e9792fbc] Badalyan Vyacheslav * Resources/res_phoneprov: fix memory leak and heap-use-after-free * heap-use-after-free happens when we free "cfg" but then use "value" which refers to it * A memory leak occurs because in some cases it is not released "defaults" ASTERISK-25721 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3807d3f4726df6864430ec144cf6265d3f538469 2016-02-11 11:21 +0000 [962a9d61f8] Etienne Lessard (license #6394) * func_iconv: Ensure output strings are properly terminated. ASTERISK-25272 #close Reported by: Etienne Lessard patches: AST-25272.patch submitted by Etienne Lessard (license #6394) Change-Id: Id75ad202300960a1e91afe15e319d992936ecc17 2016-02-10 16:16 +0000 [c1bf014ea0] gtjoseph * res_pjsip: Handle pjsip_dlg_create_uas deprecation Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically increments the lock on the returned dialog. To account for this, configure.ac now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use the original call or the new one. If the new one was used, the ref count is decremented before returning. ASTERISK-25751 #close Reported-by Josh Colp Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8 2016-02-09 23:40 +0000 [bd07b6f0dd] Badalyan Vyacheslav * Build: Added testing compiler to support the system sanitizes In older versions of the compiler was not sanitizes. Compilers other than GCC can not support the Usan and TSAN or have other options for *FLAGS. ASTERISK-25767 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916 2016-02-09 20:57 +0000 [e9e896abd1] Badalyan Vyacheslav * Build: Fix menuselect USAN conflicts USAN can be used together with other sanitizers. Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f 2016-02-09 14:21 +0000 [93e8ed0154] Corey Farrell * Simplify and fix conditional in FD_SET. FD_SET contains a conditional statement to protect against buffer overruns. The statement was overly complicated and prevented use of the last array element of ast_fdset. We now just verify the fd is less than ast_FDMAX. Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40 2016-02-09 07:11 +0000 [a7c8d4cd6b] Joshua Colp * tests/test_sorcery_memory_cache_thrash: Improve termination process. When terminating the threads thrashing a sorcery memory cache each would be told to stop and then we would wait on them. During at least one thrashing test this was problematic due to the specific usage pattern in use. It would take some time for termination of the thread to occur. This would occur due to contention between the threads retrieving and the threads updating the cache. As the retrieving threads are given priority it may be some time before the updating threads are able to proceed. This change makes it so all threads are told to stop and then each are joined to ensure they stop. This way all the threads should stop at around the same time instead of waiting for one to stop, the next to stop, then the next, and so on. As a result of this the execution time for each thrash test is much closer to their expected value than previously seen as well. Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827 2016-01-29 17:56 +0000 [2451d4e455] gtjoseph * res_pjsip: Fix infinite recursion when loading transports from realtime Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19 2016-01-25 17:36 +0000 [6f978fbfe5] Richard Mudgett * app_confbridge: Only use b_profile options from the conference. A user cannot set new bridge options after the conference is created by the first user. Attempting to do so is documented as undefined behavior. This patch ensures that the bridge profile options used are from the conference and not what a subsequent user may have tried to set. Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266 2016-02-05 10:29 +0000 [ec8fd6714d] gtjoseph * chan_misdn: Fix a few issues causing compile errors Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98 2016-02-04 16:17 +0000 [6a799cd78f] Mark Michelson * Check for OpenSSL defines before trying to use them. The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior to OpenSSL version 1.0.1. A recent commit attempts to, by default, set these options, which can cause problems on systems with older OpenSSL installations. This commit adds a configure script check for those defines and will not attempt to make use of those if they do not exist. We will print a warning urging the user to upgrade their OpenSSL installation if those defines are not present. Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d 2016-02-03 14:25 +0000 [953d1cc11a] gtjoseph * pjsip/alembic: Add missing columns to system and registration ps_systems needed disable_tcp_switch ps_registrations needed line and endpoint ASTERISK-25737 #close Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19 2016-02-04 11:39 +0000 [23829b3253] Mark Michelson * res_stasis_device_state: Fix refcounting error. Device state subscription lifetimes were governed by when the subscription was established and unsubscribed from. However, it is possible that at the time of unsubscription, there could be device state events still in flight. When those device state events occur, the device state callback could attempt to dereference a freed pointer. Crash. This change ensures that the lifetime of the device state subscription does not end until the underlying stasis subscription has confirmed that its final message has been sent. Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2 2016-01-27 10:44 +0000 [4e8e6d3922] Sean Bright * res_rtp_asterisk: Allow ICE host candidates to be overriden During ICE negotiation the IPs of the local interfaces are sent to the remote peer as host candidates. In many cases Asterisk is behind a static one-to-one NAT, so these host addresses will be internal IP addresses. To help in hiding the topology of the internal network, this patch adds the ability to override the host candidates by matching them against a user-defined list of replacements. Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f 2015-12-07 12:46 +0000 [c6b1b2b1c8] Richard Mudgett * AST-2016-003 udptl.c: Fix uninitialized values. Sending UDPTL packets to Asterisk with the right amount of missing sequence numbers and enough redundant 0-length IFP packets, can make Asterisk crash. ASTERISK-25603 #close Reported by: Walter Doekes ASTERISK-25742 #close Reported by: Torrey Searle Change-Id: I97df8375041be986f3f266ac1946a538023a5255 2016-02-03 12:05 +0000 [f8acadde2c] Joshua Colp * AST-2016-001 http: Provide greater control of TLS and set modern defaults. This change exposes the configuration of various aspects of the TLS support and sets the default to the modern standards. The TLS cipher is now set to the best values according to the Mozilla OpSec team, different TLS versions can now be disabled, and the cipher order can be forced to be that of the server instead of the client. ASTERISK-24972 #close Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8 2015-09-28 17:07 +0000 [3c81a052c8] Richard Mudgett * AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. Setting the sip.conf timert1 value to a value higher than 1245 can cause an integer overflow and result in large retransmit timeout times. These large timeout times hold system file descriptors hostage and can cause the system to run out of file descriptors. NOTE: The default sip.conf timert1 value is 500 which does not expose the vulnerability. * The overflow is now detected and the previous timeout time is calculated. ASTERISK-25397 #close Reported by: Alexander Traud Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290 2016-02-03 14:07 +0000 [2a6ee8caeb] gtjoseph * logging: Remove/fix some message annoyances test_dlinklists doesn't need to NOTICE everyone that every macro worked. res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or provider was registered. res_odbc was missing a newline at the end of one message. Change-Id: I6c06361518ef3711821795e535acd439782a995e 2016-02-02 10:52 +0000 [32fc784284] Alexei Gradinari License #5691 * res_sorcery_realtime: Fix regex regression. A regression was introduced where searching for realtime PJSIP objects by regex by starting the regex with a leading "^" would cause no items to be returned. This was due to a change which attempted to drop the requirement for a leading "^" to be present due to how some CLI commands formulate their regexes. However, the change, rather than simply eliminating the requirement, caused any regexes that did begin with "^" to end up not returning the expected results. This change fixes the problem by inspecting the regex and formulating the realtime query differently depending on if it begins with "^". ASTERISK-25702 #close Reported by Nic Colledge Patches: realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693 2016-02-02 04:05 +0000 [0405c31756] Karsten Wemheuer * res_xmpp: Does not connect in component mode The module res_xmpp does not accept usernames in the form used in component mode (XEP-0114). In component mode there is no @something in the name. In component mode the connection is now not dropped anymore. If the xmpp server sends out a "stream" tag before handshake is finished, the connection gets dropped in res_xmpp. Now this tag will be ignored and the connection will be established. After connecting there will be an exchange of presence states. This does not work as expected in component mode. The responsible function "xmpp_pak_presence" is left before the states get sent out. Sending presence states in component mode is now moved to the top of the function. ASTERISK-25735 #close Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c 2016-02-01 13:04 +0000 [8804d0973c] gtjoseph * build_system: Fix some warnings highlighted by clang Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd 2016-02-01 13:16 +0000 [109b0aff6b] gtjoseph * res/Makefile: Fix bug in "clean" target for ari The "clean" target was attempting to clean res/ari from inside the res directory which doesn't remove anything. Removed the res/ prefix. Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6 2016-01-31 20:13 +0000 [a85fab7c44] gtjoseph * pjsip/alembic: Fix definition of qualify_timeout A recent commit set qualify_timeout to Decimal which isn't supported. This path corrects it to Float. Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf 2016-01-29 07:39 +0000 [aa9348ab9a] Stefan Engström * chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) AMI action: SIPnotify or b) cli command: sip notify , I expect asterisk to include the same value for its own ip in both cases a) and b), but it seems a) produces a contact header like Contact: whereas b) produces a contact header like . 0.0.0.0:8060 is my udpbindaddr in sip.conf My guess is that manager_sipnotify should call ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, because after applying this patch, both cases a) and b) produce the contact header that I expect: Reported by: Stefan Engström Tested by: Stefan Engström Change-Id: I86af5e209db64aab82c25417de6c768fb645f476 2015-12-23 15:07 +0000 [65bd4fcc3f] Mark Michelson * res_odbc: Remove connection management Asterisk by default will create a single database connection and share it among all threads that attempt to access the database. In previous versions of Asterisk, this was tolerable, because the most used channel driver, chan_sip, mostly accessed the database from a single thread. With PJSIP, however, many threads may be attempting to perform database operations, and there is the potential for many more database accesses, meaning the concurrency is a horrible bottleneck if only one connection is shared. Asterisk has a connection pooling facility built into it, but the implementation has flaws. For one, there is a strict limit on the number of simultaneous connections that could be made to the database. Anything beyond the maximum would result in a failed operation. Attempting to predict what the maximum should be is nearly impossible even for someone intimately familiar with Asterisk's threading model. In addition, use of transactions in the dialplan can cause some severe bugs if connection pooling is enabled. This commit seeks to fix the concurrency problem by removing all connection management code from Asterisk and leaving that to the underlying unixODBC code instead. Now, Asterisk does not share a single connection, nor does it try to maintain a connection pool. Instead, all Asterisk ever does is request a connection from unixODBC and allow unixODBC to either allocate those connections or retrieve them from a pool. Doing this has a bit of a ripple effect. For one, since connections are not long-lived objects, several of the safeguards that previously existed have been removed. We don't have to worry about trying to use a connection that has gone stale. In every case, when we request a connection, it has just been made and we don't need to perform any sanity checks to be sure it's still active. Another major player affected by this change is transactions. Transactions and their respective connections were so tightly coupled that it was almost pornographic. This code change moves transaction-related code to its own file separate from the core ODBC functionality. This way, the core of ODBC does not even have to know that transactions exist. In making this large change, I had to look at a lot of code and understand it. When making this change, I discovered several places where the behavior is definitely not ideal, but it seemed outside the scope of this change to be fixing it. Instead, any place where I saw some sort of room for improvement has had a XXX comment added explaining what could be altered to improve it. Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf 2016-01-28 12:44 +0000 [2a9e623ff9] Richard Mudgett * config_options.c: Fix warning message wording. Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4 2016-01-25 17:34 +0000 [ed3c9c1512] Richard Mudgett * app_confbridge.c: Replace inlined code with existing function. Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51 2016-01-25 16:05 +0000 [1d0abf86e7] Richard Mudgett * app_confbridge: Add ability to get the muted conference state. * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. * Added Muted header to AMI ConfbridgeListRooms action response list events to indicate the muted conference state. * Added Muted column to CLI "confbridge list" output to indicate the muted conference state and made the locked column a yes/no value instead of a locked/unlocked value. ASTERISK-20987 Reported by: hristo Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1 2016-01-26 17:59 +0000 [f0d40afa69] Richard Mudgett * app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7 2016-01-25 15:48 +0000 [3e51e5c7fd] Richard Mudgett * app_confbridge: Make non-admin users join a muted conference muted. ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38 2016-01-27 13:02 +0000 [9da18af992] gtjoseph * res_pjsip: Add res_pjproject dependency to UPGRADE.txt and samples Since res_pjsip now depends on res_pjproject, this is now mentioned in UPGRADE.txt and the basic-pbx modules.conf has been updated. Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d 2016-01-27 10:29 +0000 [aee8448bc2] gtjoseph * build_system: Prevent goals needing makeopts from running when it's missing The Makefile only optionally includes makeopts so when goals like uninstall that dont depend on anything else are run after a distclean, rules like 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts to remove everything in the root directory. Although there's a rule defined for makeopts which prints a message and does an 'exit 1', since '-include makepopts' was specified (with the -), the exit was ignored letting the rest of the rules run. This patch makes makeopts required unless the goal has the string 'clean' in it. ASTERISK-25730 #close Reported-by: George Joseph Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7 2016-01-25 09:35 +0000 [f22074e5d9] Joshua Colp * config: Allow options to register when documentation is unavailable. The config options framework is strict in that configuration options must be documented unless XML documentation support is not available. In practice this is useful as it ensures documentation exists however in off-nominal cases this can cause strange problems. If it is expected that a config option has a non-zero or non-empty default value but the config option documentation is unavailable this reasonable expectation will not be met. This can cause obscure crashes and weirdness depending on how the code handles it. This change tweaks the behavior to ensure that the config option is still allowed to register, apply default values, and be set when devmode is not enabled. If devmode is enabled then the option can NOT be set. This also does not remove the initial documentation error message that is output on load when registering the configuration option. ASTERISK-25725 #close Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8 2016-01-25 10:23 +0000 [4a3275abb9] Mark Michelson * Stasis: Use custom structure when setting variables. A recent change to queue channel variable setting to the Stasis control queue caused a regression. When setting channel variables, it is possible to give a NULL channel variable value in order to unset the variable (i.e. remove it from the channel variable list). The change introduced a call to ast_variable_new(), which is not tolerant of NULL channel variable values. This new change switches from using ast_variable to using a custom channel variable struct that is lighter weight and NULL value-tolerant. Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d 2016-01-25 16:56 +0000 [b2c8a99f9e] Rusty Newton * sounds/Makefile: Incremented core and extra sounds versions to 1.5 Core and extra sounds 1.5 was recently released! The tarballs contain change descriptions however I figure more people will see this one so I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra to Core for en, en_GB, fr and added for languages that didn't already have Extra sound sets (it,ja,ru). In addition all of the English and Russian sounds have been completely re-recorded. Sounds moved and added: activated,added,all-circuits-busy-now,astcc-followed-by-pound at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy ,call-fwd-unconditional,calling,call-waiting,cancelled, cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to ,location,number,number-not-answering,num-was-successfully,one-moment-please ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered ,your There were also a few random fixes here and there to file names for a few of the languages. ASTERISK-25068 #close Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3 2016-01-25 16:51 +0000 [8261bda1bf] Mark Michelson * res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. A test recently uncovered that running an ill-timed AMI command to show inbound subscriptions could cause a crash since Asterisk will try to operate on a freed subscription. The fix for this is to remove the subscription tree from the list of subscriptions at the time that we are sending our final NOTIFY request out. This way, as the subscription is in the process of dying, it is inaccessible from AMI. Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23 2016-01-25 11:03 +0000 [a6823bb0c4] Corey Farrell * chan_sip: Fix buffer overrun in sip_sipredirect. sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer of 256 characters. This patch reduces the copy to 255 characters to leave room for the string null terminator. ASTERISK-25722 #close Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab 2016-01-22 15:08 +0000 [1003c2eb05] Mark Michelson * Stasis: Fix potential memory leak of control data. When queuing tasks onto the Stasis control queue, you can pass an arbitrary data pointer and a function to free that data. All ARI commands that use the Stasis control queue made the assumption that the destructor function would be called in all paths, whether the task was queued successfully or not. However, this was not correct. If a task was queued onto a control structure that was already completed, the allocated data would not be freed properly. This patch corrects this by making sure that all return paths call the data destructor. Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb 2016-01-21 10:58 +0000 [eedd77fda0] Mark Michelson * Stasis: Use control queue to prevent crash. A crash occurred when attempting to set a channel variable on a channel that had already been hung up. This is because there is a small window between when a control is grabbed and when the channel variable is set that the channel can be hung up. The fix here is to queue the setting of the channel variable onto the control queue. This way, the manipulation of the channel happens in a thread where it is safe to be done. In this change, I also noticed that the setting of bridge roles on channels was being done outside of the control queue, so I also changed those operations to be done in the control queue. ASTERISK-25709 #close Reported by Mark Michelson Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78 2016-01-22 11:48 +0000 [1c95b211a0] Richard Mudgett * logger.c: Fix buffer overrun found by address sanitizer. The null terminator of the tail struct member was not being allocated when no logger.conf config file is installed. ASTERISK-25714 #close Reported by: Badalian Vyacheslav Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30 2016-01-21 16:40 +0000 [6ff945ab87] Corey Farrell * Build System: Add support for checking alembic branches. * Add 'check-alembic' target to root Makefile. * Create build_tools/make_check_alembic to do the actual checks. ASTERISK-25685 Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004 2016-01-19 18:20 +0000 [02035212de] Richard Mudgett * res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. ASTERISK-25712 #close Reported by: Richard Mudgett Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f 2016-01-18 03:49 +0000 [c68c66c61f] Diederik de Groot * main/asterisk.c: ast_el_read_char Make sure buf[res] is not accessed at res=-1 (buffer underrun). Address Sanitizer will complain about this quite loudly. ASTERISK-24801 #close Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9 2016-01-13 16:49 +0000 [f87c3275cc] Richard Mudgett * res_pjsip: Add CLI "pjsip dump endpt [details]" Dump the res_pjsip endpt internals. In non-developer mode we will not document or make easily accessible the "details" option even though it is still available. The user has to know it exists to use it. Presumably they would also be aware of the potential crash warning below. Warning: PJPROJECT documents that the function used by this CLI command may cause a crash when asking for details because it tries to access all active memory pools. Change-Id: If2d98a3641c9873364d1daaad971376311aef3cb 2016-01-18 17:16 +0000 [46b2de55f9] Matt Jordan * funcs/func_cdr: Correctly report high precision values for duration and billsec When CDRs were refactored, func_cdr's ability to report high precision values for duration and billsec (the 'f' option) was broken. This was due to func_cdr incorrectly interpreting the duration/billsec values provided by the CDR engine in milliseconds, as opposed to seconds. Since the CDR engine only provides duration and billsec in seconds, and does not expose either attribute with sufficient precision to merely pass back the underlying value, this patch fixes the bug by re-calculating duration and billsec with microsecond precision based on the start/answer/end times on the CDR. ASTERISK-25179 #close Change-Id: I8bc63822b496537a5bf80baf6102c06206bee841 2016-01-18 19:20 +0000 [137fe5ae01] gtjoseph * res_pjproject: Add module providing pjproject logging and utils res_pjsip_log_forwarder has been renamed to res_pjproject and enhanced as follows: As a follow-on to the recent 'Add CLI "pjsip show buildopts"' patch, a new ast_pjproject_get_buildopt function has been added. It allows the caller to get the value of one of the buildopts. The initial use case is retrieving the runtime value of PJ_MAX_HOSTNAME to insure we don't send a hostname greater than pjproject can handle. Since it can differ between the version of pjproject that Asterisk was compiled against and the version of pjproject that Asterisk is running against, we can't use the PJ_MAX_HOSTNAME macro directly in Asterisk source code. Change-Id: Iab6e82fec3d7cf00c1cf6185c42be3e7569dee1e 2016-01-19 17:15 +0000 [b5c13c1545] Joshua Colp * test_threadpool: Wait for each task to complete and fix memory leak. This change makes the thread_timeout_thrash unit test wait for each task to complete. This fixes the problem where the test would prematurely end when all threads were gone and a new one had to be started to handle the last task. It also increases the thrasing as it is now more likely for each task to encounter the above scenario. This also fixes a memory leak where the data for each task was not being freed. ASTERISK-25611 #close Change-Id: I5017d621a4dc911f509074c16229b86bff2fb3c6 2016-01-18 19:44 +0000 [0ab89182d9] Richard Mudgett * taskprocessor.c: Increase CLI "core ping taskprocessor" timeout. Change-Id: I4892d6acbb580d6c207d006341eaf5e0f8f2a029 2016-01-18 19:43 +0000 [a2a8ea3330] Richard Mudgett * taskprocessor.c: Fix some taskprocessor unrefs. You have to call ast_taskprocessor_unref() outside of the taskprocessor implementation code. Taskprocessor use since v12 has become more transient than just the singleton uses in earlier versions. Change-Id: If7675299924c0cc65f2a43a85254e6f06f2d61bb 2016-01-19 13:44 +0000 [d604a9afc8] Richard Mudgett * Fix alembic branches on v13. Change-Id: I313449b609ede18ad1e1763a655dd23b9210a8e0 2016-01-18 18:45 +0000 [a0c79f3a4f] gtjoseph * pjsip_loging_refactor: Rename res_pjsip_log_forwarder to res_pjproject Change-Id: I5387821f29e5caa0cba0b7d62b0fc0d341e7e20b 2016-01-14 09:26 +0000 [018ccf680b] Rusty Newton * func_channel: Add help text for undocumented CHANNEL function arguments Adding help text documentation for: * hangupsource * appname * appdata * exten * context * channame * uniqueid * linkedid ASTERISK-24097 #close Reported by: Steven T. Wheeler Tested by: Rusty Newton Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d 2016-01-16 13:18 +0000 [5644bca9f9] Daniel Journo * Update version number in features.conf.sample Update the version number in the comments from Asterisk 12 to Asterisk 12+ Change-Id: Ie692ac8cda3c993c3bf10f27f51a1cca3317ec7b 2016-01-15 19:52 +0000 [3f5f30cf82] Corey Farrell * main/config: Clean config maps on shutdown. ASTERISK-25700 #close Change-Id: I096da84f9c62c6095f68bcf98eac4b7c7868e808 2016-01-14 14:42 +0000 [660fedecb7] Kevin Harwell * bridge_basic: don't cache xferfailsound during an attended transfer The xferfailsound was read from the channel at the beginning of the transfer, and that value is "cached" for the duration of the transfer. Therefore, changing the xferfailsound on the channel using the FEATURE() dialplan function does nothing once the transfer is under way. This makes it so the transfer code instead gets the xferfailsound configuration options from the channel when it is actually going to be used. This patch also fixes a potential memory leak of the props object as well as making sure the condition variable gets initialized before being destroyed. ASTERISK-25696 #close Change-Id: Ic726b0f54ef588bd9c9c67f4b0e4d787934f85e4 2015-07-10 10:37 +0000 [9cda1de34d] Richard Mudgett * taskprocessor.c: Simplify ast_taskprocessor_get() return code. Change-Id: Id5bd18ef1f60ef8be453e677e98478298358a9d1 2016-01-13 18:20 +0000 [a79af2b312] Richard Mudgett * astmm.c: Add more stats to CLI "memory show" commands. * Add freed regions totals to allocations and summary. * Add totals for all allocations and not just the selected allocations. Change-Id: I61d5a5112617b0733097f2545a3006a344b4032a 2016-01-14 16:00 +0000 [83feb7db3b] Kevin Harwell * bridge_basic: don't play an attended transfer fail sound after target hangs up If the attended transfer destination answers (picks call up or goes to voicemail) and then hangs up on the transferer then transferer hears the fail sound. This patch makes it so the fail sound is not played when the transfer destination/target hangs up after answering. ASTERISK-25697 #close Change-Id: I97f142fe4fc2805d1a24b7c16143069dc03d9ded 2016-01-14 13:22 +0000 [935d641f3b] Mark Michelson * Remove res/ari/* content during 'make clean'. 'make clean' and 'make distclean' can leave behind .o files in the res/ari/ directory. One observed consequence of this is that running Asterisk with MALLOC_DEBUG can cause Asterisk to crash immediately on startup sometimes. By ensuring that we are making a clean build, we can be sure that stale files are not being included in the build and causing problems when build options should have caused files to be re-built. ASTERISK-25683 #close Reported by yaron nahum Change-Id: I1f48baa904d2468eddeefb42ee68a56af7adc7b7 2016-01-13 15:58 +0000 [46f21df302] Daniel Journo * pjsip/alembic: Fix qualify_timeout column definition Corrects the qualify_timeout column type from Integer to Decimal ASTERISK-25686 #close Reported-by: Marcelo Terres Change-Id: I757d0e3c011ee9be6cd5abd48bc92441a405d3c8 2016-01-12 11:14 +0000 [32b29d7b02] Joshua Colp * app: Queue hangup if channel is hung up during sub or macro execution. This issue was exposed when executing a connected line subroutine. When connected or redirected subroutines or macros are executed it is expected that the underlying applications and logic invoked are fast and do not consume frames. In practice this constraint is not enforced and if not adhered to will cause channels to continue when they shouldn't. This is because each caller of the connected or redirected logic does not check whether the channel has been hung up on return. As a result the the hung up channel continues. This change makes it so when the API to execute a subroutine or macro is invoked the channel is checked to determine if it has hung up. If it has then a hangup is queued again so the caller will see it and stop. ASTERISK-25690 #close Change-Id: I1f9a8ceb1487df0389f0d346ce0f6dcbcaf476ea 2016-01-13 07:20 +0000 [e7cfda0b38] Sean Bright * res_musiconhold: Prevent multiple simultaneous reloads. There are two ways in which the reload() function in res_musiconhold can be called from the CLI: * module reload res_musiconhold.so * moh reload In the former case, the module loader holds a lock that prevents multiple concurrent calls, but in the latter there is no such protection. This patch changes the 'moh reload' CLI command to invoke the module loader directly, rather than call reload() explicitly. ASTERISK-25687 #close Change-Id: I408968b4c8932864411b7f9ad88cfdc7b9ba711c 2016-01-12 14:25 +0000 [5586abc957] Richard Mudgett * res_pjsip_log_forwarder.c: Add CLI "pjsip show buildopts". PJPROJECT has a function available to dump the compile time options used when building the library. * Add CLI "pjsip show buildopts" command. * Update contrib/scripts/autosupport to get pjproject information. Change-Id: Id93a6a916d765b2a2e5a1aeb54caaf83206be748 2016-01-12 10:36 +0000 [4cd58c3b20] Mark Michelson * res_sorcery_realtime: Remove leading ^ requirement. res_sorcery_realtime's search-by-regex callback performed a check to ensure that the passed-in regex began with a caret (^). If it did not, then no results would be returned. This callback only started to become used when "like" support was added to PJSIP CLI commands. The CLI command for listing objects would pass an empty regex ("") to the sorcery backend if no "like" statement was present. For most sorcery backends, this resulted in returning all objects. However, for realtime, this resulted in returning no objects. This commit seeks to fix the regression by removing the requirement from res_sorcery_realtime for the passed-in-regex to begin with a caret. ASTERISK-25689 #close Reported by Marcelo Terres Change-Id: I22b4dc5d7f3f11bb29ac2e42ef94682e9bab3b20 2016-01-07 11:57 +0000 [219c204a41] gtjoseph * pjsip_sdp_rtp: Add option endpoint/bind_rtp_to_media_address On a system with multiple ip addresses in the same subnet, if a transport is bound to a specific ip address and endpoint/media_address is set, the SIP/SDP will have the correct address in all fields but the rtp stream MAY still originate from one of the other ip addresses, most probably the "primary" ip address. This happens because res_pjsip_sdp_rtp/create_rtp always calls ast_instance_new with the "all" ip address (0.0.0.0 or ::). The new option causes res_pjsip_sdp_rtp/create_rtp to call ast_rtp_instance_new with the endpoint's media_address (if specified) instead of the "all" address. This causes the packets to originate from the specified address. ASTERISK-25632 ASTERISK-25637 Reported-by: Olivier Krief Reported-by: Dan Journo Change-Id: I3dfaa079e54ba7fb7c4fd1f5f7bd9509bbf8bd88 2016-01-10 16:22 +0000 [22801a06ee] Daniel Journo * pjsip: Add option global/regcontext Added new global option (regcontext) to pjsip. When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given endpoint who registers or unregisters with us. ASTERISK-25670 #close Reported-by: Daniel Journo Change-Id: Ib1530c5b45340625805c057f8ff1fb240a43ea62 2016-01-08 15:22 +0000 [1600ebca7d] Kevin Harwell * pbx: Deadlock between contexts container and context_merge locks Recent changes (ASTERISK-25394 commit 2bd27d12223fe33b58c453965ed5c6ed3af7c4f5) introduced the possibility of a deadlock. Due to the mentioned modifications ast_change_hints now needs to keep both merge/delete and state callbacks from occurring while it executes. Unfortunately, sometimes ast_change_hints can be called with the contexts container locked. When this happens it's possible for another thread to grab the context_merge_lock before the thread calling into ast_change_hints does and then try to obtain the contexts container lock. This of course causes a deadlock between the two threads. The thread calling into ast_change_hints waits for the other thread to release context_merge_lock and the other thread is waiting on that one to release the contexts container lock. Unfortunately, there is not a great way to fix this problem. When hints change, the subsequent state callbacks cannot run at the same time as a merge/delete, nor when the usual state callbacks do. This patch alleviates the problem by having those particular callbacks (the ones run after a hint change) occur in a serialized task. By moving the context_merge_lock to a task it can now safely be attempted or held without a deadlock occurring. ASTERISK-25640 #close Reported by: Krzysztof Trempala Change-Id: If2210ea241afd1585dc2594c16faff84579bf302 2016-01-10 17:08 +0000 [0fc3dad965] Corey Farrell * devicestate: Cleanup engine thread during graceful shutdown. ASTERISK-25681 #close Change-Id: I64337c70f0ebd8c77f70792042684607c950c8f1 2016-01-10 13:51 +0000 [f34dd10495] Corey Farrell * manager: Cleanup manager_channelvars during shutdown. ASTERISK-25680 #close Change-Id: I3251d781cbc3f48a6a7e1b969ac4983f552b2446 2016-01-10 13:27 +0000 [1d3a1167fc] Corey Farrell * res_calendar: Cleanup scheduler context at unload. ASTERISK-25679 #close Change-Id: I839159bf6882cccc1b23494c7aa2bc2a2624613f 2016-01-08 11:49 +0000 [3a160cdbf6] Joshua Colp * res_rtp_asterisk: Revert DTLS negotiation changes. Due to locking issues within pjnath these changes are being reverted until pjnath can be changed. ASTERISK-25645 Revert "res_rtp_asterisk.c: Fix DTLS negotiation delays." This reverts commit 24ae124e4f7310cfa64c187b944b2ffc060da28d. Change-Id: I2986cfb2c43dc14455c1bcaf92c3804f9da49705 Revert "res_rtp_asterisk: Resolve further timing issues with DTLS negotiation" This reverts commit 965a0eee46d24321f74c244e23c5a5f45e67e12b. Change-Id: Ie68fafde27dad4b03cb7a1e27ce2a8502c3f7bbe 2016-01-09 17:57 +0000 [4b10fc9173] gtjoseph * Revert "pjsip_location: Delete contact_status object when contact is deleted" This reverts commit 0a9941de9d24093b5ff44096d1d7406f29d11e45. Matt, This patch causes another problem and should not have been needed. Before this patch, persistent_endpoint_contact_deleted_observer WAS deleting the contact_status when ast_sip_location_delete_contact was called. By deleting it yourself in ast_sip_location_delete_contact it was gone before the observer could run and the observer therefore was throwing an error and not sending stasis/AMI/statsd messages. So, I don't think this was the cause of your original issue. I also had verified the contact AMI and statsd lifecycle and it was working. I'll double check now though. ASTERISK-25675 Reported-by: Daniel Journo Change-Id: Ib586a6b7f90acb641b0c410f659743ab90e84f1a 2016-01-09 18:04 +0000 [79b4309881] Corey Farrell * pbx_dundi: Run cleanup on failed load. During failed startup of pbx_dundi no cleanup was performed. Add a call to unload_module before returning AST_MODULE_LOAD_DECLINE. ASTERISK-25677 #close Change-Id: I8ffa226fda4365ee7068ac1f464473f1a4ebbb29 2016-01-09 13:28 +0000 [a5406b1f9e] Corey Farrell * res_crypto: Perform cleanup at shutdown. This change causes res_crypto to unregister CLI at shutdown while still preventing the module from being unloaded. ASTERISK-25673 #close Change-Id: Ie5d57338dc2752abfc0dd05d0eec86413f2304fc 2016-01-06 19:10 +0000 [cf8e7a580b] Richard Mudgett * res_pjsip: Create human friendly serializer names. PJSIP name formats: pjsip/aor/- -- registrar thread pool serializer pjsip/default- -- default thread pool serializer pjsip/messaging -- messaging thread pool serializer pjsip/outreg/- -- outbound registration thread pool serializer pjsip/pubsub/- -- pubsub thread pool serializer pjsip/refer/- -- REFER thread pool serializer pjsip/session/- -- session thread pool serializer pjsip/websocket- -- websocket thread pool serializer Change-Id: Iff9df8da3ddae1132cb2ef65f64df0c465c5e084 2016-01-06 19:09 +0000 [4276f185f0] Richard Mudgett * Sorcery: Create human friendly serializer names. Sorcery name formats: sorcery/- -- Sorcery thread pool serializer Change-Id: Idc2e5d3dbab15c825b97c38c028319a0d2315c47 2016-01-06 19:09 +0000 [f02ac1b7f9] Richard Mudgett * Stasis: Create human friendly taskprocessor/serializer names. Stasis name formats: subm:- -- Stasis subscription mailbox task processor subp:- -- Stasis subscription thread pool serializer Change-Id: Id19234b306e3594530bb040bc95d977f18ac7bfd 2016-01-07 16:15 +0000 [ec1f1c6742] Richard Mudgett * taskprocessor.c: New API for human friendly taskprocessor names. * Add new API call to get a sequence number for use in human friendly taskprocessor names. * Add new API call to create a taskprocessor name in a given buffer and append a sequence number. Change-Id: Iac458f05b45232315ed64aa31b1df05b875537a9 2016-01-06 17:19 +0000 [d8bc3e0c8b] Richard Mudgett * taskprocessor.c: Fix CLI "core show taskprocessors" output format. Update the CLI "core show taskprocessors" output format to not be distorted because UUID names are longer than previously used taskprocessor names. Change-Id: I1a5c82ce3e8f765a0627796aba87f8f7be077601 2016-01-07 21:07 +0000 [2c4b7502de] Richard Mudgett * taskprocessor.c: Fix CLI "core show taskprocessors" unref. Change-Id: I1d9f4e532caa6dfabe034745dd16d06134efdce5 2016-01-07 20:44 +0000 [3b33ac7a46] Richard Mudgett * taskprocessor.c: Sort CLI "core show taskprocessors" output. Change-Id: I71e7bf57c7b908c8b8c71f1816348ed7c5a5d51e 2016-01-06 19:00 +0000 [0fc32c4dd3] Richard Mudgett * ccss.c: Replace space in taskprocessor name. The CLI "core ping taskprocessor" command does not work very well with taskprocessor names that have spaces in them. You have to put quotes around the name so using tab completion becomes awkward. Change-Id: I29e806dd0a8a0256f4e2e0a7ab88c9e19ab0eda0 2016-01-05 16:54 +0000 [0e0c24ad78] Richard Mudgett * taskprocessor.c: Add CLI "core ping taskprocessor" missing unlock. Change-Id: I78247e0faf978bf850b5ba4e9f4933ab3c59d17b 2016-01-07 03:33 +0000 [0f79c8839b] Diederik de Groot * main: Use ast_strdup instead of strdup Fix compile error in main/utils.c because strdup was used in dummy_start Change-Id: Id61a6cf4f3cbf235450441e10e7da101a6335793 2016-01-07 03:21 +0000 [4285dee778] Diederik de Groot * include/asterisk/time.h: Renamed global declaration:tv Renamed global declaration:tv to dummy_tv_var_for_types, which would oltherwise cause 'shadow' warnings when 'tv' was declared as a local variable elsewhere. Added comment to note that dummy_tv_var_for_types is never really exported and only used as a place holder. ASTERISK-25627 #close Change-Id: I9a6e17995006584f3627efe8988e3f8aa0f5dc28 2016-01-07 15:37 +0000 [96094feab6] Mark Michelson * PJSIP: Prevent deadlock due to dialog/transaction lock inversion. A deadlock was observed where the monitor thread was stuck, therefore resulting in no incoming SIP traffic being processed. The problem occurred when two 200 OK responses arrived in response to a terminating NOTIFY request sent from Asterisk. The first 200 OK was dispatched to a threadpool worker, who locked the corresponding transaction. The second 200 OK arrived, resulting in the monitor thread locking the dialog. At this point, the two threads are at odds, because the monitor thread attempts to lock the transaction, and the threadpool thread loops attempting to try to lock the dialog. In this case, the fix is to not have the monitor thread attempt to hold both the dialog and transaction locks at the same time. Instead, we release the dialog lock before attempting to lock the transaction. There have also been some debug messages added to the process in an attempt to make it more clear what is going on in the process. ASTERISK-25668 #close Reported by Mark Michelson Change-Id: I4db0705f1403737b4360e33a8e6276805d086d4a 2016-01-07 09:39 +0000 [52e9de0016] Corey Farrell * ast_format_cap_append_by_type: Resolve codec reference leak. This resolves a reference leak caused by ASTERISK-25535. The pointer returned by ast_format_get_codec is saved so it can be released. ASTERISK-25664 #close Change-Id: If9941b1bf4320b2c59056546d6bce9422726d1ec 2016-01-04 04:26 +0000 [86eae38d7e] Aaron An * cel/cel_radius: Fix wrong pointer. The macro ADD_VENDOR_CODE defined in the cel_radius.c should use the parameter y not the address of y. I capture the radius UDP packet via tcpdump, and the AV pairs are not correct, then i review the source code and compare it with cdr/cdr_radius.c. Fix it and it works. ASTERISK-25647 #close Reported by: Aaron An Tested by: Aaron An Change-Id: I72889bccd8fde120d47aa659edc0e7e6d4d019f0 2016-01-05 14:52 +0000 [881dc862e0] gtjoseph * asterisk.h: Add ASTERISK_REGISTER_FILE macro The 11/13 branches and master use 2 different file version macros. 11/13 uses ASTERISK_FILE_VERSION but master uses ASTERISK_REGISTER_FILE. This means a new file added to 11/13 can't just be cherry-picked to master because the macro has to be changed. To make cherry-picking possible, ASTERISK_REGISTER_FILE was added to asterisk.h as a simple alias for ASTERISK_FILE_VERSION(__FILE__, NULL) The "$Revision$" tag doesn't do anything since Asterisk moved to git so just passing NULL as the verison works fine. asterisk.h was also annotated to deprecate ASTERISK_FILE_VERSION and suggest using ASTERISK_REGISTER_FILE for all new files. Finally, 2 recent file additions, pbx_builtins.c and pbx_functions.c, were modified to use the new macro to make sure it actually worked. 'core show file version' showed the correct output. Change-Id: I5867ed898818d26ee49bb6e5c7d4c1a45d4789a5 2016-01-05 11:06 +0000 [d228b62fd4] gtjoseph * stasis_cache_pattern: Backport to 13 Somehow stasis_cache_pattern got out of sync between 13 and master and it was causing duplicate channel message issues in 13 when related to a specific endpoint. I.E. from statsd, 'endpoints.PJSIP.1174.channels 0|g' was being emitted twice. Backporting stasis_cache_pattern from master to 13 solved the issue and running the unit and testsuite tests confirmed that no new ones were created. ASTERISK-25317 #close Change-Id: Ia8707462f62d15eed14541c37f332a7bbbceb548 2016-01-04 20:23 +0000 [e462f0063f] Corey Farrell * main/pbx: Move hangup handler routines to pbx_hangup_handler.c. This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves hangup handler management functions to their own source. Change-Id: Ib25a75aa57fc7d5c4294479e5cc46775912fb104 2016-01-04 19:46 +0000 [ab191d124c] Corey Farrell * main/pbx: Move dialplan application management routines to pbx_app.c. This is the sixth patch in a series meant to reduce the bulk of pbx.c. This moves dialplan application management functions to their own source. Change-Id: I444c10fb90a3cdf9f3047605d6a8aad49c22c44c 2016-01-04 18:20 +0000 [09a9b93896] Corey Farrell * main/pbx: Move switch routines to pbx_switch.c. This is the fifth patch in a series meant to reduce the bulk of pbx.c. This moves ast_switch functions to their own source. Change-Id: Ic2592a18a5c4d8a3c2dcf9786c9a6f650a8c628e 2016-01-04 18:00 +0000 [c608274a39] Corey Farrell * main/pbx: Move timing routines to pbx_timing.c. This is the fourth patch in a series meant to reduce the bulk of pbx.c. This moves pbx timing functions to their own source. Change-Id: I05c45186cb11edfc901e95f6be4e6a8abf129cd6 2015-12-29 04:31 +0000 [338a8ffed6] Martin Tomec * app_queue: Add member flag "in_call" to prevent reading wrong lastcall time Member lastcall time is updated later than member status. There was chance to check wrapuptime for available member with wrong (old) lastcall time. New boolean flag "in_call" is set to true right before connecting call, and reset to false after update of lastcall time. Members with "in_call" set to true are treat as unavailable. ASTERISK-19820 #close Change-Id: I1923230cf9859ee51563a8ed420a0628b4d2e500 2015-12-28 17:23 +0000 [e13719bff1] Rodrigo Ramírez Norambuena * app_queue: Added reason pause of member In app_queue added value Paused Reason on QueueMemberStatus when a member on queue is paused and the reason was set. ASTERISK-25480 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: Ia5db503482f50764c15e2020196c785f59d4a68e 2015-12-30 10:49 +0000 [4ec85a9f07] gtjoseph * voicemail: Move app_voicemail / res_mwi_external conflict to runtime The menuselect conflict between app_voicemail and res_mwi_external makes it hard to package 1 version of Asterisk. There no actual build dependencies between the 2 so moving this check to runtime seems like a better solution. The ast_vm_register and ast_vm_greeter_register functions in app.c were modified to return AST_MODULE_LOAD_DECLINE instead of -1 if there is already a voicemail module registered. The modules' load_module functions were then modified to return DECLINE instead of -1 to the loader. Since -1 is interpreted by the loader as AST_MODULE_LOAD_FAILURE, the modules were incorrectly causing Asterisk to stop so this needed to be cleaned up anyway. Now you can build both and use modules.conf to decide which voicemail implementation to load. The default menuselect options still build app_voicemail and not res_mwi_external but if both ARE built, res_mwi_external will load first and become the voicemail provider unless modules.conf rules prevent it. This is noted in CHANGES. Change-Id: I7d98d4e8a3b87b8df9e51c2608f0da6ddfb89247 2016-01-04 16:22 +0000 [7fdcfd7724] Corey Farrell * main/pbx: Move variable routines to pbx_variables.c. This is the third patch in a series meant to reduce the bulk of pbx.c. This moves channel and global variable routines to their own source. Change-Id: Ibe8fb4647db11598591d443a99e3f99200a56bc6 2015-12-04 17:22 +0000 [80a8b2a4cd] Richard Mudgett * app_dial: Immediately exit dial if the caller is already hung up. If a caller hangs up before dial is executed within an AGI then the AGI has likely eaten all queued frames before executing the dial in DeadAGI mode. With the caller hung up and no pending frames from the caller's read queue, dial would not know that the call has hung up until a called channel answers. It is rather annoying to whoever just answered the non-existent call. Dial should not continue execution in DeadAGI mode, hangup handlers, or the h exten. * Added a check early in dial to abort dialing if the caller has hungup. ASTERISK-25307 #close Reported by: David Cunningham Change-Id: Icd1bc0764726ef8c809f76743ca008d0f102f418 2016-01-02 10:26 +0000 [1087b0c6ed] Matt Jordan * main/cdr: Allow setting properties on a finalized CDR if it is the last one Prior to this patch, we explicitly disallowed setting any properties on a finalized CDR. This seemed like a good idea at the time; in practice, it was more restrictive. There are weird and strange scenarios where setting a property on a finalized CDR is definitely wrong. For example, we may Fork a CDR, finalizing the previous one, then change a property. In said case, the old CDR is supposed to now be 'immutable' (so to speak), and should not be updated. From the perspective of the code, a forked CDR that is finalized is just finalized. Hence why we decided these should not be updated. In practice, it is much more common to want to set a property on a CDR in the h extension or in a hangup handler. Disallowing a common scenario to make an esoteric behaviour work isn't good. This patch fixes this by allowing callers to set a property IF we are the last CDR in the chain. This preserves the finalized CDR if it was forked, while allowing the more common case to function. ASTERISK-25458 #close Change-Id: Icf3553c607b9f561152a41e6d8381d594ccdf4b9 2016-01-02 10:23 +0000 [1f23e65b89] Matt Jordan * main/cdr: Set the end time on a CDR if endbeforehexten is Yes Prior to this patch, the CDR engine attempted to set the end time on a CDR that was executing hangup logic and with endbeforehexten set to Yes by calling a function that inspects the properties on the Party A snapshot to determine if we are ready to set the end time. That always failed. This is because a Party A snapshot is not updated for CDRs that are executing hangup logic with endbeforehexten=Yes. Instead of calling a function that looks at the Party A snapshot, we just simply set the end time on the CDR. This is safe to call multiple times, and is safe to call at this point as we know that (a) we are executing hangup logic, and (b) we are supposed to set the end time at this point. ASTERISK-25458 Change-Id: I0c27b493861f9c13c43addbbb21257f79047a3b3 2015-12-30 20:51 +0000 [2ffade4574] Corey Farrell * main/pbx: Move custom function routines to pbx_functions.c. This is the second patch in a series meant to reduce the bulk of pbx.c. This moves custom function management routines to their own source. Change-Id: I34a6190282f781cdbbd3ce9d3adeac3c3805e177 2015-12-28 19:18 +0000 [20b8474f20] gtjoseph * main/pbx: Move pbx_builtin dialplan applications to pbx_builtins.c We joked about splitting pbx.c into multiple files but this first step was fairly easy. All of the pbx_builtin dialplan applications have been moved into pbx_builtins.c and a new pbx_private.h file was added. load_pbx_builtins() is called by asterisk.c just after load_pbx(). A few functions were renamed and are cross-exposed between the 2 source files. Change-Id: I87066be3dbf7f5822942ac1449d98cc43fc7561a 2015-12-24 20:26 +0000 [e4a566918a] Matt Jordan * tests/test_stasis_endpoints: Remove expected duplicate events The cache_clear test was written to expect duplicate Stasis messages sent from the technology endpoint to the all caching topic. This patch fixes the test to no longer expect these duplicate messages. ASTERISK-25137 Change-Id: I58075d70d6cdf42e792e0fb63ba624720bfce981 2015-12-28 14:02 +0000 [a280400758] Joshua Colp * test_time: Provide a timeout when waiting. The test_timezone_watch unit test is written to expect a condition to be signaled when the inotify daemon thread runs. There exists a small window where the test_timezone_watch thread can signal the inotify daemon thread while it is not reading on the underlying file descriptor. If this occurs the test_timezone_watch thread will wait indefinitely for a signal that will never arrive. This change adds a timeout to the condition so it will return regardless after a period of time. Change-Id: Ifed981879df6de3d93acd3ee0a70f92546517390 2015-05-27 13:22 +0000 [3a1c4885be] gtjoseph * endpoint/stasis: Eliminate duplicate events on endpoint status change When an endpoint is created, its messages are forwarded to both the tech endpoint topic and the all endpoints topic. This is done so that various parties interested in endpoint messages can subscribe to just the tech endpoint and receive all messages associated with that particular technology, as opposed to subscribing to the all endpoints topic. Unfortunately, when the tech endpoint is created, it also forwards all of its messages to the all topic. This results in duplicate messages whenever an endpoint publishes its messages. This patch resolves the duplicate message issue by creating a new function for Stasis caching topics, stasis_cp_sink_create. In most respects, this acts as a normal caching topic, save that it no longer forwards messages it receives to the all endpoints topic. This allows it to act as an aggregation "sink", while preserving the necessary caching behaviour. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: Ie47784adfb973ab0063e59fc18f390d7dd26d17b 2015-12-24 22:19 +0000 [136c537695] Dade Brandon * res_http_websocket.c: prevent avoidable disconnections caused by write errors Updated ast_websocket_write to encode the entire frame in to one write operation, to ensure that we don't end up with a situation where the websocket header has been sent, while the body can not be written. Previous to August's patch in commit b9bd3c14, certain network conditions could cause the header to be written, and then the sub-sequent body to fail - which would cause the next successful write to contain a new header, and a new body (resulting in the peer receiving two headers - the second of which would be read as part of the body for the first header). This was patched to have both write operations individually fail by closing the websocket. In a case available to the submitter of this patch, the same body which would consistently fail to write, would succeed if written at the same time as the header. This update merges the two operations in to one, adds debug messages indicating the reason for a websocket connection being closed during a write operation, and clarifies some variable names for code legibility. Change-Id: I4db7a586af1c7a57184c31d3d55bf146f1a40598 2015-12-27 22:38 +0000 [f2efbb5d75] Corey Farrell * Remove res_jabber file that was left behind. Change-Id: I9d88fac0394d5bbaff0900a2ee911c4e4478846b 2015-12-13 13:09 +0000 [dde7f3c1c4] Matt Jordan * res_pjsip_history: Add a module that provides PJSIP history for debugging This patch adds a new module, res_pjsip_history, that provides a slightly better way of debugging SIP message traffic on a busy Asterisk system. The existing mechanisms all rely on passively dumping a SIP message to the CLI. While this is perfectly fine for logging purposes and well controlled environments, on many installations, the amount of SIP messages Asterisk receives will quickly swamp the CLI. This makes it difficult to view/capture those messages that you want to diagnose in real time. This patch provides another way of handling this. When enabled, the module will store SIP message traffic in memory. This traffic can then be queried at leisure. In order to make the querying useful, a CLI command has been implemented, 'pjsip show history', that supports a basic expression syntax similar to SQL or other query languages. A small number of useful fields have been added in this initial patch; additional fields can easily be added in later improvements. Those fields are: - number: The entry index in the history - timestamp: The time the message was recieved - addr: The source/destination address of the message - sip.msg.request.method: The request method - sip.msg.call-id: The Call-ID header Note - this is a resurrection of the module initially proposed on Review Board here: https://reviewboard.asterisk.org/r/4053/ Change-Id: I39bd74ce998e99ad5ebc0aab3e84df3a150f8e36 2015-12-25 09:56 +0000 [be050f2638] Dade Brandon * chan_sip.c: fix websocket_write_timeout default value websocket_write_timeout was not being set to its default value during sip config reload, which meant that prior to this commit, 1) the default value of 100 was not used, unless an invalid value (or 1) was specified in sip.conf for websocket_write_timeout, and 2) if the websocket_write_timeout directive was removed from sip.conf without a full restart of asterisk, then the previous value would continue to be used indefinitely. This essentially lead to a 0ms write timeout (the first write attempt in ast_careful_fwrite must have succeeded) in websocket write requests from chan_sip, unless websocket_write_timeout was explicitely set in sip.conf. Changes to websocket_write_timeout still only apply to new websocket sessions, after the sip reload -- timeouts on existing sessions are not adjusted during sip reload. Change-Id: Ibed3816ed29cc354af6564c5ab3e75eab72cb953 2015-12-23 17:40 +0000 [b3024cad10] Richard Mudgett * bridge_basic.c: Fix GOTO_ON_BLINDXFR Use of GOTO_ON_BLINDXFR would not work at all. The target location would never be executed by the transferring channel. * Made feature_blind_transfer() call ast_bridge_set_after_go_on() with valid context, exten, and priority parameters from the transferring channel. * Renamed some feature_blind_transfer() local variables for clarity. ASTERISK-25641 #close Reported by Dmitry Melekhov Change-Id: I19bead9ffdc4aee8d58c654ca05a198da1e4b7ac 2015-12-24 12:19 +0000 [0a9941de9d] Matt Jordan * res/res_pjsip_location: Delete contact_status object when contact is deleted In 450579e908, a change was made that removed the deletion of the 'contact_status' object when a 'contact' object is deleted in sorcery. This unfortunately means that the 'contact_status' object persists, even when something has explicitly removed a contact. The result is that the state of the contact will not be regenerated if that contact is re-created, and the stale state will be reported/used for that contact. It also results in no ContactStatusChanged events being generated for either ARI or AMI. This patch restores the deletion logic that was removed. Doing so now results in the expected events being generated again. Change-Id: I28789a112e845072308b5b34522690e3faf58f07 2015-12-24 10:18 +0000 [1e24a0ca8a] Kevin Harwell * res_rtp_asterisk: rtp->ice check not wrapped in HAVE_PJPROJECT ifdef Change-Id: I19b49112e1b630bd04e859f14ccf96f8ebd6b151 2015-12-20 21:33 +0000 [1d3d20dd68] Dade Brandon * app_amd: Correct documentation to reflect functionality Update documentation to reflect that maximum_number_of_words has functionality inconsistent with the variable name (and inconsistent with prior documentation.) Update documentation for silence_threshold, which previously implied that it was measuring time, rather than noise averages in the sample. Update the comments in amd.conf.sample. ASTERISK-25639 #close Change-Id: I4b1451e5dc9cb3cb06d59b6ab872f5275ba79093 2015-12-17 19:05 +0000 [965a0eee46] Dade Brandon * res_rtp_asterisk: Resolve further timing issues with DTLS negotiation Resolves an edge case dtls negotiation delay for certain networks which somehow manage to drop the rtcp side's packet when these are both sent ast_rtp_remote_address_set, causing it to have to time-out and restart the handshake. Move dtls pending bio flush in to it's own function, and call it from ast_rtp_on_ice_complete, when we're rtp->ice, rather than when ast_rtp_remote_address_set. Keep the existing flush from the recent change to res_rtp_remote_address_set if ice is not being used. ASTERISK-25614 #close Reported-by: XenCALL Tested by: XenCALL Change-Id: Ie2caedbdee1783159f375589b6fd3845c8577ba5 2015-12-18 09:54 +0000 [ae428d8460] Carlos Oliva * app_queue: update RT members when the 1st call joins a queue with no agents If a call enters on a queue and the members on that queue are updated in realtime (ex: using mysql inserting a new agent) the queue members are never refreshed and the call will stay in the queue until other event occurs. This happens only if this is the first call of the queue and there is no agents servicing. This patch prevent this issue, ensuring realtime members are updated if there is one call in the queue and no available agents ASTERISK-25442 #close Change-Id: If1e036d013a5c1d8b0bf60d71d48fe98694a8682 2015-12-05 10:01 +0000 [59d5bb0613] Joshua Colp * res_sorcery_memory_cache: Add support for a full backend cache. This change introduces the configuration option 'full_backend_cache' which changes the cache to be a full mirror of the backend instead of a per-object cache. This allows all sorcery retrieval operations to be carried out against it and is useful for object types which are used in a "retrieve all" or "retrieve some" pattern. ASTERISK-25625 #close Change-Id: Ie2993487e9c19de563413ad5561c7403b48caab5 2015-12-17 10:25 +0000 [0cefcabd58] Joshua Colp * rtp_engine: Ignore empty filenames in DTLS configuration. When applying an empty DTLS configuration the filenames in the configuration will be empty. This is actually valid to do and each filename should simply be ignored. Change-Id: Ib761dc235638a3fb701df337952f831fc3e69539 2015-12-17 08:10 +0000 [158a0a5422] Joshua Colp * chan_sip: Enable WebSocket support by default. Per the documentation the WebSocket support in chan_sip is supposed to be enabled by default but is not. This change corrects that. Change-Id: Icb02bbcad47b11a795c14ce20a9bf29649a54423 2015-12-14 12:04 +0000 [a9d6fc571d] Joshua Colp * json: Audit ast_json_* usage for thread safety. The JSON library Asterisk uses, jansson, is not thread safe for us in a few ways. To help with this wrappers for JSON object reference count increasing and decreasing were added which use a global lock to ensure they don't clobber over each other. This does not extend to reference count manipulation within the jansson library itself. This means you can't safely use the object borrowing specifier (O) in ast_json_pack and you can't share JSON instances between objects. This change removes uses of the O specifier and replaces them with the o specifier and an explicit ast_json_ref. Some cases of instance sharing have also been removed. ASTERISK-25601 #close Change-Id: I06550d8b0cc1bfeb56cab580a4e608ae4f1ec7d1 2015-12-16 11:28 +0000 [53bd5a539a] Mark Michelson * Alembic: Increase column size of PJSIP AOR "contact". When running the PJSIP AMI "show_endpoint" test with automatic conversion to realtime, the test would fail. This was because the AOR "contact" column was sized at 40, and the configured contact was larger than that. This commit increases the size of the contact column to 255 characters. Change-Id: Ia65bc7fd37699b7c0eaef9629a1a31eab9a24ba1 2015-12-16 11:25 +0000 [da17dc4d75] Mark Michelson * Alembic: Add PJSIP global keep_alive_interval. The keep_alive_interval option was added about a year ago, but no alembic revision was created to add the appropriate column to the database. This commit fixes the problem and adds the column. This was discovered by running the testsuite with automatic conversion to realtime enabled. Change-Id: If3ef92a7c4f4844d08f8aae170d2178aec5c4c1a 2015-12-14 13:53 +0000 [24ae124e4f] server-pandora * res_rtp_asterisk.c: Fix DTLS negotiation delays. - Trigger pending DTLS packets to send out, once the RTP instance's remote address is set. - Avoids locking the DTLS structure unnecessarily by only doing this if DTLS is passive. - Add DTLS locks around the structurally sensitive calls in the SSL portion of __rtp_recvfrom, since dtls_srtp_check_pending does not lock inside of itself, and we're dealing with the SSL BIO in at least two threads. WebRTC channels may receive a DTLS handshake before ast_rtp_remote_address_set is called, which causes there to be a pending response to send out. Previous to 1ad827, this was handled by calling dtls_srtp_check_pending on receipt of any RTP packet - a STUN or RTP packet could trigger the pending handshake response. Since that was rightfully removed, whenever the DTLS handshake is received before the remote address is set, we would have to wait until another SSL packet arrives. As of Chrome M47's optimizations to their handshake process, WebRTC conversations between Chrome M47+ and Asterisk, where Asterisk is passive, experience a 1 second delay without this patch, because the SSL handshake is received before ICE negotation stores the remote_address, and the next SSL packet isn't received until after a 1 second timeout in Chrome, which causes a new handshake request. ASTERISK-25614 #close Change-Id: I547f1be7e302dbf71f6553dd8cbc0657b1d0b908 2015-12-14 15:25 +0000 [36097a185d] Richard Mudgett * Fix sscanf() format string type mismatch. ASTERISK-25615 Reported by: George Joseph Change-Id: Ieff35307254ca193f3d473cff2e396ca57c7ce0b 2015-12-13 13:13 +0000 [94f9927784] Matt Jordan * main/utils: Don't emit an ERROR message if the read end of a pipe closes An ERROR or WARNING message should generally indicate that something has gone wrong in Asterisk. In the case of writing to a file descriptor, Asterisk is not in control of when the far end closes its reading on a file descriptor. If the far end does close the file descriptor in an unclean fashion, this isn't a bug or error in Asterisk, particularly when the situation can be gracefully handled in Asterisk. Currently, when this happens, a user would see the following somewhat cryptic ERROR message: "utils.c: write() returned error: Broken pipe" There's a few problems with this: (1) It doesn't provide any context, other than 'something broke a pipe' (2) As noted, it isn't actually an error in Asterisk (3) It can get rather spammy if the thing breaking the pipe occurs often, such as a FastAGI server (4) Spammy ERROR messages make Asterisk appear to be having issues, or can even mask legitimate issues This patch changes ast_carefulwrite to only log an ERROR if we actually had one that was reasonably under our control. For debugging purposes, we still emit a debug message if we detect that the far side has stopped reading. Change-Id: Ia503bb1efcec685fa6f3017bedf98061f8e1b566 2015-12-12 11:08 +0000 [5b867fa904] gtjoseph * pjsip/config_transport: Check pjproject version at runtime for async ops pjproject < 2.5.0 will segfault on a tls transport if async_operations is greater than 1. A runtime version check has been added to throw an error if the version is < 2.5.0 and async_operations > 1. To assist in the check, a new api "ast_compare_versions" was added to utils which compares 2 major.minor.patch.extra version strings. ASTERISK-25615 #close Change-Id: I8e88bb49cbcfbca88d9de705496d6f6a8c938a98 Reported-by: George Joseph Tested-by: George Joseph 2015-12-10 11:44 +0000 [14b41115e3] Jonathan Rose * chan_sip: Add TCP/TLS keepalive to TCP/TLS server Adds the TCP Keep Alive option to TCP and TLS server sockets. Previously this option was only being set on session sockets. http://www.tldp.org/HOWTO/html_single/TCP-Keepalive-HOWTO/ According to the link above, the SO_KEEPALIVE option is useful for knowing when a TCP connected endpoint has severed communication without indicating it or has become unreachable for some reason. Without this patch, keep alive is not set on the socket listening for incoming TCP sessions and in Komatsu's report this resulted in the thread listening for TCP becoming stuck in a waiting state. ASTERISK-25364 #close Reported by: Hiroaki Komatsu Change-Id: I7ed7bcfa982b367dc64b4b73fbd962da49b9af36 2015-12-08 13:04 +0000 [fe8011cc50] sungtae kim * AMI: Fixed OriginateResponse message When the asterisk sending OriginateResponse message, it doesn't set the "Uniqueid". And it didn't support correct response message for Application originate. ASTERISK-25624 #close Change-Id: I26f54f677ccfb0b7cfd4967a844a1657fd69b74d 2015-12-09 09:48 +0000 [cd119ed4a2] Tyler Cambron * res_chan_stats: Fix bug to send correct statistics to StatsD Fixed a bug that originally would show a negative number of active calls occuring in Asterisk. A gauge is persistent so incrementing and decrementing it results in a more consistent performance. Also changed to the call to StatsD to use ast_statsd_log_string() so that a "+" could be sent to StatsD. ASTERISK-25619 #close Change-Id: Iaaeff5c4c6a46535366b4d16ea0ed0ee75ab2ee7 2015-12-07 13:07 +0000 [ddf4dddf4f] Corey Farrell * app_meetme: Set default value for audio_buffers. The default value was never set for audio_buffers, causing bad audio quality. This ensures the default is always set. ASTERISK-25569 #close Change-Id: I2d2ee3e644120b0f9f6ea6ab9286d7d590942a44 2015-12-08 01:57 +0000 [142d4fefb8] Filip Jenicek * chan_sip: Check sip_pvt pointer in ast_channel_get_t38_state(c) Asterisk may crash when calling ast_channel_get_t38_state(c) on a locked channel which is being hung up. ASTERISK-25609 #close Change-Id: Ifaa707c04b865a290ffab719bd2e5c48ff667c7b 2015-12-08 17:49 +0000 [21962dad93] gtjoseph * res_pjsip: Add existence and readablity checks for tls related files Both transport and endpoint now check for the existence and readability of tls certificate and key files before passing them on to pjproject. This will cause the object to not load rather than waiting for pjproject to discover that there's a problem when a session is attempted. NOTE: chan_sip also uses ast_rtp_dtls_cfg_parse but it's located in build_peer which is gigantic and I didn't want to disturb it. Error messages will emit but it won't interrupt chan_sip loading. ASTERISK-25618 #close Change-Id: Ie43f2c1d653ac1fda6a6f6faecb7c2ebadaf47c9 Reported-by: George Joseph Tested-by: George Joseph 2015-12-02 12:42 +0000 [28d9243079] Eugene Voityuk * chan_sip.c: Start ICE negotiation when response is sent or received. The current logic for ICE negotiation starts it when receiving an SDP with ICE candidates. This is incorrect as ICE negotiation can only start when each call party have at least one pair of local and remote candidate. Starting ICE negotiation early would result in negotiation failure and ultimately no audio. This change makes it so ICE negotiation is only started when a response with SDP is received or when a response with SDP is sent. ASTERISK-24146 Change-Id: I55a632bde9e9827871b09141d82747e08379a8ca 2015-12-08 11:03 +0000 [e03582a1c2] gtjoseph * res_pjsip/config_transport: Prevent async_operations > 1 when protocol = tls See ASTERISK-25615. If the transport protocol is tls and async_operations > 1, pjproject will segfault if more than one operation is attempted on the same socket. Until this is fixed upstream, a check has been added to throw an error if a tls transport config has async_operations set to > 1. ASTERISK-25615 Change-Id: I76b9a5b2a5a0054fe71ca5851e635f2dca7685a6 Reported-by: George Joseph Tested-by: George Joseph 2015-12-08 08:39 +0000 [876600ce6e] Alexander Traud * codec_resample: Increase buffer for Opus Codec with FEC. ASTERISK-25599 #close Change-Id: Idbd187f711b2ec63dda949ca0f79aa0c1a0a0b6e 2015-12-08 03:46 +0000 [69e3d40ad7] Alexander Traud * translate: Avoid a warning message when doing FEC within Opus Codec. ASTERISK-25616 #close Change-Id: Ibe729aaf2e6e25506cff247cec5149ec1e589319 2015-12-04 15:36 +0000 [2b992014dc] Richard Mudgett * chan_sip: Fix crash involving the bogus peer during sip reload. A crash happens sometimes when performing a CLI "sip reload". The bogus peer gets refreshed while it is in use by a new call which can cause the crash. * Protected the global bogus peer object with an ao2 global object container. ASTERISK-25610 #close Change-Id: I5b528c742195681abcf713c6e1011ea65354eeed 2015-12-06 16:32 +0000 [529535f0c2] Matt Jordan * Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" This reverts commit 6614babea27fbafbe11820ea03737dd5c4f9ecec. Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks in core_unreal/chan_local. Local channels attempt to reach across both their peer and the peer's bridge to inspect T.38 state. Given the propensity of Local channel chains, managing the locking situation in such a scenario is practically infeasible. Change-Id: Ic687397ffea08dfb899345a443bd990ec3d0416a 2015-12-04 16:23 +0000 [450579e908] gtjoseph * res_pjsip/contacts/statsd: Make contact lifecycle events more consistent It will never be perfect or even pretty, mostly because of the differences between static and dynamic contacts. Created: Can't use the contact or contact_status alloc functions because the objects come and go regardless of the actual state. Can't use the contact_apply_handler, ast_sip_location_add_contact or a sorcery created handler because they only get called for dynamic contacts. Similarly, permanent_uri_handler only gets called for static contacts. So, Matt had it right. :) ast_res_pjsip_find_or_create_contact_status is the only place it can go and not have duplicated code. Both permanent_uri_handler and contact_apply_handler call find_or_create. Removed: Can't use the destructors for the same reason as above. The only place to put this is in persistent_endpoint_contact_deleted_observer which I believe is the "correct" place but even that will handle only dynamic contacts. This doesn't called on shutdown however. There is no hook to use for static contacts that may be removed because of a config change while asterisk is in operation. I moved the cleanup of contact_status from ast_sip_location_delete_contact to the handler as well. Status Change and RTT: Although they worked fine where they were (in update_contact_status) I moved them to persistent_endpoint_contact_status_observer to make it more consistent with removed. There was logic there already to detect a state change. Finally, fixed a nit in permanent_uri_handler rmudgett reported eralier. ASTERISK-25608 #close Change-Id: I4b56e7dfc3be3baaaf6f1eac5b2068a0b79e357d Reported-by: George Joseph Tested-by: George Joseph 2015-11-21 06:02 +0000 [5a18193dc0] Alexander Traud * res_format_attr_vp8: In SDP, forward max-fr and max-fs for video-codec VP8. ASTERISK-25584 #close Change-Id: Iae00071b4ff1ae76f24995aeac4d00284fd14f91 2015-11-21 05:21 +0000 [3e2178c05e] Alexander Traud * res_format_attr_opus: Update to latest RFC 7587. Beside that, the format-attribute module sends only non-default values in the line fmtp, now. This avoids unnecessary overhead in SDP messages. Furthermore, previously the parameter stereo was not parsed when being the first parameter. ASTERISK-25583 #close Change-Id: Iae85ba3e5960bfd5d51cf65bcffad00dd4875a73 2015-12-02 14:11 +0000 [072d94183c] Jonathan Rose * Fix crash in audiohook translate to slin This patch fixes a crash which would occur when an audiohook was applied to a channel using an audio codec that could not be translated to signed linear (such as when using pass-through codecs like OPUS or when the codec translator module for the format in use is not loaded). ASTERISK-25498 #close Reported by: Ben Langfeld Change-Id: Ib6ea7373fcc22e537cad373996136636201f4384 2015-12-03 12:07 +0000 [9184fbeb34] gtjoseph * res_pjsip: Use a MD5 hash for static Contact IDs When 90d9a70789 was merged, it mostly tested dynamic contacts created as a result of registering a PJSIP endpoint. Contacts generated in this fashion typically have a long alphanumeric string as their object identifier, which maps reasonably well for StatsD. Unfortunately, this doesn't work in the general case. StatsD treats both '.' and ':' characters as special characters. In particular, having a ':' appear in the middle of a StatsD metric will result in the metric being rejected. This causes some obvious issues with SIP URIs. The StatsD API should not be responsible for escaping the metric name passed to it. The metric is treated as a single long string, and it would be challenging to know what to escape in the string passed to the function. Likewise, we don't want to escape the metric in PJSIP, as that involves overhead that is wasted when either res_statsd isn't loaded or enabled. This patch takes an alternative approach. The Contact ID has been changed to be "aor@@uri_hash" instead of "aor@@uri". This (a) won't contain any of the aforementioned special characters, (b) can be done on Contact creation, which has minimal impact on run-time performance, and (c) also conforms to an earlier commit that changed the ID for dynamic contacts. The downside of this is that StatsD users will have to map SHA1 hashes back to the Contacts that are emitting the statistics. To that end, the CLI commands have been updated to include the first 10 characters of the MD5 hash, which should be enough to match what is shown in Graphite (or some other StatsD backend). ASTERISK-25595 #close Change-Id: Ic674a3307280365b4a45864a3571c295b48a01e2 Reported-by: Matt Jordan Tested-by: George Joseph 2015-11-30 22:19 +0000 [ed9134282e] gtjoseph * res_pjsip: Update logging to show contact->uri in messages An earlier commit changed the id of dynamic contacts to contain a hash instead of the uri. This patch updates status change logging to show the aor/uri instead of the id. This required adding the aor id to contact and contact_status and adding uri to contact_status. The aor id gets added to contact and contact_status in their allocators and the uri gets added to contact_status in pjsip_options when the contact_status is created or updated. ASTERISK-25598 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I56cbec1d2ddbe8461367dd8b6da8a6f47f6fe511 2015-12-01 16:11 +0000 [eadad24b59] Jonathan Rose * Unset BRIDGEPEER when leaving a bridge Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da 2015-11-30 14:22 +0000 [bb0b60619d] Richard Mudgett * res_sorcery_memory_cache.c: Fix off nominal ref leak. Change-Id: If83d63cf11cbc6df9b15251848b01feb570ade49 2015-11-30 16:42 +0000 [e7c88e11aa] Richard Mudgett * sched.c: Make not return a sched id of 0. According to the API doxygen a sched ID of 0 is valid. Unfortunately, 0 was never returned historically and several users incorrectly coded usage of the returned sched ID assuming that 0 was invalid. ASTERISK-25476 Change-Id: Ib19c7ebb44ec9fd393ef6646dea806d4f34e3a20 2015-11-25 12:23 +0000 [4aed349a7b] Richard Mudgett * Audit improper usage of scheduler exposed by 5c713fdf18f. (v13 additions) chan_sip.c: * Initialize mwi subscription scheduler ids earlier because of ASTOBJ to ao2 conversion. * Initialize register scheduler ids earlier because of ASTOBJ to ao2 conversion. chan_skinny.c: * Fix more scheduler usage for the valid 0 id value. ASTERISK-25476 Change-Id: If9f0e5d99638b2f9d102d1ebc9c5a14b2d706e95 2015-11-24 12:44 +0000 [6d9156d10f] Richard Mudgett * Audit improper usage of scheduler exposed by 5c713fdf18f. channels/chan_iax2.c: * Initialize struct chan_iax2_pvt scheduler ids earlier because of iax2_destroy_helper(). channels/chan_sip.c: channels/sip/config_parser.c: * Fix initialization of scheduler id struct members. Some off nominal paths had 0 as a scheduler id to be destroyed when it was never started. chan_skinny.c: * Fix some scheduler id comparisons that excluded the valid 0 id. channel.c: * Fix channel initialization of the video stream scheduler id. pbx_dundi.c: * Fix channel initialization of the packet retransmission scheduler id. ASTERISK-25476 Change-Id: I07a3449f728f671d326a22fcbd071f150ba2e8c8 2015-12-01 07:55 +0000 [b76c196e13] Alexander Traud * codec_resample: Increase buffer for Opus Codec. ASTERISK-25599 #close Change-Id: I1f88a88c59fb4e1e62bbdbb100c7152d48e73f10 2015-11-28 08:46 +0000 [6614babea2] Matt Jordan * bridges/bridge_t38: Add a bridging module for managing T.38 state When 4875e5ac32 was merged, it fixed several issues with a direct media bridge transitioning to handling a T.38 fax. However, it uncovered a race condition caused by the bridging core. When a channel involved in a T.38 fax leaves a bridge, the frame queued by the channel driver that should inform the far side that it is no longer in a T.38 fax may not make it across the bridge. The bridging framework is *extremely* aggressive in tearing down the bridge, and control frames that are currently in flight *may* get dropped. This patch adds a new module to the bridging framework, bridge_t38. This module maintains some notion of the T.38 state for the two channels in a bridge. When the bridge detects that it is being torn down or when one of the two channels leaves, it informs the respective channel(s) that they should stop faxing. This ensures that channels switch back to audio if they survive and are ejected out of a bridge while faxing. ASTERISK-25582 Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0 2015-11-27 07:39 +0000 [3fcf160fae] Niklas Larsson * CHANGES: Fix a typo Change-Id: Iceb3d9bb78140c376174a7bee197dfcf8ef9cda7 2015-11-25 15:26 +0000 [45efbf8503] Kevin Harwell * fastagi: record file closed after sending result The fastagi record-file testsuite test sometimes fails reporting an empty recorded file. This was happening because Asterisk was sending the agi result notification prior to actually closing the file and the data, being buffered, had not been written to the file yet when the test attempts to check the file size. This patch makes it so the record file stream is closed prior to sending the agi result notification. ASTERISK-25593 #close Change-Id: I6b2b3be3ae37f7c7b18e672c419a89b3b8513cde 2015-11-25 13:29 +0000 [b2787876d6] Walter Doekes * main: Slight refactor of main. Improve color situation. Several issues are addressed here: - main() is large, and half of it is only used if we're not rasterisk; fixed by spliting up the daemon part into a separate function. - Call ast_term_init from rasterisk as well. - Remove duplicate code reading/writing asterisk history file. - Attempt to tackle background color issues and color changes that occur. Tested by starting asterisk -c until the colors stopped changing at odd locations. ASTERISK-25585 #close Change-Id: Ib641a0964c59ef9fe6f59efa8ccb481a9580c52f 2015-11-24 13:54 +0000 [59881fbb99] David M. Lee * Fixed some typos Fixes some minor typos in the CHANGES file, plus an embarrasing typo in the StatsD API. Change-Id: I9ca4858c64a4a07d2643b81baa64baebb27a4eb7 2015-11-24 13:07 +0000 [b75f587d15] Corey Farrell * res_pjsip_notify: Fix CLI usage info The usage info for 'pjsip send notify' previously referenced the chan_sip configuration sip_notify.conf. Fix this to reference the correct configuration pjsip_notify.conf. ASTERISK-25590 #close Change-Id: I3898271a8e8a8b1db201741e790ebe2c6bf5cdea 2015-11-23 14:27 +0000 [fc45f4040d] Richard Mudgett * res_sorcery_realtime.c: Fix crash from NULL sorcery object type. If the sorcery object type is not found a NULL is returned. Unfortunately, sorcery_realtime_filter_objectset() will crash after complaining about not finding the object type and saying to expect errors. * Use ao2_cleanup() instead of ao2_ref() to prevent the crash. ASTERISK-25165 Reported by Corey Farrell Change-Id: Ic3b64453ea3058cb68d5c26d97d4fe7b8eea2e97 2015-11-20 21:08 +0000 [4875e5ac32] Matt Jordan * chan_pjsip: Handle T.38 faxes with direct media bridges When a channel is in a direct media bridge, a re-INVITE may arrive that forces Asterisk to re-negotiate the media to a T.38 fax. When this occurs, the bridge must change its technology to a simple bridge, and re-INVITE the media back to Asterisk. Generally, this logic mostly already exists in Asterisk. However, prior to this patch, there were a few bugs: (1) The T.38 framehook currently prevents a channel capable of T.38 faxes from ever entering into a direct media bridge. This applies even when the only media being passed over the channel is audio. This patch fixes this bug by having the framehook specify that it defers caring about any frame type. This allows the channels to enter into a direct media bridge, which will be broken when a re-INVITE is received. (2) When a re-INVITE is received, nothing instructed the bridging layer to re-inspect the allowed bridging technology. This now occurs when either a re-INVITE is received from a peer, or when a response is received from the far end (that is, when the T.38 state changes to either T38_PEER_REINVITE or T38_LOCAL_REINVITE). (3) chan_pjsip needs to do a small amount of work to prevent a direct media bridge from being chosen when a T.38 session is in progress. When a T.38 session supplement has a t38 datastore - which is added when we detect we should start thinking about T.38 on a channel - we now refuse a native RTP bridge. (4) When a BYE request is received, we don't terminate the T.38 session. If the other side of a T.38 fax survives the hangup (due to the 'g' flag in Dial, for example), we don't currently re-INVITE the media on the other channel back to audio. This patch now has res_pjsip_t38 intercept BYE requests and inform the far side that the T.38 session is terminated. This naturally causes the correct re-INVITEs to be sent. ASTERISK-25582 Change-Id: Iabd6aa578e633d16e6b9f342091264e4324a79eb 2015-11-20 21:07 +0000 [2b94d9a10d] Matt Jordan * res/res_pjsip_t38: Add debug statements This patch adds some debug statements to res_pjsip_t38. These statements help to determine which SDP negotiation callbacks are being executed, and, when a particular callback exits, why a callback may not have applied its logic to the local or remote SDP. Change-Id: I61b3fb9183b7ebbb5da8e9f48b59a5d9d7042d77 2015-10-22 09:44 +0000 [af288b2d96] Matt Jordan * main/cli: Use proper string methods to check existence of context/exten/app Because the context, extension, and application are stored in stringfields, checking for them being NULL doesn't work so well. This patch uses the appropriate string library call, ast_strlen_zero, to see if there is a value in the context/exten/app values. Change-Id: Ie09623bfdf35f5a8d3b23dd596647fe3c97b9a23 2015-11-18 09:43 +0000 [d27aac0a9d] Matt Jordan * res/res_endpoint_stats: Add module to emit endpoint StatsD statistics This patch adds a module that emits StatsD statistics about Asterisk endpoints. This includes: * A GUAGE statistic for endpoint states, tracking how many endpoints are in a particular state. * A GUAGE statistic for each endpoint, counting the number of channels currently associated with an endpoint. ASTERISK-25572 Change-Id: If7e1333c5aeda8d136850b30c2101c0ee1c97305 2015-11-18 10:07 +0000 [90d9a70789] Matt Jordan * res_pjsip/pjsip_options: Add StatsD statistics for PJSIP contacts This patch adds the ability to send StatsD statistics related to the state of PJSIP contacts. This includes: * A GUAGE statistic measuring the count of contacts in a particular state. This measures how many contacts are reachable, unreachable, etc. * The RTT time for each contact, if those contacts are qualified. This provides StatsD engines useful time-based data about each contact. ASTERISK-25571 Change-Id: Ib8378d73afedfc622be0643b87c542557e0b332c 2015-11-13 10:34 +0000 [75097a0955] Matt Jordan * res/res_pjsip_outbound_registration: Add registration statistics for StatsD This patch adds outbound registration statistics for StatsD. This includes the following: * A GUAGE metric for the overall count of outbound registrations. * A GUAGE metric for each state an outbound registration can be in. As the outbound registrations change state, the overall count of how many outbound registrations are in the particular state is changed. These statistics are particularly useful for systems with a large number of SIP trunks, and where measuring the change in state of the trunks is useful for monitoring. ASTERISK-25571 Change-Id: Iba6ff248f5d1c1e01acbb63e9f0da1901692eb37 2015-11-19 09:40 +0000 [8f71263e72] Matt Jordan * res/res_pjsip_outbound_registration: Apply configuration on object type load When Asterisk is configured to use a dynamic sorcery backend (such as res_sorcery_astdb) with 'registration' objects, it will fail to create the internal state objects associated with the registration objects on module load. This is due to nothing actually querying for the specific objects and calling their sorcery apply handler during module load. This patch fixes that by calling get_registrations in the sorcery observer's object_type_loaded handler. Doing this causes the sorcery backends to be asked for the current state of all registration objects, which causes the apply handler to be called and the internal run-time state to be created. ASTERISK-25575 #close Change-Id: Ie9306e797098c6d4da7bcf4a5434a15891508b23 2015-11-11 11:51 +0000 [0b508789ab] Alexander Traud * translate: Provide translation modules the result of SDP negotiation. Previously, a trancoding module did not have access to the joint but cached format. Therefore, the module did not have access to the attributes negotiated via SDP (line fmtp). Now, a translation module receives the joint format. ASTERISK-25545 #close Change-Id: Id6878a989b50573298dab115d3371ea369e1a718 2015-11-19 01:14 +0000 [1aa552b2a2] Alexander Traud * res_format_attr_h264: Do not reset string buffer. When no parameter is present, Asterisk does not generate the line fmtp, as expected. However, because a buffer was reset, even rtpmap and fmtp of previous media codecs got removed. Now, Asterisk does not reset other codecs in case of no parameter for H.264. ASTERISK-25573 #close Change-Id: I93811331f4a28c45418a9e14ee46c0debd47a286 2015-11-18 10:05 +0000 [3354b325c6] Matt Jordan * res_statsd: Add functions that support variable arguments Often, the metric names of statistics we are generating for StatsD have some dynamic component to them. This can be the name of a particular resource, or some internal status label in Asterisk. With the current set of functions, callers of the statsd API must first build the metric name themselves, then pass this to the API functions. This results in a large amount of boilerplate code and usage of either fixed length static buffers or dynamic memory allocation, neither of which is desireable. This patch adds two new functions to the StatsD API that support a printf style format specifier for constructing the metric name. A dynamic string, allocated in threadstorage, is used to build the metric name. This eases the burden on users of the StatsD API. Change-Id: If533c72d1afa26d807508ea48b4d8c7b32f414ea 2015-11-17 14:53 +0000 [d4a522d587] Richard Mudgett * res_pjsip_outbound_registration.c: Be tolerant of short registration timeouts. Change-Id: Ie16f5053ebde0dc6507845393709b4d6a3ea526d 2015-11-17 14:53 +0000 [e44ab3816c] Richard Mudgett * res_pjsip_outbound_registration.c: Fix 423 response handling. Receiving a 423 Interval Too Brief response after authentication for an outbound registration attempt results in assuming that the registrar has rejected the registration permanently. If there are no configured retries for fatal responses then the outbound registration is stopped for that endpoint. For registrations, PJSIP/PJPROJECT intercepts the handling of 423 responses and does not include any authentication in the updated registration request. When the updated request is challenged then the Asterisk code assumes that we were challenged again because the peer rejected the authentication we sent earlier. * Made registration challenges keep track of the CSeq number to determine if the received challenge response was for the request we thought we sent. If the response's CSeq number differs from the CSeq number we last sent with authentication then authenticate again because it is a challenge to a different request. Change-Id: I81b4bd36d1be095bab606e34b8b44e6302971b09 2015-11-03 14:36 +0000 [1e0040b88f] Tyler Cambron * StatsD: Add res_statsd compatibility Added a new api to res_statsd.c to allow it to receive a character pointer for the value argument. This allows for a '+' and a '-' to easily be sent with the value. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611 2015-11-16 13:56 +0000 [f62b642fe3] Matt Jordan * res/res_pjsip: Fix off nominal crash with requests that fail and have a timer When a request is sent using pjsip_endpt_send_request and fails, a condition exists where the request wrapper, which is an AO2 object, may be de-ref'd more times than it should. This occurs when the request's callback is called, and, in the callback, the timer on the PJSIP heap is cancelled. When that occurs, the request wrapper's lifetime is decremented. When pjsip_endpt_send_request fails, we unilaterally decrement the lifetime of the request wrapper again, even though we've already cancelled the reference associated with the timer. This patch checks the return result of pj_timer_heap_cancel_if_active before removing the reference associated with the timer. We now only decrement it in this case if a timer is cancelled as a result of the function call. Change-Id: I21332343a1a019c1117076f9bf2df27be2850102 2015-11-13 14:03 +0000 [fdd2afcd16] Mark Michelson * Confbridge: Add a user timeout option This option adds the ability to specify a timeout, in seconds, for a participant in a ConfBridge. When the user's timeout has been reached, the user is ejected from the conference with the CONFBRIDGE_RESULT channel variable set to "TIMEOUT". The rationale for this change is that there have been times where we have seen channels get "stuck" in ConfBridge because a network issue results in a SIP BYE not being received by Asterisk. While these channels can be hung up manually via CLI/AMI/ARI, adding some sort of automatic cleanup of the channels is a nice feature to have. ASTERISK-25549 #close Reported by Mark Michelson Change-Id: I2996b6c5e16a3dda27595f8352abad0bda9c2d98 2015-11-16 04:29 +0000 [7debb986a5] Alec Davis * app_queue: (try_calling): mutex 'qe->chan' freed more times than we've locked! commit aae45acbd (Mark Michelson 2015-04-15 10:38:02 -0500 6525) refer ASTERISK-24958 above commit removed ast_channel_lock(qe->chan); but failed to remove corresponding ast_channel_unlock(qe->chan); ASTERISK-25561 #close Reported Alec Davis Change-Id: Ie05f4e2d08912606178bf1fded57cc022c7a2e1a 2015-11-14 07:02 +0000 [afd9a89e5a] Joshua Colp * hashtab: Add NULL check when destroying iterator. The hashtab API is pretty NULL tolerant which has resulted in remaining callers not doing much checks themselves. Unfortunately the function to destroy an iterator does not do a NULL check and will result in a crash if passed NULL. This change fixes that. ASTERISK-25552 #close Change-Id: Ic1bf8eec3639e5a440f1c941d3ae3893ac6ed619 2015-11-13 14:32 +0000 [c0f2f8de45] Richard Mudgett * res_pjsip_rfc3326.c: Fix crash when channel goes away. If an authenticated incoming caller does not respond to our 200 OK INVITE response with an ACK then PJSIP will hangup the call. Unfortunately, there is a chance that the session's channel will go away between one use of the channel pointer and another when building the BYE request because the BYE is being built by the monitor thread and not the call's serializer thread. * Added a check to ensure that the thread trying to add the Reason header is the call's serializer thread. This ensures that the channel will not go away on us. Change-Id: I866388d2b97ea2032eaae3f3ab3f1ca6cbd2df89 2015-11-13 14:19 +0000 [4f43b85c92] Mark Michelson * Taskprocessors: Increase high-water mark In practical tests, we have seen certain taskprocessors, specifically Stasis subscription taskprocessors, cross the recently-added high-water mark and emit a warning. This high-water mark warning is only intended to be emitted when things have tanked on the system and things are heading south quickly. In the practical tests, the Stasis taskprocessors sometimes had a max depth of 180 tasks in them, and Asterisk wasn't in any danger at all. As such, this ups the high-water mark to 500 tasks instead. It also redefines the SIP threadpool request denial number to be a multiple of the taskprocessor high-water mark. Change-Id: Ic8d3e9497452fecd768ac427bb6f58aa616eebce 2015-11-11 11:46 +0000 [d8d3991390] Alexander Traud * format: Register format-attribute module with cached formats. In Asterisk 13, cached formats are created before their corresponding format- attribute module is registered. Cached formats are involved when a local extension is called. Therefore, ast_format_generate_sdp_fmtp did not work on local extensions. This change affects the Opus Codec, H.263 (Plus), H.264, and format-attribute modules provided externally. ASTERISK-25160 #close Change-Id: I1ea1f0483e5261e2a050112e4ebdfc22057d1354 2015-11-12 11:17 +0000 [367972e42d] Mark Michelson * res_pjsip distributor: Don't send 503 response to responses. When the SIP threadpool is backed up with tasks, we send 503 responses to ensure that we don't try to overload ourselves. The problem is that we were not insuring that we were not trying to send a 503 to an incoming SIP response. This change makes it so that we only send the 503 on incoming requests. Change-Id: Ie2b418d89c0e453cc6c2b5c7d543651c981e1404 2015-11-11 17:11 +0000 [2f9cb7d62b] Mark Michelson * res_pjsip: Deny requests when threadpool queue is backed up. We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816 2015-11-12 06:24 +0000 [4e5bf12b33] Joshua Colp * format_cap: Don't append the 'none' format when appending all. When appending all formats of a type all the codecs are iterated and added. This operation was incorrectly adding the ast_format_none format which is special in that it is supposed to be used when no format is present. It shouldn't be appended. ASTERISK-25535 Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c 2015-11-11 04:16 +0000 [07583c2888] Steve Davies * Further fixes to improper usage of scheduler When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b 2015-11-11 11:04 +0000 [b818d70533] Joshua Colp * threadpool: Handle worker thread transitioning to dead when going active. This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143 2015-11-10 09:27 +0000 [4bf84459c7] Alexander Traud * rtp_engine: Init a format-attribute module to its RFC defaults. Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52 2015-11-09 03:04 +0000 [1bff400df7] Alexander Traud * ast_format_cap_get_names: To display all formats, the buffer was increased. ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a 2015-11-09 07:04 +0000 [f3ac4d8090] Alexander Traud * ast_format_cap: Avoid format creation on module load, use cache instead. Since Asterisk 13, formats are immutable and cached. However while loading a module like chan_sip, some formats were created instead using cached ones. ASTERISK-25535 #close Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b 2015-11-06 07:54 +0000 [6d1bdb9d3b] Walter Doekes * func_callerid: Document that CALLERID(pres) is available. CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a 2015-11-06 07:52 +0000 [8410336681] Walter Doekes * docs: Fix a few typo's in app docs (more then, resourse). Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7 2015-11-06 07:36 +0000 [0d425f2eb4] Walter Doekes * xmldoc: Improve xmldoc wrapping of 'core show ...' output. Previously, the wrapping did both lookahead and lookback, which, together with color escape sequences, caused some lines to be wrapped way earlier than other lines. This led to inconsistent output. This simplifies the wrapping code and makes it more sane: if maxcolumns is hit, we simply jump back to the last space and wrap there. ASTERISK-25527 #close Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957 2015-11-06 06:57 +0000 [33752e0837] Sean Bright (license #5060) * res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP. In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual amount of channels is negotiated in-band. Therefore now, the Opus codec and its attribute rtpmap are registered with two channels. ASTERISK-24779 #close Reported by: PowerPBX Tested by: Alexander Traud patches: asterisk-24779.patch submitted by Sean Bright (license #5060) Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b 2015-11-03 16:19 +0000 [6ff48319d9] Jonathan Rose * taskprocessor: Add high water mark warnings If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c 2015-11-04 14:31 +0000 [506aea26e6] Matt Jordan * main/dial: Protect access to the format_cap structure of the requesting channel When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f 2015-10-30 22:57 +0000 [d098d00424] Corey Farrell * Fix cli display of build options. A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325 2015-11-03 11:15 +0000 [afec1b1b64] Matt Jordan * res_pjsip/location: Destroy contact_status objects on contact deletion The contact_status Sorcery objects are currently not destroyed when a contact is deleted. This causes the contact's last known RTT/status to be 'sticky' when the contact itself may no longer exist. This patch causes the contact_status objects associated with both dynamic and static contacts to be destroyed if the AoR holding those contacts is also destroyed (or via other paths where a contact may be deleted.) Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e 2015-11-03 10:58 +0000 [715f770c9f] Matt Jordan * pjsip_configuration: On delete, remove the persistent version of an endpoint When an endpoint is deleted (such as through an API), the persistent endpoint currently continues to lurk around. While this isn't harmful from a memory consumption perspective - as all persistent endpoints are reclaimed on shutdown - it does cause Stasis endpoint related operations to continue to believe that the endpoint may or may not exist. This patch causes the persistent endpoint related to a PJSIP endpoint to be destroyed if the PJSIP endpoint is deleted. Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb 2015-11-03 08:15 +0000 [f0f190af08] Matt Jordan * main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec field The JSON packing for the ContactStatusChange event forgot to include the roundtrip_usec field. As a result, the field never showed up in any event, even when the data was available. This patch corrects that error by properly packing the JSON blob with the data. Change-Id: I8df80da659a44010afbd48f645967518ff5daa17 2015-11-02 20:24 +0000 [0393bd6bed] Corey Farrell * chan_sip: Allow websockets to be disabled. This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7 2015-11-02 17:19 +0000 [6fbffe42e1] Mark Michelson * res_pjsip: Set threadpool max size default to 50. During a stress test of subscriptions, a huge blast of subscription-related traffic resulted in the threadpool expanding to a ridiculous number of threads. The balooning of threads resulted in an increase of memory, which led to a crash due to being out of memory. An easy fix for the particular test was to limit the size of the threadpool, thus reining in the amount of memory that would be used. It was decided that there really is no downside to having a non-infinite default value for the maximum size of the threadpool, so this change introduces 50 threads as the maximum threadpool size for the SIP threadpool. ASTERISK-25513 #close Reported by John Bigelow Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be 2015-11-02 06:57 +0000 [11e54b1932] Matt Jordan * pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction When an AoR is created or destroyed dynamically, the scheduled OPTIONS requests that qualify the contacts on the AoR are not necessarily started or destroyed, particularly for persistent contacts created for that AoR. This patch adds create/update/delete sorcery observers for an AoR, which schedule/unschedule the qualifies as expected. Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d 2015-10-30 13:22 +0000 [118d628e08] Matt Jordan * Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configs This patch adds a rule for installing the Super Awesome Company based 'Basic PBX' configuration files. As part of adding this rule, a bit of the content that makes up installing the configuration files under the 'samples' target was refactored into a make subroutine for usage by additional later config make targets. Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404 2015-10-29 08:28 +0000 [9a021a42ad] Joshua Colp * res_pjsip_pubsub: Fix assertion when UAS dialog creation fails. When compiled with assertions enabled one will occur when destroying the subscription tree when UAS dialog creation fails. This is because the code assumes that a dialog will always exist on a subscription tree when in reality during this specific scenario it won't. This change makes it so a dialog is not removed from the subscription tree if it is not present. ASTERISK-25505 #close Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee 2015-10-26 11:42 +0000 [1256aedf66] Alexander Traud * chan_sip: Do not send all codecs on INVITE. Since version 13, Asterisk sent all allowed codecs as callee, even when the caller did not request/support them. In case of dynamic RTP payloads, this led to the same ID for different codecs, which is not allowed by SIP/SDP. Now, the intersection between the requested and the supported codecs is send again. ASTERISK-24543 #close Change-Id: Ie90cb8bf893b0895f8d505e77343de3ba152a287 2015-10-24 13:08 +0000 [5f593e7c38] gtjoseph * build: GCC 5.1.x catches some new const, array bounds and missing paren issues Fixed 1 issue in each of the affected files. ASTERISK-25494 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I818f149cd66a93b062df421e1c73c7942f5a4a77 2015-10-20 16:02 +0000 [162acd45f7] gtjoseph * res_pjsip: Add "like" processing to pjsip list and show commands Add the ability to filter output from pjsip list and show commands using the "like" predicate like chan_sip. For endpoints, aors, auths, registrations, identifyies and transports, the modification was a simple change of an ast_sorcery_retrieve_by_fields call to ast_sorcery_retrieve_by_regex. For channels and contacts a little more work had to be done because neither of those objects are true sorcery objects. That was just removing the non-matching object from the final container. Of course, a little extra plumbing in the common pjsip_cli code was needed to parse the "like" and pass the regex to the get_container callbacks. Some of the get_container code in res_pjsip_endpoint_identifier was also refactored for simplicity. ASTERISK-25477 #close Reported by: Bryant Zimmerman Tested by: George Joseph Change-Id: I646d9326b778aac26bb3e2bcd7fa1346d24434f1 2015-10-21 11:51 +0000 [c58091737d] Kevin Harwell * res_pjsip_outbound_registration: registration stops due to fatal 4xx response During outbound registration it is possible to receive a fatal (any permanent/ non-temporary 4xx, 5xx, 6xx) response from the registrar that is simply due to a problem with the registrar itself. Upon receiving the failure response Asterisk terminates outbound registration for the given endpoint. This patch adds an option, 'fatal_retry_interval', that when set continues outbound registration at the given interval up to 'max_retries' upon receiving a fatal response. ASTERISK-25485 #close Change-Id: Ibc2c7b47164ac89cc803433c0bbe7063bfa143a2 2015-10-22 17:07 +0000 [ebe69dee0d] Mark Michelson * format_cap: Detect vector allocation failures. A crash was seen on a system that ran out of memory due to Asterisk not checking for vector allocation failures in format_cap.c. With this change, if either of the AST_VECTOR_INIT calls fail, we will return a value indicating failure. Change-Id: Ieb9c59f39dfde6d11797a92b45e0cf8ac5722bc8 2015-10-02 15:32 +0000 [3b19efefef] Mark Michelson * res_pjsip_pubsub: Prevent sending NOTIFY on destroyed dialog. A certain situation can result in our attempting to send a NOTIFY on a destroyed dialog. Say we attempt to send a NOTIFY to a subscriber, but that subscriber has dropped off the network. We end up retransmitting that NOTIFY until the appropriate SIP timer says to destroy the NOTIFY transaction. When the pjsip evsub code is told that the transaction has been terminated, it responds in kind by alerting us that the subscription has been terminated, destroying the subscription, and then removing its reference to the dialog, thus destroying the dialog. The problem is that when we get told that the subscription is being terminated, we detect that we have not sent a terminating NOTIFY request, so we queue up such a NOTIFY to be sent out. By the time that queued NOTIFY gets sent, the dialog has been destroyed, so attempting to send that NOTIFY can result in a crash. The fix being introduced here is actually a reintroduction of something the pubsub code used to employ. We hold a reference to the dialog and wait to decrement our reference to the dialog until our subscription tree object is destroyed. This way, we can send messages on the dialog even if the PJSIP evsub code wants to terminate earlier than we would like. In doing this, some NULL checks for subscription tree dialogs have been removed since NULL dialogs are no longer actually possible. Change-Id: I013f43cddd9408bb2a31b77f5db87a7972bfe1e5 2015-09-29 14:53 +0000 [0a346f095f] Mark Michelson * res_pjsip_pubsub: Ensure dialog lock balance. When sending a NOTIFY, we lock the dialog and then unlock the dialog when finished. A recent change made it so that the subscription tree's dialog pointer will be set NULL when sending the final NOTIFY request out. This means that when we attempt to unlock the dialog, we pass a NULL pointer to pjsip_dlg_dec_lock(). The result is that the dialog remains locked after we think we have unlocked it. When a response to the NOTIFY arrives, the monitor thread attempts to lock the dialog, but it cannot because we never released the dialog lock. This results in Asterisk being unable to process incoming SIP traffic any longer. The fix in this patch is to use a local pointer to save off the pointer value of the subscription tree's dialog when locking and unlocking the dialog. This way, if the subscription tree's dialog pointer is NULLed out, the local pointer will still have point to the proper place and the dialog lock will be unlocked as we expect. Change-Id: I7ddb3eaed7276cceb9a65daca701c3d5e728e63a 2015-09-28 16:36 +0000 [ad39508095] Mark Michelson * res_pjsip_pubsub: Prevent crashes on final NOTIFY. The SIP dialog is removed from the subscription tree when the final NOTIFY is sent. However, after the final NOTIFY is sent, the persistence update function still attempts to access the cseq from the dialog, resulting in a crash. This fix removes the subscription persistence at the same time that the dialog is removed from the subscription tree. This way, there is no attempt to update persistence when the subscription is being destroyed. Change-Id: Ibb46977a6cef9c51dc95f40f43446e3d11eed5bb 2015-09-17 17:28 +0000 [067f408760] Mark Michelson * res_pjsip_pubsub: Remove serializer when sending final NOTIFY. There have been crashes seen where a taskprocessor's listener is NULL unexpectedly. Looking at backtraces, the problem was specifically seen in PJSIP serializers. Subscriptions make the mistake of removing a serializer from a dialog during subscription tree destruction. Since subscription trees are reference-counted, guaranteeing the circumstances behind the destruction are not possible. This makes it so that the dialog serializer can be removed while not holding the dialog lock. This makes it possible for the distributor to get a pointer to the dialog serializer and have that serializer get freed out from under it. The fix for this is to remove the serializer from a subscription dialog when sending the final NOTIFY. This guarantees that the serializer is removed with the dialog lock held. By doing this, we guarantee that if the distributor gains access to the dialog's serializer, it will not be possible for the serializer to get freed by another thread. Change-Id: I21f5dac33529f65cec45679bdace60670800ff66 2015-09-02 09:14 +0000 [1bcc592765] Mark Michelson * res_pjsip_pubsub: Fix crash on destruction of empty subscription tree. If an old persistent subscription is recreated but then immediately destroyed because it is out of date, the subscription tree will have no leaf subscriptions on it. This was resulting in a crash when attempting to destroy the subscription tree. A simple NULL check fixes this problem. Change-Id: I85570b9e2bcc7260a3fe0ad85904b2a9bf36d2ac 2015-09-01 15:47 +0000 [b3cc2bd7df] Mark Michelson * res_pjsip_pubsub: Solidify lifetime and ownership of objects. There have been crashes and general instability seen in the pubsub code, so this patch introduces three changes to increase the stability. First, the ownership model for subscriptions has been modified. Due to RLS, subscriptions are stored in memory as a tree structure. Prior to my patch, the PJSIP subscription was the owner of the subscription tree. When the PJSIP subscription told us that it was terminating, we started destroying the subscription tree along with all of the individual leaf subscriptions that belong to the tree. The problem with this model is that the two actors in play here, the PJSIP subscription and the individual leaf subscriptions, need to have joint ownership of the subscription tree. So now, the PJSIP subscription and the individual leaf subscriptions each have a reference to the subscription tree. This way, we will not actually free memory until no players are left that care. The PJSIP subscription is a bigger stakeholder, in that if the PJSIP subscription's reference to the subscription tree is removed, the subscription tree instructs the leaf subscriptions to shut down and drop their references to the subscription tree when possible. The individual leaf subscriptions, upon being told to shut down, can drop their stasis subscriptions or whatever they use to learn of new state, and then drop their reference to the subscription tree once they are ready to die. Second, the lifetime of a PJSIP subscription's reference to our subscription tree has been altered. As I learned from doing a deep dive, the PJSIP evsub code can tell Asterisk multiple times that the subscription has been terminated, and not all of these times are especially helpful. I have altered the message flow that we use for SIP subscriptions such that we will always drop the PJSIP subscription's reference to the subscription tree when we send the NOTIFY that terminates a SIP subscription. This also means that we will now queue NOTIFY requests to be sent after responding to incoming SUBSCRIBEs so that we can have predictable state changes from the PJSIP evsub code. Third, the synchronization of operations has been improved. PJSIP can call into our code from a serializer thread (e.g. upon receiving an incoming request) or from the monitor thread (e.g. when a subscription times out). Because of this, there is the possibility of competing threads stepping on each other. PJSIP attempts to do some synchronization on its own by always keeping the dialog lock held when it calls into us. However, since we end up pushing tasks into the serializer, the result was that serialized operations were not grabbing the dialog lock and could, as a result, step on something that was being attempted by a different thread. Now we ensure that serialized operations grab the dialog lock, then check for extenuating circumstances, then proceed with their operation if they can. Change-Id: Iff2990c40178dad9cc5f6a5c7f76932ec644b2e5 2015-10-19 15:28 +0000 [c8c65dfa41] Richard Mudgett * strings.c: Fix __ast_str_helper() to always return a terminated string. Users of functions which call __ast_str_helper() such as the ones listed below are likely to not check the return value for failure so ensuring that the string is always nil terminated is a good safety measure. ast_str_set_va() ast_str_append_va() ast_str_set() ast_str_append() Change-Id: I36ab2d14bb6015868b49329dda8639d70fbcae07 2015-10-19 15:27 +0000 [b271d4a28a] Richard Mudgett * Add missing failure checks to ast_str_set_va() callers. Change-Id: I0c2cdcd53727bdc6634095c61294807255bd278f 2015-10-21 11:44 +0000 [f2725c8b77] Joshua Colp * res_pjsip: Move URI validation to use time. In a realtime based system with a limited number of threadpool threads it is possible for a deadlock to occur. This happens when permanent endpoint state is updated, which will cause database queries to be done. These queries may result in URI validation being done which is done synchronously using a PJSIP thread. If all PJSIP threads are in use processing traffic they themselves may be blocked waiting to get the permanent endpoint container lock when identifying an endpoint. This change moves URI validation to occur at use time instead of configuration time. While this comes at a cost of not seeing a problem until you use it it does solve the underlying deadlock problem. ASTERISK-25486 #close Change-Id: I2d7d167af987d23b3e8199e4a68f3359eba4c76a 2015-10-21 08:08 +0000 [84ff075d41] Alexander Traud * format: Update the maximum packetization time for iLBC 30. In September 2006, the maximum packetization time (ptime) were set to such a low value, packetization was disabled for many codecs actually. This was fixed for many codecs but not for iLBC 30. This enables packetization for iLBC which can be enabled for example via allow=ilbc:60,gsm,alaw,ulaw in the file sip.conf. ASTERISK-7803 Change-Id: I2ef90023d35efb7cb8fe96ed74f53f6846ffad12 2015-10-21 09:51 +0000 [869ef2a8ee] Alexander Traud * chan_sip: Fix autoframing=yes. With Asterisk 13, the structures ast_format and ast_codec changed. Because of that, the paketization timing (framing) of the RTP channel moved away from the formats/codecs. In the course of that change, the ptime of the callee was not honored anymore, when the optional autoframing was enabled. ASTERISK-25484 #close Change-Id: Ic600ccaa125e705922f89c72212c698215d239b4 2015-10-20 22:24 +0000 [9fd2adc204] Matt Jordan * rest-api-templates: Wikify error code response reasons Error response code descriptions may contain wiki markup that need to be escaped. Without this patch, Confluence will reject the document being sent and the responsible script will raise an exception. Change-Id: I21fcb66fee7f6332381f2b99b1b0195dff215ee5 2015-10-20 12:06 +0000 [72cbb6df55] Matt Jordan * funcs/func_holdintercept: Actually add the HOLD_INTERCEPT function When ab803ec342 was committed, it accidentally forgot to actually *add* the HOLD_INTERCEPT function. This highlights two interesting points: * Gerrit forces you to put the patch as it is going to into the repo up for review, which Review Board did not. Yay Gerrit. * No one apparently bothered to use this feature, or else they don't know about it. I'm going to go with the latter explanation. ASTERISK-24922 Change-Id: Ida38278f259dd07c334a36f9b7d5475b5db72396 2015-10-19 19:59 +0000 [9fc9777fa3] Matt Jordan * contrib/scripts/autosupport: Update for Asterisk 13 This patch adds some minor tweaks for autosupport to update it for Asterisk 13. This includes: * Finally removing most references to Zaptel * Adding support for some additional 'core' commands, and fixing nomenclature that generally hasn't been used for some time * Adding some PJSIP/SIP commands to gather endpoints/peers and active channels Change-Id: Ic997b418cbd9313588b6608e50f47b0ce6f4f1f1 2015-10-14 14:15 +0000 [dc6ec661b3] mdu113 * res_config_pgsql.c: Fix deadlock loading realtime configuration. On v13, loading several thousand PJSIP endpoints on Asterisk start causes a deadlock most of the time. Thanks to mdu113 for discovering that there was a call to pgsql_exec() not protected by the pgsql_lock reentrancy lock. {quote} I believe a code path exists that attempts to use pgsql connection without locking pgsql_lock. I believe what happens during that deadlock that I see is two concurrent threads are both attempting to send query to pgsql, one of the thread is using a code path without locking pgsql_lock. If they managed to send queries at the same time, it seems postgres ignores one of the queries and replies only to the one of them. If it happens so that the thread holding the lock didn't receive the reply it will wait for it (and hold the lock) forever (or at least for very long time), thus completely blocking all access to db. {quote} * Added missing reentrancy locking around pgsql_exec() in find_table(). * Moved unlock of pgsql_lock in unload_module() to avoid locking inversion between the psql_tables list lock and the pgsql_lock. ASTERISK-25455 #close Reported by: mdu113 Patches: res_config_pgsql.c-connlock2.diff (license #5543) patch uploaded by mdu113 Change-Id: Id9e7cdf8a3b65ff19964b0cf942ace567938c4e2 2015-10-13 14:13 +0000 [f8707ae9a5] Olle Johansson (License 5267) * channels/chan_sip: Set cause code to 44 on RTP timeout To quote Olle: "When issuing a hangup due to RTP timeouts the cause code is not set. I have selected 44 based on Cisco's implementation..." ASTERISK-25135 #close Reported by: Olle Johansson patches: rtp-timeout-cause-1.8.diff uploaded by Olle Johansson (License 5267) Change-Id: Ia62100c55077d77901caee0bcae299f8dc7375fc 2015-10-10 15:20 +0000 [486b172b50] Ivan Poddubny * Build: Add menuselect options for using compiler sanitizers This patch adds menuselect options for building Asterisk with various sanitizers provided by gcc and clang. When one of *SANITIZER flags is set in menuselect, the appropriate option is added to CFLAGS ad LDFLAGS for the build. Information on sanitizers in the project wiki: https://github.com/google/sanitizers/wiki GCC Manual: https://gcc.gnu.org/onlinedocs/gcc/Debugging-Options.html Clang Compiler User's Manual: http://clang.llvm.org/docs/UsersManual.html#controlling-code-generation ASTERISK-24718 #close Reported by: Badalian Vyacheslav Change-Id: Iafa51b792b7bcb20e848b99d16cf362d08590fa0 2015-10-12 11:21 +0000 [e14023ca35] Richard Mudgett * config.c: Fix off-nominal memory leak. Change-Id: I06e346e9a5c63cc5071e7eda537310c4b43bffe0 2015-10-12 11:20 +0000 [a99e821520] Richard Mudgett * config.c: Fix potential memory corruption after [section](+). The memory corruption could happen if the [section](+) is the last section in the file with trailing comments. In this case process_text_line() has left *last_cat is set to newcat and newcat is destroyed. Change-Id: I0d1d999f553986f591becd000e7cc6ddfb978d93 2015-10-12 11:21 +0000 [8d31d2526b] Richard Mudgett * config.c: Fix #include after [section](+). An #include right after a [section](+) would associate any variable assignments before a new section in the #include with the wrong section. * Fix section association by setting the current section to the appended section. * Fix '+' and '!' section flag interaction corner case depending upon which flag came first. If the '!' came first then it would be ignored. If the '!' came after then it would affect the appended section. The '!' will now no longer be ignored. ASTERISK-25461 #close Reported by: Sean Pimental Change-Id: Ic9d3191c8758048e2cbce6432f854b32531731c3 2015-10-06 18:01 +0000 [3329c714f7] Richard Mudgett * res_pjsip: Fix deadlock when sending out-of-dialog requests. The struct send_request_wrapper has a pjsip lock associated with it that is created non-recursive. There is a code path for the struct send_request_wrapper lock that will attempt to lock it recursively. The reporter's deadlock showed that the thread calling endpt_send_request() deadlocked itself right after the wrapper object got created. Out-of-dialog requests such as MESSAGE, qualify OPTIONS, and unsolicited MWI NOTIFY messages can hit this deadlock. * Replaced the struct send_request_wrapper pjsip lock with the mutex lock that can come with an ao2 object since all of Asterisk's mutexes are recursive. Benefits include removal of code maintaining the pjsip non-recursive lock since ao2 objects already know how to maintain their own lock and the lock will show up in the CLI "core show locks" output. ASTERISK-25435 #close Reported by: Dmitriy Serov Change-Id: I458e131dd1b9816f9e963f796c54136e9e84322d 2015-10-06 11:05 +0000 [a1435aa3fa] Stefan Engström * res/res_rtp_asterisk.c: Fix incorrect assignment of frame->subclass.frame_ending In ast_rtp_read, the value of the variable 'mark' which we try to assign to a frame->subclass.frame_ending may be 0, 1 or (1<<23), but we should translate it to 0 or 1. ASTERISK-25451 #close Change-Id: I53bdf5c026041730184a6a809009c028549ce626 2015-10-07 01:24 +0000 [3357678b94] Ivan Poddubny * func_presencestate: Return "not_set" when no data is set in AstDB Return AST_PRESENCE_NOT_SET when CustomPresence AstDB key does not exist, i.e. when a new CustomPresence is added in the dialplan. ASTERISK-25400 #close Reported by: Andrew Nagy Change-Id: I6fb17b16591b5a55fbffe96f3994ec26b1b1723a 2015-10-06 20:43 +0000 [b714b2152d] Matt Jordan * res/res_rtp_asterisk: Fix assignment after ao2 decrement When we decide we will no longer schedule an RTCP write, we remove the reference to the RTP instance, then assign -1 to the stored scheduler ID in case something else comes along and wants to see if anything is scheduled. That scheduler ID is on the RTP instance. After 60a9172d7ef2 was merged to fix the regression introduced by 3cf0f29310, this improper assignment on a potentially destroyed object started getting tripped on the build agents. Frankly, this should have been crashing a lot more often earlier. I can only assume that the timing was changed just enough by both changes to start actually hitting this problem. As it is, simply moving the assignment prior to the ao2 deference is sufficient to keep the RTP instance from being referenced when it is very, truly, aboslutely dead. (Note that it is still good practice to assign -1 to the scheduler ID when we know we won't be scheduling it again, as the ao2 deref *may* not always destroy the ao2 object.) ASTERISK-25449 Change-Id: Ie6d3cb4adc7b1a6c078b1c38c19fc84cf787cda7 2015-10-06 12:40 +0000 [f939e2bd48] Florian Sauerteig * chan_sip: Fix port parsing for IPv6 addresses in SIP Via headers. If a Via header containes an IPv6 address and a port number is ommitted, as it is the standard port, we now leave the port empty and to not set it to the value after the first colon of the IPv6 address. ASTERISK-25443 #close Change-Id: Ie3c2f05471cd006bf04ed15598589c09577b1e70 2015-10-05 16:53 +0000 [426263a64d] Richard Mudgett * chan_pjsip: Fix crash on reINVITE before initial INVITE completes. Apparently some endpoints attempt to send a reINVITE before completing the initial INVITE transaction. In this case PJSIP responds appropriately to the reINVITE with a 491 INVITE request pending. Unfortunately chan_pjsip is using the initial INVITE transaction state to determine if an INVITE is the initial INVITE or a reINVITE. Since the initial INVITE transaction has not been confirmed yet chan_pjsip thinks the reINVITE is an initial INVITE and starts another PBX thread on the channel. The extra PBX thread ensures that hilarity ensues. * Fix checks for a reINVITE on incoming requests to look for the presence of a to-tag instead of the initial INVITE transaction state. * Made caller_id_incoming_request() determine what to do if there is a channel on the session or not. After a channel is created it is too late to just store the new party id on the session because the session's party id has already been copied to the channel's caller id. ASTERISK-25404 #close Reported by: Chet Stevens Change-Id: Ie78201c304a2b13226f3a4ce59908beecc2c68be 2015-10-05 21:34 +0000 [50fa9ff997] Matt Jordan * Fix improper usage of scheduler exposed by 5c713fdf18f When 5c713fdf18f was merged, it allowed for scheduled items to have an ID of '0' returned. While this was valid per the documentation for the API, it was apparently never returned previously. As a result, several users of the scheduler API viewed the result as being invalid, causing them to reschedule already scheduled items or otherwise fail in interesting ways. This patch corrects the users such that they view '0' as valid, and a returned ID of -1 as being invalid. Note that the failing HEP RTCP tests now pass with this patch. These tests failed due to a duplicate scheduling of the RTCP transmissions. ASTERISK-25449 #close Change-Id: I019a9aa8b6997584f66876331675981ac9e07e39 2015-08-26 16:58 +0000 [8f777ab584] Debian Amtelco * chan_pjsip: Add Referred-By header to the PJSIP REFER packet. Some systems require the REFER packet to include a Referred-By header. If the channel variable SIPREFERREDBYHDR is set, it passes that value as the Referred-By header value. Otherwise, it adds the current dialog’s local info. Reported by: Dan Cropp Tested by: Dan Cropp Change-Id: I3d17912ce548667edf53cb549e88a25475eda245 2015-10-03 06:27 +0000 [74635b5638] Ivan Poddubny * manager: Fix GetConfigJSON returning invalid JSON When GetConfigJSON was introduced back in 1.6, it returned each section as an array of strings: ["key=value", "key2=value2"]. Afterwards, it was changed a few times and became ["key": "value", "key2": "value2"], which is not a correct JSON. This patch fixes that by constructing a JSON object {} instead of an array []. Also, the keys "istemplate" and "tempates" that are used to indicate templates and their inherited categories are now wrapped in quotes. ASTERISK-25391 #close Reported by: Bojan Nemčić Change-Id: Ibbe93c6a227dff14d4a54b0d152341857bcf6ad8 2015-09-30 17:28 +0000 [40c69e78f5] Richard Mudgett * res_sorcery_memory_cache.c: Fix deadlock with scheduler. A deadlock can happen when a sorcery object is being expired from the memory cache when at the same time another object is being placed into the memory cache. There are a couple other variations on this theme that could cause the deadlock. Basically if an object is being expired from the sorcery memory cache at the same time as another thread tries to update the next object expiration timer the deadlock can happen. * Add a deadlock avoidance loop in expire_objects_from_cache() to check if someone is trying to remove the scheduler callback from the scheduler. ASTERISK-25441 #close Change-Id: Iec7b0bdb81a72b39477727b1535b2539ad0cf4dc 2015-10-01 14:30 +0000 [dfeb513e85] Richard Mudgett * res_sorcery_memory_cache.c: Replace inline code with function. Make sorcery_memory_cache_close() call remove_all_from_cache() instead of partially inlining it. ASTERISK-25441 Change-Id: I1aa6cb425b1a4307096f3f914d17af8ec179a74c 2015-10-01 14:27 +0000 [ced0a2d71b] Richard Mudgett * res_sorcery_memory_cache.c: Shutdown in a less crash potential order. Basically you should shutdown in the opposite order of how you setup since later setup pieces likely depend on earlier setup pieces. e.g., Registering your external API with the rest of the system should be the last thing setup and the first thing unregistered during shutdown. Change-Id: I5715765b723100c8d3c2642e9e72cc7ad5ad115e 2015-09-30 17:27 +0000 [cc279eea11] Richard Mudgett * res_sorcery_memory_cache.c: Misc tweaks. Change-Id: I8cd32dffbb4f33bb0c39518d6e4c991e73573160 2015-09-30 17:27 +0000 [9af3b613f6] Richard Mudgett * res_sorcery_memory_cache.c: Made use OBJ_SEARCH_MASK. Change-Id: Ibca6574dc3c213b29cc93486e01ccd51f5caa46c 2015-09-30 13:42 +0000 [56ed7b9dd5] Joshua Colp * res_rtp_asterisk: Move "Set role" warning to be debug. In practice the set_role API callback can be invoked even when no ICE is present on an RTP instance. This can occur if ICE has not been enabled on it. ASTERISK-25438 #close Change-Id: I0e17e4316f0f0d7f095c78c3d4fd73a913b6ba69 2015-09-28 15:31 +0000 [ddebb217f0] Richard Mudgett * sched.c: Add warning about negative time interval request. Change-Id: Ib91435fb45b7f5f7c0fc83d0eec20b88098707bc 2015-09-29 14:53 +0000 [d30939b6e8] Kevin Harwell * ARI: Changed version from 1.8.0 to 1.9.0 Change-Id: I510991c60d28d171f47c4b58bba4947f7fc71b13 2015-09-25 18:37 +0000 [5f19c9bade] Richard Mudgett * res/ari/config.c: Fix user sort compare function. Made use the ao2 sort compare template function and OBJ_SEARCH_xxx identifiers. Change-Id: Ic53005dc5aafa7a36c72300dd89b75fb63c92f4c 2015-09-25 17:26 +0000 [3a85764039] Richard Mudgett * res/ari/config.c: Optimize conf_alloc() object init. * Now conf_alloc() has more off nominal error checking. * Eliminated RAII_VAR() use in conf_alloc(). * Eliminated a dubius shortcut when destroying cfg->general in conf_destructor() that would cause a crash if cfg->general failed to get allocated. * Add some ACO registration section comments. Change-Id: Ia40c2b1b2d0777d641605118ae019c5a73865e1a 2015-09-25 16:48 +0000 [028033e5a8] Richard Mudgett * res/ari/config.c: Fix conf_alloc() object init. Need to finish initializing the string fields in the ao2 object before putting any default strings into them. ASTERISK-25383 #close Reported by: yaron nahum Change-Id: I9f7f3a03f0c4991a01593abf8697b9a587c0ea84 2015-09-27 20:45 +0000 [90165e306d] Matt Jordan * res/res_stasis: Fix accidental subscription to 'all' bridge topic When b99a7052621700a1aa641a1c24308f5873275fc8 was merged, subscribing to a NULL bridge will now cause app_subscribe_bridge to implicitly subscribe to all bridges. Unfortunately, the res_stasis control loop did not check that a bridge changing on a channel's control object was actually also non-NULL. As a result, app_subscribe_bridge will be called with a NULL bridge when a channel leaves a bridge. This causes a new subscription to be made to the bridge. If an application has also subscribed to the bridge, the application will now have two subscriptions: (1) The explicit one created by the app (2) The implicit one accidentally created by the control structure As a result, the 'BridgeDestroyed' event can be sent multiple times. This patch corrects the control loop such that it only subscribes an application to a new bridge if the bridge pointer is non-NULL. ASTERISK-24870 Change-Id: I3510e55f6bc36517c10597ead857b964463c9f4f 2015-09-04 13:51 +0000 [e1223ff6db] Scott Griepentrog * Scripts: check file versions of Asterisk and dependencies To help in diagnosing mismatched modules and libraries, this script scans for version, repository, and source information and reports what is found. ASTERISK-25376 #close Reported by: Ashley Sanders Change-Id: Ib0642d0fb96712476f59760d6d137a24633fe2d6 2015-09-24 14:56 +0000 [6b1e7583c1] Richard Mudgett * app_queue.c: Force COLP update if outgoing channel name changed. * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 #close Reported by: John Hardin Change-Id: Ie275ea9e99c092ad369db23e0feb08c44498c172 2015-09-24 14:20 +0000 [6bf304bf25] Richard Mudgett * app_queue.c: Factor out a connected line update routine. Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: I33bbd033596fcb0208d41d8970369b4e87b806f3 2015-09-24 13:27 +0000 [e36b5f1e8e] Richard Mudgett * app_dial.c: Make 'A' option pass COLP updates. While the 'A' option is playing the announcement file allow the caller and peer to exchange COLP update frames. ASTERISK-25423 Reported by: John Hardin Change-Id: Iac6cf89b56d26452c6bb88e9363622bbf23895f9 2015-09-24 12:59 +0000 [747bfac895] Richard Mudgett * app_dial.c: Force COLP update if outgoing channel name changed. * When a call is answered and the outgoing channel name has changed then force a connected line update because the channel is no longer the same. The channel was masqueraded into by another channel. This is usually because of a call pickup. Note: Forwarded calls are handled in a controlled manner so the original channel name is replaced with the forwarded channel. ASTERISK-25423 Reported by: John Hardin Change-Id: I2e01f7a698fbbc8c26344a59c2be40c6cd98b00c 2015-09-24 12:37 +0000 [14481d9aa0] Richard Mudgett * app_dial.c: Factor out a connected line update routine. Replace inlined code with update_connected_line_from_peer(). ASTERISK-25423 Reported by: John Hardin Change-Id: Ia14f18def417645cd7fb453e1bdac682630a5091 2015-09-23 17:41 +0000 [bbeda190c3] Richard Mudgett * app_dial.c: Remove some no-op code. Change-Id: Ice1884a94315d3cb7e3bbd47a9fba76a27276c54 2015-09-23 14:02 +0000 [f050fa76eb] Mark Michelson * logger: Prevent duplicate dynamic channels from being added. There was a problem observed where the "logger add channel" CLI command would allow for a channel with the same name to be added multiple times. This would result in each message being written out to the same file multiple times. The problem was due to the difference in how logger channel filenames are stored versus the format they are allowed to be presented when they are added. For instance, if adding the logger channel "foo" through the CLI, the result would be a logger channel with the file name /var/log/asterisk/foo being stored. So when trying to add another "foo" channel, "foo" would not match "/var/log/asterisk/foo" so we'd happily add the duplicate channel. The fix presented here is to introduce two new methods in the logger code: * make_filename(): given a logger channel name, this creates the filename for that logger channel. * find_logchannel(): given a logger channel name, this calls make_filename() and then traverses the list of logchannels in order to find a match. This change has made use of make_filename() and find_logchannel() throughout to more consistently behave. ASTERISK-25305 #close Reported by Mark Michelson Change-Id: I892d52954d6007d8bc453c3cbdd9235dec9c4a36 2015-09-24 14:49 +0000 [629458d349] Mark Michelson * Do not swallow frames on channels leaving bridges. When leaving a bridge, indications on a channel could be swallowed by the internal indication logic because it appears that the channel is on its way to be hung up anyway. One such situation where this is detrimental is when channels on hold are redirected out of a bridge. The AST_CONTROL_UNHOLD indication from the bridging code is swallowed, leaving the channel in question to still appear to be on hold. The fix here is to modify the logic inside ast_indicate_data() to not drop the indication if the channel is simply leaving a bridge. This way, channels on hold redirected out of a bridge revert to their expected "in use" state after the redirection. ASTERISK-25418 #close Reported by Mark Michelson Change-Id: If6115204dfa0551c050974ee138fabd15f978949 2015-09-22 17:08 +0000 [5f15cd93f0] Richard Mudgett * app_page.c: Fix crash when forwarding with a predial handler. Page uses the async method of dialing with the dial API. When a call gets forwarded there is no calling channel available. If the predial handler was set then the calling channel could not be put into auto-service for the forwarded call because it doesn't exist. A crash is the result. * Moved the callee predial parameter string processing to before the string is passed to the dial API rather than having the dial API do it. There are a few benefits do doing this. The first is the predial parameter string processing doesn't need to be done for each channel called by the dial API. The second is in async mode and the forwarded channel is to have the predial handler executed on it then the non-existent calling channel does not need to be present to process the predial parameter string. * Don't start auto-service on a non-existent calling channel to execute the predial handler when the dial API is in async mode and forwarding a call. ASTERISK-25384 #close Reported by: Chet Stevens Change-Id: If53892b286d29f6cf955e2545b03dcffa2610981 2015-09-03 21:19 +0000 [b50e372394] Matt Jordan * ARI: Add events for Contact and Peer Status changes This patch adds support for receiving events regarding Peer status changes and Contact status changes. This is particularly useful in scenarios where we are subscribed to all endpoints and channels, where we often want to know more about the state of channel technology specific items than a single endpoint's state. ASTERISK-24870 Change-Id: I6137459cdc25ce27efc134ad58abf065653da4e9 2015-09-04 12:24 +0000 [3502c0431d] Matt Jordan * res/res_stasis_device_state: Allow for subscribing to 'all' device state This patch adds support for subscribing to all device state changes. This is done either by subscribing to an empty device, e.g., 'eventSource=deviceState:', or by the WebSocket connection specifying that it wants all state in the system. ASTERISK-24870 Change-Id: I9cfeca1c9e2231bd7ea73e45919111d44d2eda32 2015-09-04 12:25 +0000 [4c9f613309] Matt Jordan * ARI: Add the ability to subscribe to all events This patch adds the ability to subscribe to all events. There are two possible ways to accomplish this: (1) On initial WebSocket connection. This patch adds a new query parameter, 'subscribeAll'. If present and True, Asterisk will subscribe the applications to all ARI events. (2) Via the applications resource. When subscribing in this manner, an ARI client should merely specify a blank resource name, i.e., 'channels:' instead of 'channels:12354'. This will subscribe the application to all resources of the 'channels' type. ASTERISK-24870 #close Change-Id: I4a943b4db24442cf28bc64b24bfd541249790ad6 2015-09-21 08:16 +0000 [ec514ad64d] Elazar Broad * core/logging: Fix logging to more than one syslog channel Currently, Asterisk will log to the last configured syslog channel in logger.conf. This is due to the fact that the final call to openlog() supersedes all of the previous calls. This commit removes the call to openlog() and passes the facility to ast_log_vsyslog(), along with utilizing the LOG_MAKEPRI macro to ensure that the message is routed to the correct facility and with the correct priority. ASTERISK-25407 #close Reported by: Elazar Broad Tested by: Elazar Broad Change-Id: Ie2a2416bc00cce1b04e99ef40917c2011953ddd2 2015-09-21 18:06 +0000 [aeddee39fb] Kevin Harwell * app_record: RECORDED_FILE variable not being populated The RECORDED_FILE variable is empty unless a '%d' is specified in the filename. This patch makes it so the variable is always set to the filename. ASTERISK-25410 #close Change-Id: I4ec826d8eb582ae2ad184e717be8668b74d37653 2015-09-16 08:22 +0000 [2bd27d1222] Joshua Colp * pbx: Update device and presence state when changing a hint extension. When changing a hint extension without removing the hint first the device state and presence state is not updated. This causes the state of the hint to be that of the previous extension and not the current one. This state is kept until a state change occurs as a result of something (presence state change, device state change). This change updates the hint with the current device and presence state of the new extension when it is changed. Any state callbacks which may have been added before the hint extension is changed are also informed of the new device and presence state if either have changed. ASTERISK-25394 #close Change-Id: If268f1110290e502c73dd289c9e7e7b27bc8432f 2015-09-17 16:34 +0000 [c94f46080f] Scott Griepentrog * CHAOS: avoid crash if string create fails Validate string buffer allocation before using them. ASTERISK-25323 Change-Id: Ib9c338bdc1e53fb8b81366f0b39482b83ef56ce0 2015-09-17 04:52 +0000 [b59c4d82b5] Walter Doekes * chan_sip: Fix From header truncation for extremely long CALLERID(name). The CALLERID(num) and CALLERID(name) and other info are placed into the `char from[256]` in initreqprep. If the name was too long, the addr-spec and params wouldn't fit. Code is moved around so the addr-spec with params is placed there first, and then fitting in as much of the display-name as possible. ASTERISK-25396 #close Change-Id: I33632baf024f01b6a00f8c7f35c91e5f68c40260 2015-09-17 16:59 +0000 [4cc59533b9] Richard Mudgett * CHAOS: res_pjsip_diversion avoid crash if allocation fails Validate ast_malloc buffer returned before using it in set_redirecting_value(). ASTERISK-25323 Change-Id: I15d2ed7cb0546818264c0bf251aa40adeae83253 2015-09-17 16:47 +0000 [4fb95bbc4e] Kevin Harwell * app_queue: AgentComplete event has wrong reason When a queued caller transfers an agent to another extension sometimes the raised AgentComplete event has a reason of "caller" and sometimes "transfer". Since a transfer has taken place this should always be transfer. This occurs because sometimes the stasis hangup event arrives before the transfer event thus writing a different reason out. With this patch, when a hangup event is received during a transfer it will check to see if the channel that is hanging up is part of a transfer. If so it will return and let the subsequently received transfer event handler take care of the cleanup. ASTERISK-25399 #close Change-Id: Ic63c49bd9a5ed463ea7a032fd2ea3d63bc81a50d 2015-09-17 13:09 +0000 [fb6b5c684b] Scott Griepentrog * PJSIP: avoid crash when getting rtp peer Although unlikely, if the tech private is returned as a NULL, chan_pjsip_get_rtp_peer() would crash. ASTERISK-25323 Change-Id: Ie231369bfa7da926fb2b9fdaac228261a3152e6a 2015-09-17 11:31 +0000 [6409e7b11a] Kevin Harwell * app_queue: Crash when transferring During some transfer scenarios involving queues Asterisk would sometimes crash when trying to obtain a channel snapshot (could happen on caller or member channels). This occurred because the underlying channel had already disappeared when trying to obtain the latest snapshot. This patch adds a reference to both the member and caller channels that extends to the lifetime of the queue'd call, thus making sure the channels will always exist when retrieving the latest snapshots. ASTERISK-25185 #close Reported by: Etienne Lessard Change-Id: Ic397fa68fb4ff35fbc378e745da9246a7b552128 2015-09-16 17:36 +0000 [fe5077b1f8] Mark Michelson * res_pjsip_pubsub: Eliminate race during initial NOTIFY. There is a slim chance of a race condition occurring where two threads can both attempt to manipulate the same area. Thread A can be handling an incoming initial SUBSCRIBE request. Thread A lets the specific subscription handler know that the subscription has been established. At this point, Thread B may detect a state change on the subscribed resource and queue up a notification task on Thread C, the subscription serializer thread. Now Thread A attempts to generate the initial NOTIFY request to send to the subscriber at the same time that Thread C attempts to generate a state change NOTIFY request to send to the subscriber. The result is that Threads A and C can step on the same memory area, resulting in a crash. The crash has been observed as happening when attempting to allocate more space to hold the body for the NOTIFY. The solution presented here is to queue the subscription establishment and initial NOTIFY generation onto the subscription serializer thread (Thread C in the above scenario). This way, there is no way that a state change notification can occur before the initial NOTIFY is sent, and if there is a quick succession of NOTIFYs, we can guarantee that the two NOTIFY requests will be sent in succession. Change-Id: I5a89a77b5f2717928c54d6efb9955e5f6f5cf815 2015-08-28 15:42 +0000 [b88c54fa4b] Alexander Traud * translate: Fix transcoding while different in frame size. When Asterisk translates between codecs, each with a different frame size (for example between iLBC 30 and Speex-WB), too large frames were created by ast_trans_frameout. Now, ast_trans_frameout is called with the correct frame length, creating several frames when necessary. Affects all transcoding modules which used ast_trans_frameout: GSM, iLBC, LPC10, and Speex. ASTERISK-25353 #close Change-Id: I2e229569d73191d66a4e43fef35432db24000212 2015-09-10 17:19 +0000 [5c713fdf18] Mark Michelson * scheduler: Use queue for allocating sched IDs. It has been observed that on long-running busy systems, a scheduler context can eventually hit INT_MAX for its assigned IDs and end up overflowing into a very low negative number. When this occurs, this can result in odd behaviors, because a negative return is interpreted by callers as being a failure. However, the item actually was successfully scheduled. The result may be that a freed item remains in the scheduler, resulting in a crash at some point in the future. The scheduler can overflow because every time that an item is added to the scheduler, a counter is bumped and that counter's current value is assigned as the new item's ID. This patch introduces a new method for assigning scheduler IDs. Instead of assigning from a counter, a queue of available IDs is maintained. When assigning a new ID, an ID is pulled from the queue. When a scheduler item is released, its ID is pushed back onto the queue. This way, IDs may be reused when they become available, and the growth of ID numbers is directly related to concurrent activity within a scheduler context rather than the uptime of the system. Change-Id: I532708eef8f669d823457d7fefdad9a6078b99b2 2015-08-21 21:50 +0000 [865377fc38] Rodrigo Ramírez Norambuena * chan_sip.c: Validation on module reload Change validation on reload module because now used the cli function for reload. The sip_reload() function never fail and ever return NULL for this reason on reload() now use the call the sip_reload() and return AST_MODULE_LOAD_SUCCESS. This problem is dectected on reload by PUT method on ARI, getting always 404 http code when the module is reloaded. ASTERISK-25325 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I41215877fb2cfc589e0d4d464000cf6825f4d7fb 2015-08-21 17:39 +0000 [e75aff53e6] Richard Mudgett * res_pjsip_pubsub.c: Mark ast_sip_create_subscription() as not used. Change-Id: I2b8db18eac36c01a5c7eb9467699124e203fd093 2015-09-09 12:24 +0000 [4d91d01df1] Richard Mudgett * res_pjsip_pubsub.c: Add some notification comments. Change-Id: Ie62ff1f4b7adc1a12fa0303f53926af249b25e20 2015-08-21 18:01 +0000 [f36a9d1221] Richard Mudgett * res_pjsip_pubsub.c: Set dlg_status code instead of sending SIP response. We should not try to send a SIP response message because we may be restoring a persistent subscription where we are not responding to a SIP request. Change-Id: Id89167ef90320c5563f37e632db0dda6cb9e7dec 2015-08-21 17:40 +0000 [94582f8fab] Richard Mudgett * res_pjsip_pubsub.c: Fix off-nominal memory leak. Fix off-nominal visited vector leak in build_resource_tree(). Change-Id: If0399c7941c9c0b1038bcfb7b9a371760977831c 2015-08-21 15:26 +0000 [8b3ed52239] Richard Mudgett * res_pjsip_pubsub.c: Fix one byte buffer overrun error. ast_sip_pubsub_register_body_generator() did not account for the null terminator set by sprintf() in the allocated output buffer. Change-Id: I388688a132e479bca6ad1c19275eae0070969ae2 2015-08-21 15:25 +0000 [4329bd1e4c] Richard Mudgett * res_pjsip_pubsub.c: Use ast_alloca() instead of alloca(). Change-Id: Ia396096b4fedc2874649ca11137612c3f55e83e3 2015-08-21 11:04 +0000 [a456a20ecf] Richard Mudgett * res_pjsip_pubsub.c: Add missing error return in load_module(). Change-Id: I15debd0f717f16ee2f78e7f56151c3b3b97b72fc 2015-08-21 11:03 +0000 [f58f4c6e27] Richard Mudgett * res_pjsip/location.c: Use the builtin ao2_callback() match function instead. Change-Id: I364906d6d2bad3472929986704a0286b9a2cbe3f 2015-09-10 09:49 +0000 [9d1f176e29] Mark Michelson * res_pjsip: Copy default_from_user to avoid crash. The default_from_user retrieval function was pulling the default_from_user from the global configuration struct in an unsafe way. If using a database as a backend configuration store, the global configuration struct is short-lived, so grabbing a pointer from it results in referencing freed memory. The fix here is to copy the default_from_user value out of the global configuration struct. Thanks go to John Hardin for discovering this problem and proposing the patch on which this fix is based. ASTERISK-25390 #close Reported by Mark Michelson Change-Id: I6b96067a495c1259da768f4012d44e03e7c6148c 2015-09-10 08:39 +0000 [1dd0e220bf] Matt Jordan * res/res_pjsip_nat: Ignore REGISTER requests when looking for a Record-Route We will only rewrite the Contact header if there is no Record-Route header in the received request. If a malfunctioning proxy places a Record-Route header into a REGISTER request, we will decide that we shouldn't update the IP/port in the Contact header, and we will end up storing a contact with an AoR that contains the NAT'd IP address. While it is nice to have the proxy *not* send a Record-Route in a REGISTER request, it's also a good idea to not process the header in a non-dialog message. This patch updates the code to explicitly ignore the Record-Route header in REGISTER requests. ASTERISK-25387 #close Change-Id: I4bd3bcccc4003d460cc354d986b0dea2e433ef3f 2015-09-03 21:15 +0000 [4eedd9ef9d] Matt Jordan * main/config_options: Check for existance of internal object before derefing Asterisk can load and register an object type while still having an invalid sorcery mapping. This can cause an issue when a creation call is invoked. For example, mis-configuring PJSIP's endpoint identifier by IP address mapping in sorcery.conf will cause the sorcery mechanism to be invalidated; however, a subsequent ARI invocation to create the object will cause a crash, as the internal type may not be registered as sorcery expects. Merely checking for a NULL pointer here solves the issue. Change-Id: I54079fb94a1440992f4735a9a1bbf1abb1c601ac 2015-09-09 16:46 +0000 [71408df2b8] Alexander Anikin * chan_ooh323: Add ProgressIndicator IE with inband info available Add ProgressIndicator IE with inband info present to Progress and Alerting Q.931 message ASTERISK-25227 #close Reported by: Alexandr Dranchuk Change-Id: I326ad13cb1db9a72b3fd902bafed3c28a3684203 2015-09-08 10:35 +0000 [f72f9ceefc] Scott Griepentrog * pjsip: avoid possible crash req_caps allocation failure Make certain that the pjsip session has not failed to allocate the format capabilities structure, which can otherwise cause a crash when referenced. ASTERISK-25323 Change-Id: I602790ba12714741165e441cc64a3ecde4cb5750 2015-09-03 14:07 +0000 [fbf720db91] Jonathan Rose * ParkAndAnnounce: Add variable inheritance In Asterisk 11, the announcer channel would receive channel variables from the channel being parked by means of normal channel inheritance. This functionality was lost during the big res_parking project in Asterisk 12. This patch restores that functionality. ASTERISK-25369 #close Review: https://gerrit.asterisk.org/#/c/1180/ Change-Id: Ie47e618330114ad2ea91e2edcef1cb6f341eed6e 2015-09-04 16:33 +0000 [695f26cbb7] David M. Lee * res_rtp_asterisk: Add more ICE debugging In working through a recent ICE negotiation bug, I found the debug logging in res_rtp_asterisk to be lacking. This patch adds a number of debug and warning statements that were helpful. Change-Id: I950c6d8f13a41f14b3d6334b4cafe7d4e997be80 2015-09-01 10:16 +0000 [4ed9c9a280] Guido Falsi * Core/General: Add #ifdef needed on FreeBSD. pthread_attr_init() defaults to PTHREAD_EXPLICIT_SCHED on FreeBSD too. ASTERISK-25310 #close Reported by: Guido Falsi Change-Id: Iae6befac9028b5b9795f86986a4a08a1ae6ab7c4 2015-09-08 07:21 +0000 [5469caa9dd] Joshua Colp * res_pjsip: Use hash for contact object identity instead of Contact URI. In the wild it is possible for Contact URIs to be quite long as parameters can exist on them. This can present a problem when storing them in the AstDB as the URI is used as part of the object name and there is a fixed length limit for the AstDB. This will cause the contact to not get stored. This change uses the MD5 hash of the Contact URI as part of the object name instead. This has a fixed length which is guaranteed to not exceed the AstDB length limit. ASTERISK-25295 #close Change-Id: Ie8252a75331ca00b41b9f308f42cc1fbdf701a02 2015-09-07 13:19 +0000 [480c443e26] Alexander Anikin * chan_ooh323: call ast_rtp_instance_stop on ooh323_destroy Call ast_rtp_instance_stop on ooh323_destroy to free resources allocated by rtp instance ASTERISK-25299 #close Report by: Alexandr Dranchuk Change-Id: I455096bd7da016b871afe90af86067c2c7c9f33f 2015-09-07 11:15 +0000 [c3e6debdb9] Matt Jordan * res/res_pjsip: Purge contacts when an AoR is deleted When an AoR is deleted by an external mechanism, such as through ARI, we currently do not remove dynamic contacts that were created for that AoR as a result of a received REGISTER request. As a result, re-creating the AoR will cause the dynamic contact to be interpreted as a persistent contact, leading to some rather strange state being created for the contacts/endpoints. This patch adds a sorcery observer for the 'aor' object. When a delete is issued on the underlying sorcery object, the observer is called, and all contacts created and persisted in sorcery for that AoR are also removed. Note that we don't want to perform this action when an AO2 object that is an AoR is destroyed, as the AoR can still exist in the backing storage (and we would thus be removing valid contacts from an AoR that still "exists".) ASTERISK-25381 #close Change-Id: I6697e51ef6b2858b5d63401f35dc378bb0f90328 2015-09-05 14:58 +0000 [78d0b9d97e] Matt Jordan * channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a 2015-09-04 16:06 +0000 [61c6c6aa6c] David M. Lee * Fix when remote candidates exceed PJ_ICE_MAX_CAND We were passing the wrong count into pj_ice_sess_create_check_list(), causing the create to fail if we ever received more than PJ_ICE_MAX_CAND candidates. Change-Id: I0303d8e1ecb20a8de9fe629a3209d216c4028378 2015-09-04 14:40 +0000 [ac62928d6b] Mark Michelson * res_pjsip: Change default from user value. When Asterisk sends an outbound SIP request, if there is no direct reason to place a specific value for the username in the From header, Asterisk would generate a UUID. For example, this would happen when sending outbound OPTIONS requests when qualifying or when sending outbound INVITE requests when originating (if no explicit caller ID were provided). The issue is that some SIP providers reject these sorts of requests with a "Name too long" error response. This patch aims to fix this by changing the default outbound username in From headers to "asterisk". This value can be overridden by changing the default_from_user option in the global options if desired. ASTERISK-25377 #close Reported by Mark Michelson Change-Id: I6a4d34a56ff73ff4f661b0075aeba5461b7f3190 2015-09-04 09:26 +0000 [6002472a62] Scott Griepentrog * endpoint snapshot: avoid second cleanup on alloc failure In ast_endpoint_snapshot_create(), a failure to init the string fields results in two attempts to ao2_cleanup the same pointer. Removed RAII_VAR to eliminate problem. ASTERISK-25375 #close Reported by: Scott Griepentrog Change-Id: If4d9dfb1bbe3836b623642ec690b6d49b25e8979 2015-09-04 05:33 +0000 [d32e516c7c] Martin Tomec * res/pjsip: Mark WSS transport as secure Pjsip is refusing to use unsecure transport with "sips" in url. WSS should be considered as secure transport. ASTERISK-24602 #comment Partially fixed by setting WSS as secure Change-Id: Iddac406c6deba6240c41a603b8859dfefe1a5353 2015-09-02 17:26 +0000 [ad9cb6c2ce] Mark Michelson * res_pjsip: Fix contact refleak on stateful responses. When sending a stateful response, creation of the transaction can fail, most commonly because we are trying to create a transaction from a retransmitted request. When creation of the transaction fails, we end up leaking a reference to a contact that was bumped when the response was created. This patch adds the missing deref and fixes the reference leak. Change-Id: I2f97ad512aeb1b17e87ca29ae0abacb4d6395f07 2015-09-02 12:41 +0000 [cc1363209e] Joshua Colp * pbx: Fix crash when issuing "core show hints" with long pattern match. When issuing the "core show hints" CLI command a combination of both the hint extension and context is created. This uses a fixed size buffer expecting that the extension will not exceed maximum extension length. When the extension is actually a pattern match this constraint does not hold true, and the extension may exceed the maximum extension length. In this case extra characters are written past the end of the fixed size buffer. This change makes it so the construction of the combined hint extension and context can not exceed the size of the buffer. ASTERISK-25367 #close Change-Id: Idfa1b95d0d4dc38e675be7c1de8900b3f981f499 2015-09-01 09:05 +0000 [d58c8d73af] Mark Michelson * res_pjsip_pubsub: re-re-fix persistent subscription storage. A recent change to res_pjsip_pubsub switched to using pjsip_msg_print as a means of writing an appropriate packet to persistent storage. While this partially solved the issue, it had its own problems. pjsip_msg_print will always add a Content-Length header to the message it prints. Frequent restarts of Asterisk can result in persistent subscriptions being written with five or more Content-Length headers. In addition, sometimes some apparent corruption of individual headers could be seen. This aims to fix the problem by not running a parsed message through an interpreter but rather by taking the raw message and saving it. The logic for what to save is going to be different depending on whether a SUBSCRIBE was received from the wire or if it was pulled from persistence. When receiving a packet from the wire, when using a streaming transport, the rdata->pkt_info.packet may contain multiple SIP messages or fragments. However, the rdata->msg_info.msg_buf will always contain the current SIP message to be processed. When pulling from persistence, though, the rdata->msg_info.msg_buf will be NULL since no transport actually handled the packet. However, since we know that we will always ever pull one SIP message from persistence, we are free to save directly from rdata->pkt_info.packet instead. ASTERISK-25365 #close Reported by Mark Michelson Change-Id: I33153b10d0b4dc8e3801aaaee2f48173b867855b 2015-08-31 15:24 +0000 [03fe79f29e] Mark Michelson * Fix deadlock on presence state changes. A deadlock was observed where three threads were competing for different locks: * One thread held the hints lock and was attempting to lock a specific hint. * One thread was holding the specific hint's lock and was attempting to lock the contexts lock * One thread was holding the contexts lock and attempting to lock the hints lock. Clearly the second thread was doing the wrong thing here. The fix for this is to make sure that the hint's lock is not held on presence state changes. Something similar is already done (and commented about) for device state changes. ASTERISK-25362 #close Reported by Mark Michelson Change-Id: I15ec2416b92978a4c0c08273b2d46cb21aff97e2 2015-08-29 10:36 +0000 [a676ba2aad] Joshua Colp * taskprocessor: Fix race condition between unreferencing and finding. When unreferencing a taskprocessor its reference count is checked to determine if it should be unlinked from the taskprocessors container and its listener shut down. In between the time when the reference count is checked and unlinking it is possible for another thread to jump in, find it, and get a reference to it. If the thread then uses the taskprocessor it may find that it is not in the state it expects. This change locks the taskprocessors container during almost the entire unreference operation to ensure that any other thread which may attempt to find the taskprocessor has to wait. ASTERISK-25295 Change-Id: Icb842db82fe1cf238da55df92e95938a4419377c 2015-08-28 20:22 +0000 [1b1561f4c8] Joshua Colp * res_pjsip_sdp_rtp: Fix multiple keepalive scheduled items. The keepalive support in res_pjsip_sdp_rtp currently assumes that a stream will only be negotiated once. This is false. If the stream is replaced and later added back it can be negotiated again causing multiple keepalive scheduled items to exist. This change explicitly deletes the existing keepalive scheduled item before adding the new one. The res_pjsip_sdp_rtp module also does not stop RTP keepalives or timeout timer if the stream has been replaced. This change adds a callback to the session media interface to allow a media stream to be stopped without the resources being destroyed. This allows the scheduled items and RTP to be stopped when the stream no longer exists. ASTERISK-25356 #close Change-Id: Ibe6a7cc0927c87326fd5f1c0d4ad889dbfbea1de 2015-08-28 19:57 +0000 [85e1cb51b2] Joshua Colp * sched: ast_sched_del may return prematurely due to spurious wakeup When deleting a scheduled item if the item in question is currently executing the ast_sched_del function waits until it has completed. This is accomplished using ast_cond_wait. Unfortunately the ast_cond_wait function can suffer from spurious wakeups so the predicate needs to be checked after it returns to make sure it has really woken up as a result of being signaled. This change adds a loop around the ast_cond_wait to make sure that it only exits when the executing task has really completed. ASTERISK-25355 #close Change-Id: I51198270eb0b637c956c61aa409f46283432be61 2015-08-27 12:26 +0000 [c2c7319082] Joshua Colp * res_pjsip_session: Don't invoke session supplements twice for BYE requests. When a BYE request is received the PJSIP invite session implementation creates and sends a 200 OK response before we are aware of it. This causes the INVITE session state callback to be called into and ultimately the session supplements run on the BYE request. Once this response has been sent the normal transaction state callback is invoked which invokes the session supplements on the BYE request again. This can be problematic in particular with res_pjsip_rfc3326 as it may attempt to update the hangup cause code on the channel while it is in the process of being hung up. This change makes it so the session supplements are only invoked once by the INVITE session state callback. ASTERISK-25318 #close Change-Id: I69c17df55ccbb61ef779ac38cc8c6b411376c19a 2015-08-26 15:26 +0000 [6862c2a167] Scott Griepentrog * Chaos: handle failed allocation in get_media_encryption_type If the ast_strndup() call fails to allocate a copy of the transport string for parsing, fail gracefully. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: Ia4b905ce6d03da53fea526224455c1044b1a5a28 2015-08-26 14:25 +0000 [f1cd636658] Scott Griepentrog * Chaos: make hangup NULL tolerant In chan_pjsip_new, if allocation of the pvt structure fails, ast_hangup is called. But it was written to assume pvt was valid, and this change corrects that. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I5f47860fe9cee4cd56abd3f79b108678ab72cc87 2015-08-26 05:40 +0000 [c01111223f] Joshua Colp * chan_sip: Allow call pickup to set the hangup cause. The call pickup implementation in chan_sip currently sets the channel hangup cause to "normal clearing" if call pickup is successfully performed. This action overwrites the "answered elsewhere" hangup cause set by the call pickup code and can result in the SIP device in question showing a missed call when it should not. This change sets the hangup cause to "normal clearing" as a default initially but allows the call pickup to change it as needed. ASTERISK-25346 #close Change-Id: I00ac2c269cee9e29586ee2c65e83c70e52a02cff 2015-08-25 07:17 +0000 [2a4eee0cd9] Joshua Colp * res_pjsip: Add common ast_sip_get_host_ip API. Modules commonly used the pj_gethostip function for retrieving the IP address of the host. This function does not cache the result and may result in a DNS lookup occurring, or additional work. If the DNS server is unreachable or network issues arise this can cause the pj_gethostip function to block for a period of time. This change adds an ast_sip_get_host_ip and ast_sip_get_host_ip_string function which does the same thing but caches the host IP address at module load time. This results in no additional work being done each time the local host IP address is needed. ASTERISK-25342 #close Change-Id: I3205deb679b01fa5ac05a94b623bfd620a2abe1e 2015-08-24 11:04 +0000 [7c4d0c3506] Joshua Colp * res_pjsip_pubsub: On recreated notify fail deleted sub_tree is referenced When recreating a subscription it is possible for a freed sub_tree to be referenced when the initial NOTIFY fails to be created. Change-Id: I681c215309aad01b21d611c2de47b3b0a6022788 2015-08-24 06:21 +0000 [6c2dab1e88] Joshua Colp * bridge: Kick channel from bridge if hung up during action. When executing an action in a bridge it is possible for the channel to be hung up without the bridge becoming aware of it. This is most easily reproducible by hanging up when the bridge is streaming DTMF due to a feature timeout. This change makes it so after action execution the channel is checked to determine if it has been hung up and if it has it is kicked from the bridge. ASTERISK-25341 #close Change-Id: I6dd8b0c3f5888da1c57afed9e8a802ae0a053062 2015-08-23 18:26 +0000 [bc6fe07f5c] Matt Jordan * res_pjsip/pjsip_configuration: Disregard empty auth values When an endpoint is backed by a non-static conf file backend (such as the AstDB or Realtime), the 'auth' object may be returned as being an empty string. Currently, res_pjsip will interpret that as being a valid auth object, and will attempt to authenticate inbound requests. This isn't desired; is an auth value is empty (which the name of an auth object cannot be), we should instead interpret that as being an invalid auth object and skip it. ASTERISK-25339 #close Change-Id: Ic32b0c6eb5575107d5164a8c40099e687cd722c7 2015-08-19 12:10 +0000 [0582776f7f] Richard Mudgett * ari/ari_websockets.c: Fix ast_debug parameter type mismatch. This is a type mismatch fix of the debugging commit c63316eec10e1990a88bf4712238d6deb375bfa9 made to find out why a testsuite test was failing only on one of the continuous integration build agents. Change-Id: Iba34f6e87cec331f6ac80e4daff6476ea6f00a75 2015-08-19 10:30 +0000 [504213f542] Scott Griepentrog * contrib: script install_prereq should install sqlite3 Asterisk needs the sqlite 3 library, which is package sqlite-devel in CentOS. By adding this package to the script, a problem with configure failing is resolved. ASTERISK-25331 #close Reported by: Kevin Harwell Change-Id: I90efaf6a01914fea03f21e5cdbd91c348f44b0ec 2015-08-18 16:06 +0000 [77518d5434] Richard Mudgett * res_http_websocket.c: Fix some off nominal path cleanup. * Remove extraneous unlock on off-nominal path. * Add missing HTTP error reply. Change-Id: I1f402bfe448fba8696b507477cab5f060ccd9b2b 2015-08-18 14:46 +0000 [c61547fee6] Richard Mudgett * res_ari.c: Add missing off nominal unlock and remove a RAII_VAR(). Change-Id: I0c5e7b34057f26dadb39489c4dac3015c52f5dbf 2015-08-17 16:41 +0000 [bd867cd078] Richard Mudgett * app_queue.c: Extract some functions for simpler code. * Extract set_queue_member_pause() from set_member_paused() for simpler and more consistent code. * Extract set_queue_member_ringinuse() from set_member_ringinuse_help_members() for simpler code. Change-Id: Iecc1f4119c63347341d7ea6b65f5fc4963706306 2015-08-14 12:55 +0000 [e5f5b9f384] Richard Mudgett * app_queue.c: Fix setting QUEUE_MEMBER 'paused' and 'ringinuse'. Setting the 'paused' and 'ringinuse' options on a queue member using the dialplan function QUEUE_MEMBER did not behave the same way as the equivalent dialplan applications or AMI actions. * Made queue_function_mem_write() call the set_member_paused() and set_member_value() for the 'paused' and 'ringinuse' options respectively. A beneficial side effect is that the queue name is now optional and sets the value in all queues the interface is a member. * Update QUEUE_MEMBER XML documentation. * Fix error checking in QUEUE_MEMBER() write. ASTERISK-25215 #close Reported by: Lorne Gaetz Change-Id: I3a016be8dc94d63a9cc155295ff9c9afa5f707cb 2015-08-17 13:34 +0000 [ded51e3d77] Richard Mudgett * app_queue.c: Fix error checking in QUEUE_MEMBER() read. Change-Id: I7294e13d27875851c2f4ef6818adba507509d224 2015-08-17 11:00 +0000 [ab373f2cef] Scott Griepentrog * CHAOS: prevent sorcery object with null id When allocating a sorcery object, fail if the id value was not allocated. ASTERISK-25323 Reported by: Scott Griepentrog Change-Id: I152133fb7545a4efcf7a0080ada77332d038669e 2015-08-14 15:46 +0000 [b719f56c72] Mark Michelson * res_pjsip_sdp_rtp: Restore removed NULL check. When sending an RTP keepalive, we need to be sure we're not dealing with a NULL RTP instance. There had been a NULL check, but the commit that added the rtp_timeout and rtp_hold_timeout options removed the NULL check. Change-Id: I2d7dcd5022697cfc6bf3d9e19245419078e79b64 2015-08-13 12:30 +0000 [cea5dc7b8a] Richard Mudgett * audiohook.c: Simplify variable usage in audiohook_read_frame_both(). Change-Id: I58bed58631a94295b267991c5b61a3a93c167f0c 2015-08-13 12:22 +0000 [b3a56bee83] Richard Mudgett * audiohook.c: Fix MixMonitor crash when using the r() or t() options. The built frame format in audiohook_read_frame_both() is now set to a signed linear format before the rx and tx frames are duplicated instead of only for the mixed audio frame duplication. ASTERISK-25322 #close Reported by Sean Pimental Change-Id: I86f85b5c48c49e4e2d3b770797b9d484250a1538 2015-08-12 12:59 +0000 [25af2d71c8] Kevin Harwell * chan_sip.c: wrong peer searched in sip_report_security_event In chan_sip, after handling an incoming invite a security event is raised describing authorization (success, failure, etc...). However, it was doing a lookup of the peer by extension. This is fine for register messages, but in the case of an invite it may search and find the wrong peer, or a non existent one (for instance, in the case of call pickup). Also, if the peers are configured through realtime this may cause an unnecessary database lookup when caching is enabled. This patch makes it so that sip_report_security_event searches by IP address when looking for a peer instead of by extension after an invite is processed. ASTERISK-25320 #close Change-Id: I9b3f11549efb475b6561c64f0e6da1a481d98bc4 2015-08-13 05:26 +0000 [e18c300550] Joshua Colp * res_http_websocket: When shutting down a session don't close closed socket Due to the use of ast_websocket_close in session termination it is possible for the underlying socket to already be closed when the session is terminated. This occurs when the close frame is attempted to be written out but fails. Change-Id: I7572583529a42a7dc911ea77a974d8307d5c0c8b 2015-08-11 05:24 +0000 [b4e9416138] Joshua Colp * res_http_websocket: Forcefully terminate on write errors. The res_http_websocket module will currently attempt to close the WebSocket connection if fatal cases occur, such as when attempting to write out data and being unable to. When the fatal cases occur the code attempts to write a WebSocket close frame out to have the remote side close the connection. If writing this fails then the connection is not terminated. This change forcefully terminates the connection if the WebSocket is to be closed but is unable to send the close frame. ASTERISK-25312 #close Change-Id: I10973086671cc192a76424060d9ec8e688602845 2015-08-10 13:43 +0000 [256bc52b66] Richard Mudgett * chan_dahdi.c: Flush the DAHDI write buffer after starting DTMF. Pressing DTMF digits on a phone to go out on a DAHDI channel can result in the digit not being recognized or even heard by the peer. Phone -> Asterisk -> DAHDI/channel Turns out the DAHDI behavior with DTMF generation (and any other generated tones) is exposed by the "buffers=" setting in chan_dahdi.conf. When Asterisk requests to start sending DTMF then DAHDI waits until its write buffer is empty before generating any samples for the DTMF tones. When Asterisk subsequently requests DAHDI to stop sending DTMF then DAHDI immediately stops generating the DTMF samples. As a result, the more samples there are in the DAHDI write buffer the shorter the time DTMF actually gets sent on the wire. If there are more samples in the write buffer than the time DTMF is supposed to be sent then no DTMF gets sent on the wire. With the "buffers=12,half" setting and each buffer representing 20 ms of samples then the DAHDI write buffer is going to contain around 120 ms of samples. For DTMF to be recognized by the peer the actual sent DTMF duration needs to be a minimum of 40 ms. Therefore, the intended duration needs to be a minimum of 160 ms for the peer to receive the minimum DTMF digit duration to recognize it. A simple and effective solution to work around the DAHDI behavior is for Asterisk to flush the DAHDI write buffer when sending DTMF so the full duration of DTMF is actually sent on the wire. When someone is going to send DTMF they are not likely to be talking before sending the tones so the flushed write samples are expected to just contain silence. * Made dahdi_digit_begin() flush the DAHDI write buffer after requesting to send a DTMF digit. ASTERISK-25315 #close Reported by John Hardin Change-Id: Ib56262c708cb7858082156bfc70ebd0a220efa6a 2015-08-05 14:21 +0000 [800e0ea48d] Richard Mudgett * chan_dahdi.c: Lock private struct for ast_write(). There is a window of opportunity for DTMF to not go out if an audio frame is in the process of being written to DAHDI while another thread starts sending DTMF. The thread sending the audio frame could be past the currently dialing check before being preempted by another thread starting a DTMF generation request. When the thread sending the audio frame resumes it will then cause DAHDI to stop the DTMF tone generation. The result is no DTMF goes out. * Made dahdi_write() lock the private struct before writing to the DAHDI file descriptor. ASTERISK-25315 Reported by John Hardin Change-Id: Ib4e0264cf63305ed5da701188447668e72ec9abb 2015-08-10 18:23 +0000 [c126afe18f] Richard Mudgett * res_pjsip.c: Fix crash from corrupt saved SUBSCRIBE message. If the saved SUBSCRIBE message is not parseable for whatever reason then Asterisk could crash when libpjsip tries to parse the message and adds an error message to the parse error list. * Made ast_sip_create_rdata() initialize the parse error rdata list. The list is checked after parsing to see that it remains empty for the function to return successful. ASTERISK-25306 Reported by Mark Michelson Change-Id: Ie0677f69f707503b1a37df18723bd59418085256 2015-08-10 07:40 +0000 [f68c995bc9] Alexander Traud * chan_sip: Fix negotiation of iLBC 30. iLBC 20 was advertised in a SIP/SDP negotiation. However, only iLBC 30 is supported. Removes "a=fmtp:x mode=y" from SDP. Because of RFC 3952 section 5, only iLBC 30 is negotiated now. ASTERISK-25309 #close Change-Id: I92d724600a183eec3114da0ac607b994b1a793da 2015-08-09 18:42 +0000 [8e194047ac] Matt Jordan * res/res_format_attr_silk: Expose format attributes to other modules This patch adds the .get callback to the format attribute module, such that the Asterisk core or other third party modules can query for the negotiated format attributes. Change-Id: Ia24f55cf9b661d651ce89b4f4b023d921380f19c 2015-08-09 17:56 +0000 [a0f451c35e] Matt Jordan * main/format: Add an API call for retrieving format attributes Some codecs that may be a third party library to Asterisk need to have knowledge of the format attributes that were negotiated. Unfortunately, when the great format migration of Asterisk 13 occurred, that ability was lost. This patch adds an API call, ast_format_attribute_get, to the core format API, along with updates to the unit test to check the new API call. A new callback is also now available for format attribute modules, such that they can provide the format attribute values they manage. Note that the API returns a void *. This is done as the format attribute modules themselves may store format attributes in any particular manner they like. Care should be taken by consumers of the API to check the return value before casting and dereferencing. Consumers will obviously need to have a priori knowledge of the type of the format attribute as well. Change-Id: Ieec76883dfb46ecd7aff3dc81a52c81f4dc1b9e3 2015-08-07 22:11 +0000 [26f0559a94] David M. Lee * Replace htobe64 with htonll We don't have a compatability function to fill in a missing htobe64; but we already have one for the identical htonll. Change-Id: Ic0a95db1c5b0041e14e6b127432fb533b97e4cac 2015-08-07 14:20 +0000 [df9ce36366] Scott Emidy * ARI: Retrieve existing log channels An http request can be sent to get the existing Asterisk logs. The command "curl -v -u user:pass -X GET 'http://localhost:8088 /ari/asterisk/logging'" can be run in the terminal to access the newly implemented functionality. * Retrieve all existing log channels ASTERISK-25252 Change-Id: I7bb08b93e3b938c991f3f56cc5d188654768a808 2015-08-07 11:14 +0000 [e9f1bc08cb] Scott Emidy * ARI: Creating log channels An http request can be sent to create a log channel in Asterisk. The command "curl -v -u user:pass -X POST 'http://localhost:088/ari/asterisk/logging/mylog? configuration=notice,warning'" can be run in the terminal to access the newly implemented functionality for ARI. * Ability to create log channels using ARI ASTERISK-25252 Change-Id: I9a20e5c75716dfbb6b62fd3474faf55be20bd782 2015-08-06 15:18 +0000 [78364132ce] Scott Emidy * ARI: Deleting log channels An http request can be sent to delete a log channel in Asterisk. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/logging/mylog'" can be run in the terminal to access the newly implemented functionally for ARI. * Able to delete log channels using ARI ASTERISK-25252 Change-Id: Id6eeb54ebcc511595f0418d586ff55914bc3aae6 2015-08-06 12:48 +0000 [e25569ef95] Mark Michelson * res_pjsip_pubsub: More accurately persist packet. The pjsip_rx_data structure has a pkt_info.packet field on it that is the packet that was read from the transport. For datagram transports, the packet read from the transport will correspond to the SIP message that arrived. For streamed transports, however, it is possible to read multiple SIP messages in one packet. In a recent case, Asterisk crashed on a system where TCP was being used. This is because at some point, a read from the TCP socket resulted in a 200 OK response as well as an incoming SUBSCRIBE request being stored in rdata->pkt_info.packet. When the SUBSCRIBE was processed, the combination 200 OK and SUBSCRIBE was saved in persistent storage. Later, a restart of Asterisk resulted in the crash because the persistent subscription recreation code ended up building the 200 OK response instead of a SUBSCRIBE request, and we attempted to access request-specific data. The fix here is to use the pjsip_msg_print() function in order to persist SUBSCRIBE requests. This way, rather than using the raw socket data, we use the parsed SIP message that PJSIP has given us. If we receive multiple SIP messages from a single read, we will be sure only to save off the relevant SIP message. There also is a safeguard put in place to make sure that if we do end up reconstructing a SIP response, it will not cause a crash. ASTERISK-25306 #close Reported by Mark Michelson Change-Id: I4bf16f7b76a2541d10b55de82bcd14c6e542afb2 2015-08-04 16:12 +0000 [8521a86367] Joshua Colp * res_pjsip: Ensure sanitized XML is NULL terminated. The ast_sip_sanitize_xml function is used to sanitize a string for placement into XML. This is done by examining an input string and then appending values to an output buffer. The function used by its implementation, strncat, has specific behavior that was not taken into account. If the size of the input string exceeded the available output buffer size it was possible for the sanitization function to write past the output buffer itself causing a crash. The crash would either occur because it was writing into memory it shouldn't be or because the resulting string was not NULL terminated. This change keeps count of how much remaining space is available in the output buffer for text and only allows strncat to use that amount. Since this was exposed by the res_pjsip_pidf_digium_body_supplement module attempting to send a large message the maximum allowed message size has also been increased in it. A unit test has also been added which confirms that the ast_sip_sanitize_xml function is providing NULL terminated output even when the input length exceeds the output buffer size. ASTERISK-25304 #close Change-Id: I743dd9809d3e13d722df1b0509dfe34621398302 2015-08-05 05:23 +0000 [9a12804e59] Joshua Colp * res_rtp_asterisk: Don't leak temporary key when enabling PFS. A change recently went in which enabled perfect forward secrecy for DTLS in res_rtp_asterisk. This was accomplished two different ways depending on the availability of a feature in OpenSSL. The fallback method created a temporary instance of a key but did not free it. This change fixes that. ASTERISK-25265 Change-Id: Iadc031b67a91410bbefb17ffb4218d615d051396 2015-08-04 09:47 +0000 [27dc2094e9] Mark Michelson * res_http_websocket: Debug write lengths. Commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee attempted to fix a test failure observed on 32 bit test agents by ensuring that a cast from a 32 bit unsigned integer to a 64 bit unsigned integer was happening in a predictable place. As it turns out, this did not cause test runs to succeed. This commit adds several redundant debug messages that print the payload lengths of websocket frames. The idea here is that this commit will not cause tests to succeed for the faulty test agent, but we might deduce where the fault lies more easily this way by observing at what point the expected value (537) changes to some ungangly huge number. If you are wondering why something like this is being committed to the branch, keep in mind that in commit 39cc28f6ea2140ad6d561fd4c9e9a66f065cecee I noted that the observed test failures only happen when automated tests are run. Attempts to run the tests by hand manually on the test agent result in the tests passing. Change-Id: I14a65c19d8af40dadcdbd52348de3b0016e1ae8d 2015-08-03 11:06 +0000 [39cc28f6ea] Mark Michelson * res_http_websocket: Avoid passing strlen() to ast_websocket_write(). We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67 2015-07-28 05:33 +0000 [aed068844c] Mark Duncan * res/res_rtp_asterisk: Add ECDH support This will add ECDH support to Asterisk. It will detect auto ECDH support in OpenSSL (1.0.2b and above) during ./configure. If this is available, it will use it, otherwise it will fall back to prime256v1 (this behavior is consistent with other projects such as Apache and nginx). This fixes WebRTC being broken in Firefox 38+ due to Firefox now only supporting ciphers with perfect forward secrecy. ASTERISK-25265 #close Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b 2015-07-29 14:17 +0000 [1ae762634c] Benjamin Ford * ARI: Rotate log channels. An http request can be sent to rotate a specified log channel. If the channel does not exist, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/logging/logChannelName/rotate'" can be run in the terminal to access this new functionality. * Added the ability to rotate log files through ARI ASTERISK-25252 Change-Id: Iaefa21cbbc1b29effb33004ee3d89c977e76ab01 2015-07-29 13:49 +0000 [aeeb170fc4] Richard Mudgett * rtp_engine.c: Fix performance issue with several channel drivers that use RTP. ast_rtp_codecs_get_payload() gets called once or twice for every received RTP frame so it would be nice to not allocate an ao2 object to then have it destroyed shortly thereafter. The ao2 object gets allocated only if the payload type is not set by the channel driver as a negotiated value. The issue affects chan_skinny, chan_unistim, chan_rtp, and chan_ooh323. * Made static_RTP_PT[] an array of ao2 objects that ast_rtp_codecs_get_payload() can return instead of an array of structs that must be copied into a created ao2 object. ASTERISK-25296 #close Reported by: Richard Mudgett Change-Id: Icb6de5cd90bfae07d44403a1352963db9109dac0 2015-07-29 17:00 +0000 [84262749d2] Richard Mudgett * res_rtp_asterisk.c: Fix off-nominal crash potential. ASTERISK-25296 Reported by: Richard Mudgett Change-Id: I08549fb7c3ab40a559f41a3940f3732a4059b55b 2015-07-29 13:48 +0000 [1519eb44a7] Richard Mudgett * rtp_engine.c: Must protect mime_types_len with mime_types_lock. Change-Id: I44220dd369cc151ebf5281d5119d84bb9e54d54e 2015-07-24 18:42 +0000 [a93b7a927c] Richard Mudgett * res_pjsip_sdp_rtp.c: Fix processing wrong SDP media list. Change-Id: I7c076826c2d3c6ae8c923ca73b7a71980cca11f2 2015-07-24 18:38 +0000 [741fa0d26d] Richard Mudgett * res_pjsip_sdp_rtp.c: Fixup some whitespace. Change-Id: Ib4eb7ef7dcaf93ddc26538f0a498aaf110d7a973 2015-07-27 19:10 +0000 [89b21fd9a3] Richard Mudgett * rtp_engine.h: No sense allowing payload types larger than RFC allows. * Tweaked add_static_payload() to not use magic numbers. Change-Id: I1719ff0f6d3ce537a91572501eae5bcd912a420b 2015-07-23 14:04 +0000 [7427c7f13b] Richard Mudgett * rtp_engine.c: Minor tweaks. * Fix off nominial ref leak of new_type in ast_rtp_codecs_payloads_set_m_type(). * No need to lock static_RTP_PT_lock in ast_rtp_codecs_payloads_set_m_type() and ast_rtp_codecs_payloads_set_rtpmap_type_rate() before the payload type parameter sanity check. * No need to create ast_rtp_payload_type ao2 objects with a lock since the lock is not used. Change-Id: I64dd1bb4dfabdc7e981e3f61448beac9bb7504d4 2015-07-23 12:41 +0000 [e20f435b60] Richard Mudgett * rtp_engine.h: Misc comment fixes. Change-Id: If98139264d5d97427b4685ecbdc54518f725bc43 2015-07-17 16:23 +0000 [bc5d7f9c37] Richard Mudgett * chan_sip.c: Tweak glue->update_peer() parameter nil value. Change glue->update_peer() parameter from 0 to NULL to better indicate it is a pointer. Change-Id: I8ff2e5087f0e19f6998e3488a712a2470cc823bd 2015-07-30 17:05 +0000 [13eb491e35] Richard Mudgett * res_pjsip_session.c: Fix crashes seen when call cancelled. Two testsuite tests crashed in the same place as a result of an INVITE being CANCELed. tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp The session pointer is no longer in the inv->mod_data[session_module.id] location because the INVITE transaction has reached the terminated state. ASTERISK-25297 #close Reported by: Richard Mudgett Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427 2015-07-29 14:35 +0000 [48698a5e21] Mark Michelson * res_http_websocket: Properly encode 64 bit payload A test agent was continuously failing all ARI tests when run against Asterisk 13. As it turns out, the reason for this is that on those test runs, for some reason we decided to use the super extended 64 bit payload length for websocket text frames instead of the extended 16 bit payload length. For 64-bit payloads, the expected byte order over the network is 7, 6, 5, 4, 3, 2, 1, 0 However, we were sending the payload as 3, 2, 1, 0, 7, 6, 5, 4 This meant that we were saying to expect an absolutely MASSIVE payload to arrive. Since we did not follow through on this expected payload size, the client would sit patiently waiting for the rest of the payload to arrive until the test would time out. With this change, we use the htobe64() function instead of htonl() so that a 64-bit byte-swap is performed instead of a 32 bit byte-swap. Change-Id: Ibcd8552392845fbcdd017a8c8c1043b7fe35964a 2015-07-29 12:23 +0000 [10ba72a927] Mark Michelson * Add a test event for inband ringing. This event is necessary for the bridge_wait_e_options test to be able to confirm that ringing is being played on the local channel that runs the BridgeWait() application with the e(r) option. ASTERISK-25292 #close Reported by Kevin Harwell Change-Id: Ifd3d3d2bebc73344d4b5310d0d55c7675359d72e 2015-07-16 12:16 +0000 [8458b8d441] Jonathan Rose * holding_bridge: ensure moh participants get frames Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 2015-07-24 12:56 +0000 [f78a4b52b8] Matt Jordan * Bump the ARI version to 1.8.0 Due to backwards compatible changes, the ARI version should be bumped to 1.8.0 prior to the release of 13.5.0. Note that a previous patch already bumped the version of AMI for this release. Change-Id: I419033bfbbc0d3533a29ccb32b2981f39e0883e7 2015-07-18 11:16 +0000 [2749721791] Joshua Colp * pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9 2015-07-24 09:46 +0000 [b4e19e414a] Mark Michelson * res_pjsip: Add rtp_keepalive to sample config file. Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19 2015-07-23 13:11 +0000 [f635520527] Mark Michelson * Local channels: Alternate solution to ringback problem. Commit 54b25c80c8387aea9eb20f9f4f077486cbdf3e5d solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39 2015-07-22 12:24 +0000 [f509730cb9] Joshua Colp * audiohook: Use manipulated frame instead of dropping it. Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13 2015-07-22 09:46 +0000 [54b25c80c8] Mark Michelson * Local channels: Do not block control -1 payloads. Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815 2015-07-22 05:16 +0000 [f1493f900e] Joshua Colp * audiohook: Read the correct number of samples based on audiohook format. Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d 2015-07-20 12:39 +0000 [62c64c3bd1] Rusty Newton * Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca 2015-07-09 14:17 +0000 [d9094ddd73] Mark Michelson * res_pjsip: Add rtp_keepalive endpoint option. This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I06660ba672c0a343814af4cec838e6025cafd54b 2015-07-16 09:13 +0000 [a23adcca3d] Michael Cargile * res/res_musiconhold: Add a warning when MOH does not exist Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b 2015-07-19 09:11 +0000 [03064daeb2] Matt Jordan * res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf Misconfiguring sorcery.conf with a 'config' wizard with no extra data will currently crash Asterisk on startup, as the wizard requires a comma delineated list to parse. This patch updates res_sorcery_config to check for the presence of the data before it starts manipulating it. Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847 2015-07-16 09:46 +0000 [2c626ceb64] Joshua Colp * chan_pjsip: Don't change formats when frame of unsupported format is received. Receipt of an RTP packet currently causes the formats on an PJSIP channel to change to the format of the RTP packet. In some off-nominal cases it's possible for this to be a format that has not been configured or negotiated. This change makes it so only formats explicitly configured on the endpoint are allowed. ASTERISK-25258 #close Change-Id: If93d641fb6418a285928839300d7854cab8c1020 2015-07-17 04:59 +0000 [abb14ac5b8] Patric Marschall * sig_pri.h: force_restart_unavailable_chans in wrong scope In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549 2015-07-14 16:55 +0000 [875aee4c09] Richard Mudgett * pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable. ASTERISK-25256 #close Reported by: Richard Mudgett Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3 2015-07-08 16:39 +0000 [8bcf6d2801] Matt Jordan * ARI: Add support for push configuration of dynamic object This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c 2015-07-15 15:40 +0000 [e31cb6b248] Richard Mudgett * strings.h: Fix issues with escape string functions. Fixes for issues with the ASTERISK-24934 patch. * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is an empty string. If it were an empty string the functions returned NULL as if there were a memory allocation failure. This failure caused the AMI VarSet event to not get posted if the new value was an empty string. * Fixed dest buffer overwrite potential in ast_escape() and ast_escape_c(). If the dest buffer size is smaller than the space needed by the escaped s parameter string then the dest buffer would be written beyond the end by the nul string terminator. The num parameter was really the dest buffer size parameter so I renamed it to size. * Made nul terminate the dest buffer if the source string parameter s was an empty string in ast_escape() and ast_escape_c(). * Updated ast_escape() and ast_escape_c() doxygen function description comments to reflect reality. * Added some more unit test cases to /main/strings/escape to cover the empty source string issues. ASTERISK-25255 #close Reported by: Richard Mudgett Change-Id: Id77fc704600ebcce81615c1200296f74de254104 2015-07-14 14:29 +0000 [243c0d1609] Richard Mudgett * parking_applications.c: Fix ast_verb() line terminator. Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775 2015-07-14 14:36 +0000 [c782320c68] Richard Mudgett * res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. setup_park_common_datastore() was assuming that a non-NULL string returned for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty strings. Things got crashy as a result. * Made setup_park_common_datastore() treat the channel variable values the same whether they are NULL or empty for ATTENDEDTRANSFER and BLINDTRANSFER. ASTERISK-25254 #close Reported by: Richard Mudgett Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2 2015-07-10 18:01 +0000 [2735dd5b2d] Richard Mudgett * res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb 2015-07-10 10:42 +0000 [3d0ca343ca] Richard Mudgett * res_pjsip_session.c: Add some helpful comments and minor tweaks. Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743 2015-07-10 10:43 +0000 [8d08bb179c] Richard Mudgett * res_pjsip_session.c: Fix off nominal crash potential in debug message. Change-Id: I09928297927ee85f7655289acee3a586816466bc 2015-07-15 10:31 +0000 [0a1a550593] Matt Jordan * apps/app_dictate: Fix typo in attribution Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106 2015-07-15 10:28 +0000 [3384e64ef6] Benjamin Ford * ARI: Fixed unload mode for unload module. Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, which would unload a module even if it was in use. * Changed unload mode to proper mode ASTERISK-25173 Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533 2015-07-08 16:38 +0000 [0b6ff77afb] Matt Jordan * res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails Having a debug message tell us that we attempted to look up an item but failed is nice in circumstances when it isn't clear if the wizard was queried correctly or not. Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7 2015-07-08 16:37 +0000 [2f0d6d346c] Matt Jordan * res/res_pjsip_outbound_registration: Fix WARNING message Newlines are nice. Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42 2015-07-08 16:35 +0000 [cd2213f1ae] Matt Jordan * res_pjsip/configuration: Fix a variety of default value problems This patch fixes some bad default value handling in the following settings: * The 'message_context' and 'accountcode' settings are not mandatory. As such, we can allow their stringfield values to be empty. * The 'media_encryption' setting applies a default value of 'none' to the setting, which it then can't parse or understand. Since the value is documented to be 'no', this will now apply that as the default value. Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83 2015-07-08 16:32 +0000 [2e4bdbd78a] Matt Jordan * main/sorcery: Provide log messages when a wizard does not support an operation If a sorcery wizard does not support one of the 'optional' CRUD operations (namely the CUD), log a WARNING message so we are aware of why the operation failed. This also removes an assert in this case, as the CUD operation may have been triggered by an external system, in which case it is not a programming error but a configuration error. Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53 2015-07-10 18:17 +0000 [653f2087e0] Richard Mudgett * res_pjsip_session.c: Fix crash on call disconnect. The crash fix for ASTERISK-25183 backported some code from master to try to make sure that a BYE response is processed by the same serializer used by the BYE request. The identified race condition causing that backport was the BYE request code had not finished processing after sending the BYE before the BYE response came in for processing under a different thread. Unfortunately, there is still a race condition. Now the race condition is between destroying the call session's serializer in ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a reference to the serializer for a BYE response. Even worse, the new race condition is a design limitation of the taskprocessor implementation that didn't matter in versions before v12. Back then, taskprocessors were only destroyed when a module unloaded. Now res_pjsip can destroy them when a call ends. However, as noted on the ASTERISK-25183 commit, session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED. This is a tad too soon because our BYE request transaction has not completed yet. * Split session_end() that is called by session_inv_on_state_changed() to hold off session destruction until the BYE transaction timeout occurs or a failed initial INVITE transaction timeout occurs in session_inv_on_tsx_state_changed(). ASTERISK-25201 #close Reported by: Matt Jordan Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961 2015-07-14 13:12 +0000 [1aafadf814] Benjamin Ford * ARI: Added new functionality to reload a single module. An http request can be sent to reload an Asterisk module. If the module can not be reloaded or is not already loaded, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, based on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be reloaded through http requests ASTERISK-25173 Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1 2015-07-14 08:55 +0000 [9dcae23cfc] Benjamin Ford * ARI: Added new functionality to unload a single module. An http request can be sent to unload an Asterisk module. If the module can not be unloaded or is already unloaded, an error response will be returned. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be unloaded through http requests ASTERISK-25173 Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57 2015-07-13 16:00 +0000 [c219a98d2b] Benjamin Ford * ARI: Added new functionality to load a single module. An http request can be sent to load an Asterisk module. If the module can not be loaded or is loaded already, an error response will be returned. The command curl -v -u user:pass -X POST 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be loaded through http requests ASTERISK-25173 Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33 2015-07-13 10:54 +0000 [73e35d20de] Benjamin Ford * ARI: Added new functionality to get information on a single module. An http request can be sent to retrieve information on a single module, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on a single module can now be retrieved ASTERISK-25173 Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463 2015-07-08 14:56 +0000 [97ee0ee6c6] Kevin Harwell * bridge.c: Fixed race condition during attended transfer During an attended transfer a thread is started that handles imparting the bridge channel. From the start of the thread to when the bridge channel is ready exists a gap that can potentially cause problems (for instance, the channel being swapped is hung up before the replacement channel enters the bridge thus stopping the transfer). This patch adds a condition that waits for the impart thread to get to a point of acceptable readiness before allowing the initiating thread to continue. ASTERISK-24782 Reported by: John Bigelow Change-Id: I08fe33a2560da924e676df55b181e46fca604577 2015-07-08 16:28 +0000 [bb76b88baf] Matt Jordan * main/sorcery: Don't fail object set creation from JSON if field fails Some individual fields may fail their conversion due to their default values being invalid for their custom handlers. In particular, configuration values that depend on others being enabled (and thus have an empty default value) are notorious for tripping this routine up. An example of this are any of the DTLS options for endpoints. Any of the DTLS options will fail to be applied (as DTLS is not enabled), causing the entire object set to be aborted. This patch makes it so that we log a debug message when skipping a field, and rumble on anyway. ASTERISK-25238 Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76 2015-07-08 16:21 +0000 [5f13c2226a] Matt Jordan * main/format_cap: Parse capabilities generated by ast_format_cap_get_names We have a strange relationship between the parsing of format capabilities from a string and their representation as a string. We expect the format capabilities to be expressed as a string in the following format: allow = !all,ulaw,alaw disallow = g722 While we would generate the string representation of those formats as: allow = (ulaw|alaw) disallow = (ulaw|alaw|g729...) When the configuration framework needs to store values as a string, it generates the format capabilities using the second representation; this representation however cannot be parsed when the entry is rehydrated. This patch fixes that by updating ast_format_cap_update_by_allow_disallow to parse an entry as if it were in the generated format if it has a leading '(' and a trailing ')'. ASTERISK-25238 Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca 2015-06-27 17:53 +0000 [2325b106fd] Matt Jordan * tests/test_devicestate: Add additional tests for the device state API This patch adds more tests that exercise the device state API. This includes: * Tests that cover adding a device state provider, as well as deleting a device state provider. This also verifies that you cannot add an already added device state provider, and cannot delete an already deleted device state provider. * A test that covers changing device state and receiving said updates from a device state subscriber. This also covers hitting both the device state cache as well as a custom device state provider. * A test that covers converting device state to channel state and device state values to a string representation and back. * A test that covers obtaining device state from an active channel and a channel driver that provides its own device state. Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d 2015-06-27 17:51 +0000 [328f0be806] Matt Jordan * main/devicestate: Prevent duplicate registration of device state providers Currently, the device state provider API will allow you to register a device state provider with the same case insensitive name more than once. This could cause strange issues, as the duplicate device state providers will not be queried when a device's state has to be polled. This patch updates the API such that a device state provider with the same name as one that has already registered will be rejected. Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2 2015-07-10 22:25 +0000 [bee41eec62] Matt Jordan * res/res_sorcery_memory_cache: Fix test registration issues Again, tests now need to not end with a newline. This patch makes it so the tests can register again, unit tests will actually pass, and we can stop wasting time trying to figure out why builds are failing when they really aren't failing. Change-Id: Ide519fbeba89f413c733446c5ff7b224fc4ce840 2015-07-10 21:42 +0000 [4d738e9026] Matt Jordan * tests/test_sorcery_memory_cache_thrash: Fix test loading problems Because unit tests now want descriptions to not end with a newline, the sorcery memory cache thrash tests failed to register. This patch corrects their descriptions. Change-Id: Id004b1becfdeed8ee3c846f49beab76a5c0f68b6 2015-06-26 10:57 +0000 [47ea312b24] Benjamin Ford * ARI: Added new functionality to get all module information. An http request can be sent to retrieve a list of all existing modules, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET 'http://localhost:8088/ari/ asterisk/modules" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Information on modules can now be retrieved Change-Id: I63cbbf0ec0c3544cc45ed2a588dceabe91c5e0b0 2015-07-09 09:18 +0000 [d558b00c85] Joshua Colp * bridge_native_rtp.c: Don't start native RTP bridging after attended transfer. The bridge_native_rtp module adds a frame hook to channels which are in a native RTP bridge. This frame hook is used to intercept when a hold or unhold frame traverses the bridge so native RTP can be stopped or started as appropriate. This is expected but exposes a specific bug when attended transfers are involved. Upon completion of an attended transfer an unhold frame is queued up to take one of the channels involved off hold. After this is done the channel is moved between bridges. When the frame hook is involved in this case for the unhold it releases the channel lock and acquires the bridge lock. This allows the bridge core to step in and move the channel (potentially changing the bridging techology) from another thread. Once completed the bridge lock is released by the bridge core. The frame hook is then able to acquire the bridge lock and wrongfully starts native RTP again, despite the channel no longer being in the bridge or needing to start native RTP. In fact at this point the frame hook is no longer attached to the channel. This change makes it so the native RTP bridge data is available to the frame hook when it is invoked. Whether the frame hook has been detached or not is stored on the native RTP bridge data and is checked by the frame hook before starting or stopping native RTP bridging. If the frame hook has been detached it does nothing. ASTERISK-25240 #close Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2 2015-05-16 17:02 +0000 [b74b071369] Joshua Colp * res_sorcery_memory_cache: Backport to 13 Gerrit is complaining of conflicts when trying to create a patch series of all of the cherry-picked master commits, so I have instead squashed it all into one commit. ASTERISK-25067 #close Reported by: Matt Jordan Change-Id: I6dda90343fae24a75dc5beec84980024e8d61eb9 2015-07-08 04:21 +0000 [7ff1ac8797] Joshua Colp * res_rtp_asterisk: Ensure DTLS timeout timer is -1 if DTLS is not used. This change fixes a bug where the DTLS timeout timer would be initialized to 0 if DTLS was not used for an RTP session. ASTERISK-25103 Change-Id: If8d26bb054f1d300838850da5b8db9044c2fe2ac 2015-07-01 07:55 +0000 [05e8e14982] Joshua Colp * res_rtp_asterisk: Prevent simultaneous access to DTLS SSL context. This change moves logic for setting up the DTLS SSL contexts to when the SDP is done being processed instead of when ICE negotiation completes. It also stops handshakes from being initiated when we are acting as a server. Manipulating the SSL context when ICE negotiation has completed is problematic as the SSL context is not protected and if acting as a client the remote side may have started DTLS negotiation already. The retransmission timeout timer code has also been split up and simplified some. Both RTP and RTCP now have their own timers and the points at which the timer is stopped and started is now more specific. When a packet is sent the timer is started. When a response is received but before it is processed the timer is stopped. This provides a guarantee that the timeout is not occurring while the response is processed. ASTERISK-22805 #close ASTERISK-24550 #close ASTERISK-24651 #close ASTERISK-24832 #close ASTERISK-25103 #close ASTERISK-25127 #close Change-Id: Ib75ea2546f29d6efc3d2d37c58df6986c7bd9b91 2015-06-26 16:10 +0000 [38bace4fbb] Richard Mudgett * res_pjsip_t38.c: Fix always false if test. Calling t38_change_state() sets the t38 state so it makes little sense to then check the state right after the call for something else. * Made the code in t38_interpret_parameters() reject or exit T.38 mode as intended but not implemented. Change-Id: Ib281263a6ed44da9448132c4e6df1e183b8a3df2 2015-06-30 11:17 +0000 [2f7688c788] Richard Mudgett * res_pjsip_mwi.c: Use safer loop coding in mwi_subscription_mailboxes_str(). Change-Id: I6f39d809a6d1b47b35bb32b298f5a12f35d6f907 2015-06-30 11:14 +0000 [74be3a50d7] Richard Mudgett * res_pjsip_mwi.c: Eliminate a simple RAII_VAR. Change-Id: Ib1843f81e826a6c760c424c88eb70c350d9d61da 2015-06-30 11:11 +0000 [589e93617a] Richard Mudgett * res_pjsip_mwi.c: Fix mid-line log message line breaks. * Add create_mwi_subscriptions_for_endpoint() doxygen comment. Change-Id: I3c3f921f4ec749fb65b62d2f6fa0d4d1888b94e2 2015-06-26 18:48 +0000 [0d67e04359] Richard Mudgett * res_pjsip_mwi.c: Fix MWI subscription memory corruption crash. MWI subscriptions can crash or corrupt memory when using the subscription datastore to access the MWI subscription object because the datastore is not holding a reference to the object. * Give the subscription datastore a ref to the MWI subscription object. It is unfortunate that the ref causes a circular ref chain that must be explicitly broken to allow the memory to get released. The loop is broken when the subscription is shutdown and if the subscription setup fails. ASTERISK-25168 #close Reported by: Carl Fortin Change-Id: Ice4fa823f138ff10a6c74d280699c41a82836d4f 2015-07-02 14:51 +0000 [0422433f47] Richard Mudgett * PJSIP XML, XPIDF: Fix buffer size overwrite memory corruption error. When res_pjsip body generator modules were generating XML or XPIDF response bodies, there was a chance that the generated body would be the exact size of the supplied buffer. Adding the nul string terminator would then write beyond the end of the buffer and potentially corrupt memory. * Fix MALLOC_DEBUG high fence violations caused by adding a nul string terminator on the end of a buffer for XML or XPIDF response bodies. * Made calls to pj_xml_print() safer if the XML prolog is requested. Due to a bug in pjproject, the return value could be -1 _or_ AST_PJSIP_XML_PROLOG_LEN if the supplied buffer is not large enough. * Updated the doxygen comment of AST_PJSIP_XML_PROLOG_LEN to describe the return value of pj_xml_print() when the supplied buffer is not large enough. ASTERISK-25168 Reported by: Carl Fortin Change-Id: Id70e1d373a6a2b2bd9e678b5cbc5e55b308981de 2015-06-26 10:36 +0000 [8ea214aed7] Richard Mudgett * PJSIP FAX: Fix T.38 automatic reject timer NULL channel pointer dereferences. When a caller calls a FAX number and then hangs up right after the call is answered then the T.38 re-INVITE automatic reject timer may still be running after the channel goes away. * Added session NULL channel checks on the code paths that get executed by t38_automatic_reject() to prevent a crash when the T.38 re-INVITE automatic reject timer expires. ASTERISK-25168 Reported by: Carl Fortin Change-Id: I07b6cd23815aedce5044f8f32543779e2f7a2403 2015-06-05 15:37 +0000 [ada7346792] Richard Mudgett * res_pjsip: Need to use the same serializer for a pjproject SIP transaction. All send/receive processing for a SIP transaction needs to be done under the same threadpool serializer to prevent reentrancy problems inside pjproject and res_pjsip. * Add threadpool API call to get the current serializer associated with the worker thread. * Pick a serializer from a pool of default serializers if the caller of res_pjsip.c:ast_sip_push_task() does not provide one. This is a simple way to ensure that all outgoing SIP request messages are processed under a serializer. Otherwise, any place where a pushed task is done that would result in an outgoing out-of-dialog request would need to be modified to supply a serializer. Serializers from the default serializer pool are picked in a round robin sequence for simplicity. A side effect is that the default serializer pool will limit the growth of the thread pool from random tasks. This is not necessarily a bad thing. * Made pjsip_distributor.c save the thread's serializer name on the outgoing request tdata struct so the response can be processed under the same serializer. This is a cherry-pick from master. **** ASTERISK-25115 Change-Id: Iea71c16ce1132017b5791635e198b8c27973f40a NOTE: session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog becomes PJSIP_INV_STATE_DISCONNECTED. Unfortunately this is a tad too soon because our BYE request transaction has not completed yet. ASTERISK-25183 #close Reported by: Matt Jordan Change-Id: I8bad0ae1daf18d75b8c9e55874244b7962df2d0a 2015-07-04 18:22 +0000 [55137c3d12] Joshua Colp * res/res_http_websocket: Don't send HTTP response fragmented. This change makes it so that when accepting a WebSocket connection the HTTP response is sent as one packet instead of fragmented. Browsers don't like it when you send it fragmented. ASTERISK-25103 Change-Id: I9b82c4ec2949b0bce692ad0bf6f7cea9709e7f69 2015-06-27 18:47 +0000 [49f81ddb85] Matt Jordan * Makefile: Remove coverage files on 'make clean' This patch updates a variety of Makefiles in Asterisk's build system to remove .gcda and .gcno files when 'make clean' is executed. These files are generated when '--enable-coverage' is passed to the Asterisk configure script. Change-Id: Ib70b41eea2ee2908885bff02e80faf9f40c84602 2015-07-02 09:08 +0000 [e0f565663b] Walter Doekes * chan_sip: Fix early call pickup channel leak. When handle_invite_replaces() was called, and either ast_bridge_impart() failed or there was no bridge (because the channel we're picking up was still ringing), chan_sip would leak a channel. Thanks Matt and Corey for checking the bridge path. ASTERISK-25226 #close Change-Id: Ie736bb182170a73eef5bcef0ab0376f645c260c8 2015-07-02 06:19 +0000 [a5a262be78] Walter Doekes * chan_mgcp: Don't call close on fd -1. ASTERISK-25220 #close Change-Id: Ic48f3a82f51ada87f2fb0e016c9efe0ad56f1ee3 2015-07-02 06:10 +0000 [b835312b4c] Walter Doekes * rtp_engine: Skip useless self-assignment in ast_rtp_engine_unload_format. When running valgrind on Asterisk, it complained about: ==32423== Source and destination overlap in memcpy(0x85a920, 0x85a920, 304) ==32423== at 0x4C2F71C: memcpy@@GLIBC_2.14 (in /usr/lib/valgrind/...) ==32423== by 0x55BA91: ast_rtp_engine_unload_format (rtp_engine.c:2292) ==32423== by 0x4EEFB7: ast_format_attr_unreg_interface (format.c:1437) The code in question is a struct assignment, which may be performed by memcpy as a compiler optimization. It is changed to only copy the struct contents if source and destination are different. ASTERISK-25219 #close Change-Id: I6d3546c326b03378ca8e9b8cefd41c16e0088b9a 2015-07-02 05:16 +0000 [6551e16e03] Walter Doekes * astfd: Fix buffer overflow in DEBUG_FD_LEAKS. If DEBUG_FD_LEAKS was used and more file descriptors than the default of 1024 were available, some DEBUG_FD_LEAKS-patched functions would overwrite memory past the fixed-size (1024) fdleaks buffer. This change: - adds bounds checks to __ast_fdleak_fopen and __ast_fdleak_pipe - consistently uses ARRAY_LEN() instead of sizeof() or 1023 or 1024 - stores pointers to constants instead of copying the contents - reorders the fdleaks struct for possibly tighter packing - adds a tiny bit of documentation ASTERISK-25212 #close Change-Id: Iacb69e7701c0f0a113786bd946cea5b6335a85e5 2015-07-02 04:57 +0000 [f4dd9560cf] Walter Doekes * res_timing: Don't close FD 0 when out of open files. This fixes so a failure to get a timer file descriptor does not cascade to closing FD 0. On error, both res_timing_kqueue and res_timing_timerfd would call the destructor before setting the file handle. The file handle had been initialized to 0, causing FD 0 to be closed. This in turn, resulted in floods of "CLI>" messages and an unusable terminal. ASTERISK-19277 #close Reported by: Barry Chern For the 13 branch, this was already fixed. This patch only ensures that we do not attempt to close a negative file descriptor. Change-Id: I147d7e33726c6e5a2751928d56561494f5800350 2015-07-01 17:25 +0000 [78a1f4aa46] Richard Mudgett * chan_vpb.cc: Fix compiler warning Jenkins found. Change-Id: I0ec7fd10d56d90d5a60b12b5a7d6807f265ac5e0 2015-07-01 13:34 +0000 [6b16fbfc22] Scott Griepentrog * Channel alert pipe: improve diagnostic error return When a frame is queued on a channel, any failure in ast_channel_alert_write is logged along with errno. This change improves the diagnostic message through aligning the errno value with actual failure cases. ASTERISK-25224 Reported by: Andrey Biglari Change-Id: I1bf7b3337ad392789a9f02c650589cd065d20b5b 2015-07-01 16:04 +0000 [8e07ab145d] Matt Jordan * sorcery/realtime: Add a bit of debug and warning messages for bad configs When a mapping does not exist between a sorcery.conf defined object and a realtime mapping in extconf, currently, the user will receive a slew of ERROR messages that don't really tell what is happening. Some ERROR messages may even be misleading, as they occur after the sorcery API has already given up on the attempt to load and create the sorcery object. This patch adds a bit of debug and a useful WARNING message for when a wizard's open callback fails for a particular object type. In the bad configurations that resulted in this patch, this provided a 'root cause' WARNING message that pointed in the right direction of the configuration problem. Change-Id: I1cc7344f2b015b8b9c85a7e6ebc8cb4753a8f80b 2015-06-29 12:45 +0000 [156395e743] Mark Michelson * res_sorcery_realtime: Fix leak of sorcery object type. This prevents a leak of a sorcery object type when realtime sorcery objects are retrieved by fields or when multiple objects are retrieved. The extent of this leak is that sorcery object types would be leaked. These are allocated whenever an object type is registered with sorcery, meaning that on module shutdown, these objects would be leaked. This could be problematic if many reloads were performed, but it is not as severe as if every sorcery object retrieved from realtime were being leaked. ASTERISK-25165 #close Reported by Corey Farrell Change-Id: I625c3b50eee4576670b7eeb013c81ad043b4b4f8 2015-06-26 22:02 +0000 [a5e9c4e9b2] Matt Jordan * res/res_corosync: Always decline module load, instead of failing Returns a 'failure' from the module load routine indicates to Asterisk that it should abort loading completely. This is rarely - in fact, really, never - a good option. Aborting load of Asterisk from a dynamic module implies that the core, and the rest of the dynamic modules, don't matter: we should abandon all processing. res_corosync is really not that important. This patch updates the module such that, if it fails to load, it politely declines (emitting ERROR messages along the way), and allows Asterisk to continue to function. Note that this issue was keeping Asterisk unit tests from running on certain build agents. Change-Id: I252249e81fb9b1a68e0da873f54f47e21d648f0f 2015-06-26 20:38 +0000 [399cd8bcd9] Matt Jordan * main/pbx: Resolve case sensitivity regression in PBX hints When 8297136f was merged for ASTERISK-25040, a regression was introduced surrounding the case sensitivity of device names within hints. Previously, device names - such as 'sip/foo' - were compared in a case insensitive fashion. Thus, 'sip/foo' was equivalent to 'SIP/foo'. After that patch, only the case sensitive name would match, i.e., 'SIP/foo'. As a result, some dialplan hints stopped working. This patch re-introduces case insensitive matching for device names in hints. ASTERISK-25040 ASTERISK-25202 #close Change-Id: If5046a7d14097e1e3c12b63092b9584bb1e9cb4c (cherry picked from commit 96bbcf495a1da9e607d9b04a44b5c4f49e83cc03) 2015-06-26 16:12 +0000 [24eec5a10b] Mark Michelson * res_pjsip_nat: Adjust when contact should be rewritten. A previous change made the contact only get rewritten if the dialog's route set was not marked frozen. Unfortunately, while the intent of this is correct, the dialog's route set actually gets marked as frozen earlier than expected, especially for UAS dialogs. Instead, the idea is that the contact needs to not be rewritten if there is a pre-existing route set on the dialog. This is now accomplished by checking the dialog's route set list instead of checking if the route set is frozen. Doing this causes some broken tests to begin passing again. ASTERISK-25196 Reported by Mark Michelson Change-Id: I525ab251fd40a52ede327a52a2810a56deb0529e 2015-06-19 18:27 +0000 [0ec461a637] Richard Mudgett * res_pjsip_outbound_registration.c: Add a serializer shutdown group. The client_state objects contain a serializer used to send the outbound REGISTER messages. Once all those message transactions are complete then the module can shutdown. ASTERISK-24907 #close Reported by: Kevin Harwell Change-Id: Ibb2fe558f98190f2a06da830e0fadfa25516f547 2015-06-26 10:41 +0000 [05a2cc1293] Mark Michelson * res_pjsip_refer: Prevent sending duplicate headers. res_pjsip_refer will attempt to add Referred-By or Replaces headers to outbound INVITEs at times. If the INVITE gets challenged for authentication, then we will resend the INVITE. Prior to this patch, the Referred-By or Replaces header would be re-added to the outbound INVITE, resulting in duplicated headers. ASTERISK-25204 #close Reported by Mark Michelson Change-Id: I59fb5c08b4d253c0dba9ee3d3950b5025358222d 2015-06-23 17:43 +0000 [028fa54620] Mark Michelson * res_pjsip_nat: Rewrite route set when required. When performing some provider testing, the rewrite_contact option was interfering with proper construction of a route set when sending an ACK after receiving a 200 OK response to an INVITE. The initial INVITE was sent to address sip:foo. The 200 OK had a Contact header with URI sip:bar. In addition, the 200 OK had Record-Route headers for sip:baz and sip:foo, in that order. Since the Record-Route headers had the lr parameter, the result should have been: * Set R-URI of the ACK to sip:bar. * Add Route headers for sip:foo and sip:baz, in that order. However, the rewrite_contact option resulted in our rewriting the Contact header on the 200 OK to sip:foo. The result was: * R-URI remained sip:foo. * We added Route headers for sip:foo and sip:baz, in that order. The result was that sip:bar was not indicated in the ACK at all, so the far end never received our ACK. The call eventually dropped. The intention of rewrite_contact is to rewrite the most immediate destination of our SIP request to be the same address on which we received a request or response. In the case of processing a SIP response with Record-Route headers, this means that instead of rewriting the Contact header, we should instead rewrite the bottom-most Record-Route header. In the case of processing a SIP request with Record-Route headers, this means we rewrite the top-most Record-route header. Like when we rewrite the Contact header, we also ensure to update the dialog's route set if it exists. ASTERISK-25196 #close Reported by Mark Michelson Change-Id: I9702157c3603a2d0bd8a8215ac27564d366b666f 2015-06-19 16:16 +0000 [84c12f9e0c] Richard Mudgett * threadpool, res_pjsip: Add serializer group shutdown API calls. A module trying to unload needs to wait for all serializers it creates and uses to complete processing before unloading. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I8c80b90f2f82754e8dbb02ddf3c9121e5e966059 2015-06-16 15:06 +0000 [602c4b74b5] Richard Mudgett * res_pjsip_outbound_registration.c: Fix handle_client_state_destruction() refs * handle_client_state_destruction() must always be passed a ref to client_state because it will always unref client_state. handle_registration_response() was not passing a client_state ref. * Made the final un-REGISTER message get sent normally using the pjproject register control structure in handle_client_state_destruction(). The previous code attempted to short circuit the response handling for the module to unload. That doesn't work for a couple reasons. One, pjsip_regc_send() may call the registered callback before it returns and unbalance the client_state ref count. Two, the registered callback handles any authentication for the un-REGISTER message. * Made the distinction between internal registration state and external registration status with sip_outbound_registration_status_str(). This is necessary to avoid altering documented AMI messages with internal changes. * Removed references to client_state->client outside of the serializer thread. When handle_client_state_destruction() destroys the pjproject register control structure that memory is freed and cannot be referenced anymore. These accesses were to provide information for debug and off-nominal warning messages. * In sip_outbound_registration_timer_cb() you should not access entry->id after unrefing client_state because the passed in entry is normally pointing to the timer entry in the client_state object. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: Ia7b446d8644b6b4550ef5bea49527671de65183f 2015-06-15 15:28 +0000 [8c6a95a9ac] Richard Mudgett * res_pjsip_outbound_registration.c: Use ast_sorcery_object_unregister() API The sorcery pjsip 'registration' config object needs to be destroyed on module unload. Otherwise, a reload of res_pjsip could try to use callbacks for a previously unloaded instance of the module provided by ast_sorcery_object_register() or one of the variants. Also, if res_pjsip_outbound_registration were subsequently reloaded, the sorcery config field objects would be registered in sorcery twice. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I304fad13dece2604af48353f6c6d9d5c7b064697 2015-06-25 06:42 +0000 [e4a2ef9e4e] Joshua Colp * channel: Remove ignore of answer on non-outgoing channels. Due to the way that channels can now be moved around inside of Asterisk it is possible for the outgoing flag of a channel to get cleared before it has been answered. This results in the bridge not receiving notification that the outgoing leg has been answered. This most easily exhibits itself with DTMF based blond transfers. Since the answer of the outgoing leg is ignored the other party continues to receive both a locally generated ringing and the media stream of the outgoing leg upon its answer. This results in no media being heard. This change removes the ignore of the answer and allows it to pass through. ASTERISK-25171 #close Change-Id: I82aedcec4f89f34a2e5472086dfc9a6c775bca8e 2015-06-15 15:28 +0000 [20f3d77ab9] Richard Mudgett * sorcery: Add ast_sorcery_object_unregister() API call. Find and unlink the specified sorcery object type to complement ast_sorcery_object_register(). Without this function you cannot completely unload individual modules that use sorcery for configuration. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I1c04634fe9a90921bf676725c7d6bb2aeaab1c88 2015-06-15 13:38 +0000 [4313f32969] Richard Mudgett * res_pjsip_outbound_registration.c: Reorder load_module() and unload_module(). It is best if the loading code creates and initializes the module's infrastructure before letting the system know of its existence. The unloading code needs to reverse the actions of the loading code and in the reverse order. ASTERISK-24907 Reported by: Kevin Harwell Change-Id: I5d151383e9787b5b60aa5e1627b10f040acdded4 2015-06-23 14:34 +0000 [890c923786] Richard Mudgett * AMI: Add Linkedid to the standard channel snapshot AMI event headers. * The AMI version is bumped to 2.8.0. ASTERISK-25189 #close Reported by: John Hardin Change-Id: I2b1778c3fdc1dca0ed55db4e3a639eddfb16c2ac 2015-06-24 14:30 +0000 [2602a7484b] Richard Mudgett * test.c: Add unit test registration checks for summary and description. Added checks when a unit test is registered to see that the summary and description strings do not end with a new-line '\n' for consistency. The check generates a warning message and will cause the /main/test/registrations unit test to fail. * Updated struct ast_test_info member doxygen comments. Change-Id: I295909b6bc013ed9b6882e85c05287082497534d 2015-06-24 14:39 +0000 [2b0482d699] Richard Mudgett * Unit tests: Fix unit test description strings. Analyzing the code shows that the unit test summary and description strings should not end with a new-line character. Where these strings are used in the code a new-line is provided for output. Change-Id: I129284f5e7ca93d82532334076da4c462d3d9fba 2015-06-23 11:21 +0000 [e99e654d75] Joshua Colp * app_dial: Hold reference to calling channel formats when dialing outbound. Currently when requesting a channel the native formats of the calling channel are provided to the core for usage when dialing the outbound channel. This occurs without holding the channel lock or keeping a reference to the formats. This is problematic as the channel driver may end up changing the formats during this time. In the case of chan_sip this happens when an SDP negotiation completes. This change makes it so app_dial keeps a reference to the native formats of the calling channel which guarantees that they will remain valid for the period of time needed. ASTERISK-25172 #close Change-Id: I2f0a67bd0d5d14c3bdbaae552b4b1613a283f0db 2015-06-17 05:04 +0000 [80e82dc97f] Joshua Colp * res_pjsip_mwi: Set up unsolicited MWI upon registration. The res_pjsip_mwi previously required a reload to set up the proper subscriptions to allow unsolicited MWI to work. This change makes it so the act of registering will also cause this to occur. This is particularly useful if realtime is involved as no reload needs to occur within Asterisk to cause the MWI information to get sent. ASTERISK-25180 #close Change-Id: Id847b47de4b8b3ab8858455ccc2f07b0f915f252 2015-06-22 15:11 +0000 [35a99b6394] Kevin Harwell * bridge.c: Hangup attended transfer target if bridged After completing an attended transfer the transfer target channel was not being hung up after leaving the bridge. Added an explicit softhangup to hangup said channel, but only if it was previously bridged. ASTERISK-24782 #close Reported by: John Bigelow Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada 2015-06-17 16:23 +0000 [036bc0012f] Richard Mudgett * res_pjsip_outbound_registration.c: Add missing line endings to CLI commands Change-Id: I39ae612746d892d2dbe86f3ff2d7027fa1da57f7 2015-06-12 14:29 +0000 [bec7435945] Richard Mudgett * res_pjsip_outbound_registration.c: Eliminate simple RAII_VAR() usage. Change-Id: I399cb9d61bbba706b48c98e0bf75e98984cd9a9e 2015-06-12 13:33 +0000 [c2519fdf1c] Richard Mudgett * res_pjsip_outbound_registration.c: Misc code cleanups. * Break some long lines. * Fix doxygen comment. Change-Id: I8f12ba6822f84d5e7bb575280270cd7e2fefb305 2015-06-22 09:26 +0000 [a419c69def] Alexander Traud (License 6520) * chan_sip: Reload peer without its old capabilities. On reload, previously allowed codecs were not removed. Therefore, it was not possible to remove codecs while Asterisk was running. Furthermore, newly added codecs got appended behind the previous codecs. Therefore, it was not possible to add a codec with a priority of #1. This change removes the old capabilities before the current ones are added. ASTERISK-25182 #close Reported by: Alexander Traud patches: asterisk_13_allow_codec_reload.patch uploaded by Alexander Traud (License 6520) Change-Id: I62a06bcf15e08e8c54a35612195f97179ebe5802 2015-06-20 19:38 +0000 [74616ae43d] Joshua Colp * chan_sip: Destroy peers without holding peers container lock. Due to the use of stasis_unsubscribe_and_join in the peer destructor it is possible for a deadlock to occur when an event callback is occurring at the same time. This happens because the peer may be destroyed while holding the peers container lock. If this occurs the event callback will never be able to acquire the container lock and the unsubscribe will never complete. This change makes it so the peers that have been removed from the peers container are not destroyed with the container lock held. ASTERISK-25163 #close Change-Id: Ic6bf1d9da4310142a4d196c45ddefb99317d9a33 2015-06-18 13:16 +0000 [9015bb4c8c] Mark Michelson * Resolve race conditions involving Stasis bridges. This resolves two observed race conditions. First, a bit of background on what the Stasis application does: 1a Creates a stasis_app_control structure. This structure is linked into a global container and can be looked up using a channel's unique ID. 2a Puts the channel in an event loop. The event loop can exit either because the stasis_app_control structure has been marked done, or because of some other factor, such as a hangup. In the event loop, the stasis_app_control determines if any specific ARI commands need to be run on the channel and will run them from this thread. 3a Checks if the channel is bridged. If the channel is bridged, then ast_bridge_depart() is called since channels that are added to Stasis bridges are always imparted as departable. 4a Unlink the stasis_app_control from the container. When an ARI command is received by Asterisk, the following occurs 1b A thread is spawned to handle the HTTP request 2b The stasis_app_control(s) that corresponds to the channel(s) in the request is/are retrieved. If the stasis_app_control cannot be retrieved, then it is assumed that the channel in question has exited the Stasis app or perhaps was never in Stasis in the first place. 3b A command is queued onto the stasis_app_control, and the channel's event loop thread is signaled to run the command. 4b While most ARI commands do nothing further, some, such as adding or removing channels from a bridge, will block until the command they issued has been completed by the channel's event loop. The first race condition that is solved by this patch involves a crash that can occur due to faulty detection of the channel's bridged status in step 3a. What can happen is that in step 2a, the event loop may run the ast_bridge_impart() function to asynchronously place the channel into a bridge, then immediately exit the event loop because the channel has hung up. In step 3a, we would detect that the channel was not bridged and would not call ast_bridge_depart(). The reason that the channel did not appear to be bridged was that the depart_thread that is spawned by ast_bridge_impart() had not yet started. That is the thread where the channel is marked as being bridged. Since we did not call ast_bridge_depart(), the Stasis application would exit, and then the channel would be destroyed Then the depart_thread would start up and try to manipulate the destroyed channel, causing a crash. The fix for this is to switch from using ast_channel_is_bridged() to checking the NULLity of ast_channel_internal_bridge_channel() to determine if ast_bridge_depart() needs to be called. The channel's internal bridge_channel is set when ast_bridge_impart() is called and is NULLed by the call to ast_bridge_depart(). If the channel's internal bridge_channel is non-NULL, then the channel must have been imparted into the bridge and needs to be departed, even if the actual bridging operation has not yet started. By departing the channel when necessary, the thread that is running the Stasis application will block until the bridge gives the okay that the depart_thread has exited. The second race condition that is solved by this patch involves a leak of HTTP handler threads. The problem was that step 2b would successfully retrieve a stasis_app_control structure. Then step 2a would exit the channel from the event loop due to a hangup. Steps 3a and 4a would execute, and then finally steps 3b and 4b would. The problem is that at step 4b, when attempting to add a channel to a bridge, the thread would block forever since the channel would never execute the queued command since it was finished with the event loop. This meant that the HTTP handling thread would be leaked, along with any references that thread may have owned (in my case, I was seeing bridges leaked). The fix for this is to hone in better on when the channel has exited the event loop. The stasis_app_control structure has an is_done field that is now set at each point where the channel may exit the event loop. If step 2b retrieves a valid stasis_app_control structure but the control is marked as done, then the attempted operation exits immediately since there will be nothing to service the attempted command. ASTERISK-25091 #close Reported by Ilya Trikoz Change-Id: If66265b73b4c9f8f58599124d777fedc54576628 2015-06-16 11:13 +0000 [723a9d4225] Mark Michelson * Parking: Add documentation for AMI ParkedCallSwap event. This event was added some time ago in order to clarify when a channel took the place of another channel in a parking lot. However, there was no XML documentation added for the event. This patch adds the XML documentation. ASTERISK-24900 #close Reported by Rusty Newton Change-Id: I4cfe7777c4b94bbff91c9221c6096a7a02a92eac 2015-06-15 16:40 +0000 [79bf56c78a] Corey Farrell * func_pjsip_aor: Fix leaked contact from iterator. ASTERISK-25162 #close Change-Id: Id79aa3c6fe490016ee98efc97ac4c1d3f461f97e 2015-06-12 16:58 +0000 [31c77b157b] Kevin Harwell * res_pjsip: Add option to force G.726 to be treated as AAL2 packed. Some phones send g.726 audio packed for AAL2, which differs from what is recommended by RFC 3351. If Asterisk receives audio formatted as such when negotiating g.726 then it sounds a bit distorted. Added an option to res_pjsip_endpoint that allows g.726 negotiated audio to be treated as g.726 AAL2 packed. ASTERISK-25158 #close Reported by: Steve Pitts Change-Id: Ie7e21f75493d7fe53e75e12c971e72f5afa33615 2015-06-14 19:48 +0000 [de8c7f46ed] Matt Jordan * main/cdr: Carry over the disable flag when 'disable all' is specified The CDR_PROP function (as well as the NoCDR application) set the 'disable all' flag (AST_CDR_FLAG_DISABLE_ALL) on the current CDR. This flag is supposed to be applied to all CDRs that are currently in the chain, as well as all CDRs that may be created in the future. Currently, however, the flag is only applied to the existing CDRs in the chain; new CDRs do not receive the 'disable all' flag. In particular, this affects parallel dials, which generate new CDRs for each pair of channels in the dial attempt. This patch carries over the 'disable all' flag when it is specified on a CDR and a new CDR is generated for the chain. ASTERISK-24344 #close Change-Id: I91a0f0031e4d147bdf8a68ecd08304d506fb6a0e 2015-06-12 14:28 +0000 [78ea356e78] Matt Jordan * main/cdr: Copy context/exten on chained CDRs for parallel dials in subroutines When a parallel dial occurs, a new CDR will be created for each dial attempt that is made. In most circumstances, the act of creating each CDR in the chain will include a step that updates the Party A snapshot, which causes the context/extension of the Party A to be copied onto the CDR object. However, when the Party A is in a subroutine, we explicitly do *not* copy the context/extension onto the CDR. This prevents the Macro or GoSub routine name from blowing away the context/extension that the channel was originally executing in. For the original CDR, this is not a problem: the original CDR already recorded the last known 'good' state of the channel just prior to it going into the subroutine. However, for newly generated CDRs in a chain, there is no context/extension set on them. Since we are in a subroutine, we will never set the Party A's context/extension on the CDR, and we end up with a CDR with no destination recorded on it. This patch updates the creation of a chained CDR such that it copies over the original CDR's context/extension. This is the last known "good" state of the CDR, and is a reasonable starting point for the newly generated CDR. In the case where we are not in a subroutine, subsequent code will update the location of the CDR from the Party A information; in the case where we are in a subroutine, the context/extension on the original CDR is the correct information. ASTERISK-24443 #close Change-Id: I6a3ef0d6e458d3b9b30572feaec70f2964f3bc2a 2015-06-11 08:18 +0000 [3f57f3f8ec] Damian Ivereigh * chan_sip.c: Update dialog fromtag after request with auth If a client sends and INVITE which is 401 rejected, then subsequently sends a new INVITE with the auth info and uses a different fromtag from the first INVITE, Asterisk will accept the new INVITE as part of the original dialog - match_req_to_dialog() specifically ignores the fromtag. However it does not update the stored dialog with the new fromtag. This results in Asterisk being unable to match future packets that are part of this dialog (such as the ACK to the OK or the OK to the BYE), and the call is dropped. This problem was originally found when using an NEC-i SV8100-GE (NEC SIP Card). * After a successful match of a packet to the dialog, if the packet is not a SIP_RESPONSE, authentication is present and the fromtags are different, the stored fromtag is updated with the one from the recent INVITE. ASTERISK-25154 #close Reported by: Damian Ivereigh Tested by: Damian Ivereigh Change-Id: I5c16cf3b409e5ef9f2b2fe974b6bd2a45a6aa17e 2015-06-11 18:52 +0000 [30a0f2d9ac] Matt Jordan * chan_pjsip: Set the context and extension on the channel when created Prior to this patch, chan_pjsip was failing to pass the endpoint's context and the desired extension to the ast_channel_alloc_* routine. This caused a new channel snapshot to be issued without a context and extension, which can cause some reporting issues for users of AMI, CEL, and other APIs. The channel driver would later set the context and extension on the channel such that the channel would start in the correct location in the dialplan, but the information reported in the initial event would be incorrect. This patch modifies the channel driver such that it now passes the context and extension directly into the allocation routine. This provides the information in the new channel snapshot published over Stasis. ASTERISK-25156 #close Reported by: cloos Change-Id: Ic6f8542836e596db8f662071d118e8f934fdf25e 2015-06-10 18:28 +0000 [dbb067279e] Joshua Colp * bridge: When performing a blonde transfer update connected line information. When performing a blonde transfer the code uses the old masquerade mechanism to move a channel around. As a result of this certain information, such as connected line, is moved between the channels involved. Upon completion of the move a frame is queued which is supposed to update the connected line information on the channel. This does not occur as the code considers it a redundant update since the masquerade operation updated the channel (but did not inform it of the new connected line information). The code also does not queue a connected line update to be handled by the thread handling the channel. Without this any other channel that may be loosely involved does not know it is talking to a different caller. This change does the following to resolve this: 1. The indicated connected line information is cleared upon completion of the masquerade operation when doing a blonde transfer. This prevents the connected line update from being considered redundant. 2. A connected line update frame is now queued upon the completion of the masquerade operation so any other channel loosely involved knows that there is a different caller. ASTERISK-25157 #close Reported by: Joshua Colp Change-Id: Ibb8798184a1dab3ecd35299faecc420034adbf20 2015-06-11 14:39 +0000 [a2f4d03c87] Richard Mudgett * app_directory: Fix crash when using the alias option 'a'. The voicemail.conf mailbox key/value pair is defined as: =[[,[,[,[,]]]]] Where all fields in the value including the field values are optional. Since the parsing code for the mailbox key/value pair is sloppy, this patch tightens the parsing for the directory information. * Renamed the 'pos' and 'bufptr' variables to 'name' and 'options' respectively in search_directory_sub(). Those names make more sense. * Made sure that search_directory_sub() is dealing with the voicemail.conf mailbox options field if it even exists when looking for the 'hidefromdir' and 'alias' options. * Fix crash if a voicemail.conf mailbox is just =, when the 'a' option is used. If there were no fields after the name then the 'options' pointer was not checked for NULL. * Fix users.conf alias processing if the 'a' option is used. The wrong variable was used. ASTERISK-25087 #close Reported by: Chet Stevens Change-Id: I86052ea77307beddddba5279824d39dc0d593374 2015-06-09 15:31 +0000 [a2b718f4f6] Richard Mudgett * res_pjsip.h: Fix some doxygen comments. Change-Id: I4615771077c3c6a0a7273da6d7b5f77af7e8d976 2015-06-05 13:46 +0000 [32ddf6d86b] Richard Mudgett * taskprocessor.c: Remove extra unref from off-nominal path. Change-Id: Iee3bd8c8a528776056972066698fe735f0f6cf60 2015-04-20 16:00 +0000 [cf98c744d5] Yousf Ateya * chan_iax2: Prevent deadlock between hangup and sending lagrq/ping channels/chan_iax.c: Prevent the deadlock between iax2_hangup and send_lagrq/ send_ping. This deadlock happens because the scheduled task send_lagrq(or send_ping) starts execution after the call hangup procedure starts but before it deletes the tasks in the scheduler. The solution is to delete scheduled lagrq (and ping) task asynchronously (i.e. schedule AST_SCHED_DEL for these tasks); By this, AST_SCHED_DEL will be called in a new context (doesn't have callno locked). This commit also cleans up the procedure of sending LAGRQ and PING. main/sched.c: Do not assert when deleting non existant entry from scheduler. This assert seems to be the reason for a lot of awkward code to avoid it. ASTERISK-24983 #close Reported by: Y Ateya Change-Id: I03bec1fc8faacb89630269e935fa667c6d6c080c 2015-05-31 12:37 +0000 [8af6c9cf6b] Ivan Poddubny * res_pjsip_transport_websocket: Fix use-after-free bugs. This patch fixes use-after-free bugs caught by AddressSanitizer. 1. PJSIP transport manager may decide to destroy transport on its own. For example, when the contact registered via websocket has not renewed its registration in time. The transport was destoyed, but the websocket listener thread was still active until the socket closes, and then tried to call transport_shutdown on transport that has been freed. Also, the transport destructor accessed wstransport->rdata.tp_info.pool right after freeing memory that contained wstransport itself. This patch converts transport to an ao2 object, allowing it to be refcounted, so that it is available until both websocket listener and pjsip transport manager are finished with it. 2. The websocket listener deletes the last reference on websocket session when the tcp connection is closed, and it gets destroyed, but the transport manager may still use it, for example when disconnect happens in the middle of a SIP transaction. A new reference to websocket session has been added that is released with the transport to prevent this. ASTERISK-25096 #close Reported by: Josh Kitchens ASTERISK-24963 #close Reported by: Badalian Vyacheslav Change-Id: Idc0b63eb6e459c1ddfb2430127d34b3c4d8d373b 2015-06-09 13:41 +0000 [3046bc17ed] ibercom * weakref attribute detection broken with gcc 4.6 and higher GCC 4.7 Manual: http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html weakref ("target") A weak reference is an alias that does not by itself require a definition to be given for the target symbol. ASTERISK-22559 #close Reported by: Ibercom Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf 2015-06-08 10:09 +0000 [55c8daf88b] Corey Farrell * Fix unsafe uses of ast_context pointers. Although ast_context_find, ast_context_find_or_create and ast_context_destroy perform locking of the contexts table, any context pointer can become invalid at any time that the contexts table is unlocked. This change adds locking around all complete operations involving these functions. Places where ast_context_find was followed by ast_context_destroy have been replaced with calls ast_context_destroy_by_name. ASTERISK-25094 #close Reported by: Corey Farrell Change-Id: I1866b6787730c9c4f3f836b6133ffe9c820734fa 2015-06-04 07:14 +0000 [e0090216db] ibercom * CLI: Cosmetic issue - core show uptime Show uptime information ends with an unnecessary space. Now NEEDCOMMA is better defined. Change-Id: I11b360504a0703309ff51772ff8f672287f3c5a1 2015-06-03 17:41 +0000 [88212ccb7f] Mark Michelson * res_pjsip: Prevent access of NULL channels. It is possible to receive incoming requests or responses after the channel on an ast_sip_session has been destroyed and NULLed out. Handlers of these sorts of requests or responses need to be prepared for the possibility that the channel is NULL or else they could cause a crash. While several places have been amended to deal with NULL channels, there were still a couple of places that needed updating. res_pjsip_dtmf_info.c: When handling incoming INFO requests, we need to return early if there is no channel on the session. res_pjsip_session.c: When handling a 302 response, we need to stop the redirecting attempt if there is no channel on the session. ASTERISK-25148 #close reported by Mark Michelson Change-Id: Id1a75ffc3d0eaa168b0b28188fb54d6cf9fc47a9 2015-06-01 11:45 +0000 [f5d5aa67dc] Kevin Harwell * AMI: Escape string values. So this issue is a bit complicated. Since it is possible to pass values to AMI that contain a '\r\n' (or other similar sequences) these values need to be escaped. One way to solve this is to escape the values and then pass the escaped values to the AMI variable parameter string building function. However, this puts the onus on the pre-build function to escape all string values. This potentially requires a fair amount of changes along with a lot of string allocations/freeing for all values. Surely there is a way to push this complexity down a level into the string building function itself? This of course is possible, but ends up requiring a way to distinguish between strings that need to be escaped and those that don't. The best way to handle this is by introducing a new format specifier in the format string. For instance a %s (no escape) and %S (escape). However, that is a bit weird and unexpected. So faced with those possibilities this patch implements a limited version of the first option. Instead of attempting to escape all string values this patch only escapes those values that make sense. This approach limits the number of changes and doesn't suffer from the odd format specifier problem. ASTERISK-24934 #close Reported by: warren smith Change-Id: Ib55a5b84fe0481b0f2caaaab68c566f392c0aac0 2015-06-03 13:17 +0000 [5dc9fb4198] gtjoseph * res_pjsip/location: Fix ref leak in contact_apply_handler contact_apply_handler calls ast_res_pjsip_find_or_create_contact_status to force the creation of a contact_status object whenever a new contact is added but it didn't unref the returned object. Added an ao2_cleanup(status) to plug the leak. ASTERISK-25141 Change-Id: Icc1401cae142855a1abc86ab5179dfb3ee861c40 Reported-by: Corey Farrell 2015-06-02 15:07 +0000 [d908272b7e] David M. Lee * Fixes for OS X * Add some type casting so tv_usec can really be a long, instead of some strange platform specific type. * Add some .dylib style files to .gitignore. * Switch from using -Xlinker to -Wl,. For [reasons unknown][], newer versions of GCC, when compiling the Homebrew formula for Asterisk, are not properly passing the -Xlinker options to the linker. Given that -Wl, does exactly the [same thing][], and does it properly, this patch changes the -Xlinker options to use -Wl, instead. [reasons unknown]: http://bit.ly/1SUbEYx [same thing]: https://gcc.gnu.org/onlinedocs/gcc/Link-Options.html Change-Id: Id5e6b3c6cc86282ea5fca630dc3991137c5bf4dd 2015-05-30 20:22 +0000 [9e7827e3ac] Corey Farrell * pjsip_configuration: Fix leak in persistent_endpoint_update_state. The loop to find the first available contact of an endpoint grabbed contact from the iterator, then checked for offline state. This caused the first contact after the state was found to leak a reference. ASTERISK-25141 Change-Id: Id0f1d87410fc63742db0594eb4b18b36e99aec08 2015-05-31 11:33 +0000 [888bb49618] Ivan Poddubny * Fix buffer overflow in slin sample frames generation. The length of frames retured by sample functions was twice as large as real, what caused global buffer overflow caught by AddressSanitizer. ASTERISK-24717 #close Reported by: Badalian Vyacheslav Change-Id: Iec2fe682aef13e556684912f906bedf7c18229c6 2015-05-29 16:19 +0000 [857166b5e5] gtjoseph * res_pjsip/location: Fix memory leak in permanent_uri_handler When permanent_uri_handler was creating the contact status object for each contact, it wasn't unreffing it at the end of the loop. ASTERISK-25141 #close Reported-by: Corey Farrell Change-Id: I7bb127994677bb3d459f87952f8425c9b9967b12 2015-05-29 14:52 +0000 [1558a89129] gtjoseph * Revert "endpoint/stasis: Eliminate duplicate events on endpoint status change" This reverts commit 35c699086ae2fd81b2473307ccb2ae79ad32375a. Change-Id: Ia98c2b4820cf579a5b9bb75e9e05d7a233205fb7 2015-05-27 13:22 +0000 [35c699086a] gtjoseph * endpoint/stasis: Eliminate duplicate events on endpoint status change When an endpoint was created, it's messages were being forwarded to both the tech endpoint topic and the all endpoints topic. Since the tech topic was also forwarded to all, this was resulting in duplicate messages whenever an endpoint published. This patch causes the endpoint to only forward to the tech topic and lets the tech topic forward to all. To accomplish this, the existing stasis_cp_single_create function (which both creates and forwards) was cloned and split into 2 functions, one that creates the topic and one that sets up the forwarding. This allows endpoint_internal_create to create the topic from the endpoint_all cache without forwarding it there, then allows it to do the forward to the tech's topic. ASTERISK-25137 #close Reported-by: Vitezslav Novy ASTERISK-25116 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I26d7d4926a0861748fd3bdffe316b75b549a801c 2015-05-26 13:56 +0000 [fe21f2e52f] Richard Mudgett * res_pjsip_session: Fix in-dialog authentication. When the remote peer requires authentication for in-dialog requests then re-INVITEs to the peer cause the call to be disconnected and other in-dialog requests to the peer like MESSAGE just don't go through. * Made session_inv_on_tsx_state_changed() handle in-dialog authentication for re-INVITEs and other methods. Initial INVITEs cannot be handled here because the INVITE transaction must be restarted earlier. * Pulled needed code from res/res_pjsip/pjsip_outbound_auth.c in preparation for removing the file. The generic outbound authentication code did not work as well as anticipated. * Created outbound_invite_auth() to only handle initial outbound INVITEs. Re-INVITEs cannot be handled here. The re-INVITE transaction is still in progress and the PJSIP library cannot handle the overlapping INVITE transactions. Other method types should not be handled here as this code only works on outgoing calls and we need to handle incoming and outgoing calls. ASTERISK-25131 #close Reported by: Richard Mudgett Change-Id: I12bdd7ddccc819b4ce4b091e826d1e26334601b0 2015-05-21 17:21 +0000 [262d590819] gtjoseph * res_pjsip: Add AMI events for chan_pjsip contact lifecycle changes Add a new ContactStatus AMI event. Publish the following status/state changes: Created Removed Reachable Unreachable Unknown Contact URI, new status/state, aor and endpoint names, and the last qualify rtt result are included in the event. ASTERISK-25114 #close Change-Id: Id25aae5f7122facba183273efb3e8f36c20fb61e Reported-by: George Joseph Tested-by: George Joseph 2015-05-26 07:44 +0000 [5a42397018] Joshua Colp * sorcery: Fix cache creation callback. The cache creation callback function expects to receive a sorcery_details structure and not just a standalone object. Change-Id: I3e4a5a137cb25292eb52d7a14cbb6daa09213450 2015-05-24 13:47 +0000 [97a6ce1717] Ivan Poddubny * Astobj2: Correctly treat hash_fn returning INT_MIN The code in astobj2_hash.c wrongly assumed that abs(int) is always > 0. However, abs(INT_MIN) = INT_MIN and is still negative, as well as abs(INT_MIN) % num_buckets, and as a result this led to a crash. One way to trigger the bug is using host=::80 or 0.0.0.128 in peer configuration section in chan_sip or chan_iax. This patch takes the remainder before applying abs, so that bucket number is always in range. ASTERISK-25100 #close Reported by: Mark Petersen Change-Id: Id6981400ad526f47e10bcf7b847b62bd2785e899 2015-05-23 04:36 +0000 [554bd1e39c] Ivan Poddubny * res_pjsip_transport_websocket: Fix crash on receiving large SIP packets Incoming SIP packets larger than PJSIP_MAX_PKT_LEN were themselves truncated before passing to pjsip_tpmgr_receive_packet, but the length was passed unaltered, thus causing memory corruption and segfault. ASTERISK-25122 #close Change-Id: I608a6b6b7f229eacc33a0a7d771d18e27e5b08ab 2015-05-22 21:50 +0000 [0d266cbe02] Corey Farrell * Stasis: Fix unsafe use of stasis_unsubscribe in modules. Many uses of stasis_unsubscribe in modules can be reached through unload. These have been switched to stasis_unsubscribe_and_join. Some subscription callbacks do nothing, for these I've created a noop callback function in stasis.c. This is used by some modules that monitor MWI topics in order to enable cache, since the callback does not become invalid after dlclose it is safe to use stasis_unsubscribe on these, even during module unload. ASTERISK-25121 #close Change-Id: Ifc2549fbd8eef7d703c222978e8f452e2972189c 2015-05-22 12:22 +0000 [51ffed5e61] Matt Jordan * res/res_pjsip_pubsub: Note that 'dialog' is also a valid event type for RLS In addition to specifying lists of 'presence' and 'message-summary', users can also create lists of type 'dialog'. These should be treated in the same fashion as 'presence'. Change-Id: I583bb69cd9f88b0b29bf09ddaddeac4e84189f6e 2015-05-22 12:18 +0000 [7950b65e4f] Matt Jordan * res/res_pjsip_exten_state: Fix confusing NOTICE message When a SUBSCRIBE request is made to a dialplan hint that doesn't exist, the current NOTICE message informing users of this swaps the context and extension parameters. This can cause a bit of confusion. Thanks to CptBurger in #asterisk for helping to point this out. Change-Id: Ie584d1a58ae217385c87a450ca25b55ca0e36e43 2015-05-17 20:36 +0000 [5ac65ddfb4] Matt Jordan * res/ari: Register Stasis application on WebSocket attempt Prior to this patch, when a WebSocket connection is made, ARI would not be informed of the connection until after the WebSocket layer had accepted the connection. This created a brief race condition where the ARI client would be notified that it was connected, a channel would be sent into the Stasis dialplan application, but ARI would not yet have registered the Stasis application presented in the HTTP request that established the WebSocket. This patch resolves this issue by doing the following: * When a WebSocket attempt is made, a callback is made into the ARI application layer, which verifies and registers the apps presented in the HTTP request. Because we do not yet have a WebSocket, we cannot have an event session for the corresponding applications. Some defensive checks were thus added to make the application objects tolerant to a NULL event session. * When a WebSocket connection is made, the registered application is updated with the newly created event session that wraps the WebSocket connection. ASTERISK-24988 #close Reported by: Joshua Colp Change-Id: Ia5dc60dc2b6bee76cd5aff0f69dd53b36e83f636 2015-05-20 11:11 +0000 [60e2fbfe62] gtjoseph * res_pjsip: Refactor endpt_send_transaction (qualify_timeout) This patch refactors the transaction timeout processing to eliminate calling the lower level public pjsip functions and reverts to calling pjsip_endpt_send_request again. This is the result of me noticing a possible incompatibility with pjproject-2.4 which was causing contact status flapping. The original version of this feature used the lower level calls to get access to the tsx structure in order to cancel the transaction when our own timer expires. Since we no longer have that access, if our own timer expires before the pjsip timer, we call the callbacks and just let the pjsip transaction take it's own course. When the transaction ends, it discovers the callbacks have already been run and just cleans itself up. A few messages in pjsip_configuration were also added/cleaned up. ASTERISK-25105 #close Change-Id: I0810f3999cf63f3a72607bbecac36af0a957f33e Reported-by: George Joseph Tested-by: George Joseph 2015-05-20 00:45 +0000 [42476e6633] demon-ru * res_pjsip_outbound_registration: Check request URI for line. When an inbound call is received the To header is checked for the "line" option. Some remote servers will place this in the request URI instead. This adds an additional check for the option in the request URI. ASTERISK-25072 #close Reported by: Dmitriy Serov Change-Id: Id4e44debbb80baad623b914a88574371575353c8 2015-05-21 17:51 +0000 [e7edb59db6] Corey Farrell * res_mwi_external_ami: Use module version of AMI registration. Use ast_manager_register_xml for res_mwi_external_ami manager actions. This ensures the module is held open while any of the actions are being run. ASTERISK-25117 #close Reported by: Corey Farrell Change-Id: Iececfdc2da498b2c32b9e09042f5f12292007ac7 2015-05-21 13:05 +0000 [9d8a462356] Matt Jordan * ARI: Update version to 1.7.0 This patch updates the version of ARI to 1.7.0 to reflect the backwards compatible changes that will be introduced in 13.4.0. Change-Id: I6c36e6144da426412f25828a868e4df916bff60a 2015-05-20 20:53 +0000 [9b6e228419] Corey Farrell * Logger: Reset defaults before processing config. Reset options to default values before reloading config. This ensures that if a setting is removed or commented out of the configuration file it is unset on reload. ASTERISK-25112 #close Reported by: Corey Farrell Change-Id: Id24bb1fb0885c2c14cf8bd6f69a0c2ee7cd6c5bd 2015-05-20 19:05 +0000 [7fcf0a97b8] gtjoseph * app_playback: Suppress warnings on playback if channel hung up If a channel hangs up while an audio file is playing, there's no need to clutter up the logs with a warning so suppress it if ast_check_hangup returns true. Also, change warning to debug/2 in file.c if writing a frame fails. Same reasoning. Change-Id: I2e66191af3c5b6e951c98e8f1c3fe3cf2cf7ed89 Reported-by: George Joseph Tested-by: George Joseph 2015-05-14 15:21 +0000 [b1e8c0b9eb] Kevin Harwell * audiohook.c: Difference in read/write rates caused continuous buffer resets Currently, everytime a sample rate change occurs (on read or write) the associated factory buffers are reset. If the requested sample rate on a read differed from that of a write then the buffers are continually reset on every read and write. This has the side effect of emptying the buffer, thus there being no data to read and then write to a file in the case of call recording. This patch fixes it so that an audiohook_list's rate always maintains the maximum sample rate among hooks and formats. Audiohook sample rates are only overwritten by this value when slin native compatibility is turned on. Also, the audiohook sample rate can only overwrite the list's sample rate when its rate is greater than that of the list or if compatibility is turned off. This keeps the rate from constantly switching/resetting. ASTERISK-24944 #close Reported by: Ronald Raikes Change-Id: Idab4dfef068a7922c09cc631dda27bc920a6c76f 2015-05-19 13:01 +0000 [17d6ede337] Corey Edwards * main/sdp_srtp.c: allow SDP crypto tag to be up to 9 digits ASTERISK-24887 #close Reported by: Makoto Dei Tested by: tensai Change-Id: I6a96f572adb17f76b3acafe503a01c48eb5dd9bf 2015-05-13 09:55 +0000 [31cc24aad6] Matt Jordan * res/res_http_websocket: Add a pre-session established callback This patch updates http_websocket and its corresponding implementation with a pre-session established callback. This callback allows for WebSocket server consumers to be notified when a WebSocket connection is attempted, but before we accept it. Consumers can choose to reject the connection, if their application specific logic allows for it. As a result, this patch pulls out the previously private websocket_protocol struct and makes it public, as ast_websocket_protocol. In order to preserve backwards compatibility with existing modules, the existing APIs were left as-is, and new APIs were added for the creation of the ast_websocket_protocol as well as for adding a sub-protocol to a WebSocket server. In particular, the following new API calls were added: * ast_websocket_add_protocol2 - add a protocol to the core WebSocket server * ast_websocket_server_add_protocol2 - add a protocol to a specific WebSocket server * ast_websocket_sub_protocol_alloc - allocate a sub-protocol object. Consumers can populate this with whatever callbacks they wish to support, then add it to the core server or a specified server. ASTERISK-24988 Reported by: Joshua Colp Change-Id: Ibe0bbb30c17eec6b578071bdbd197c911b620ab2 2015-05-14 22:05 +0000 [f9114179e6] snuffy * chan_pjsip: Fix crash during off-nominal when no endpoint specified. Add missing return -1 when no endpoint name is specified. ASTERISK-25086 #close Reported by: snuffy Change-Id: I9de76c2935a1f4e3f0cffe97a670106f5605e89e 2015-05-14 18:01 +0000 [dd78ab42e4] gtjoseph * res_pjsip_config_wizard/config: Fix template processing The config wizard was always pulling the first occurrence of a variable from an ast_variable list but this gets the template value from the list instead of any overridden value. This patch creates ast_variable_find_last_in_list() in config.c and updates res_pjsip_config_wizard to use it instead of ast_variable_find_in_list. Now the overridden values, where they exist, are used instead of template variables. Updated test_config to test the new API. ASTERISK-25089 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: Ifa7ddefc956a463923ee6839dd1ebe021c299de4 2015-05-15 01:54 +0000 [091b436007] snuffy * cdr: Fix 'core show channel' CDR variable truncation. When the new Bridging API was implemented, the workspace variable changed to a malloc'd string, causing sizeof() to always be 8 (char). Revert back to stored on stack string for workspace. ASTERISK-25090 #close Change-Id: I51e610ae87371df771ce7693a955510efb90f8f7 2015-05-14 00:06 +0000 [6b7282ca40] Corey Farrell * Fix potential crash after unload of func_periodic_hook or test_message. These modules save a pointer to the context they create on load, and use that pointer to destroy the context at unload. It is not safe to save this pointer, it is replaced during load of pbx_config, pbx_lua or pbx_ael. This change causes the modules to pass NULL to ast_context_destroy, a safer way to perform the unregistration since it does not use a pointer that could become invalid. ASTERISK-25085 #close Reported by: Corey Farrell Change-Id: I6a00ec8e38046058f97dc703e1adcde9bf517835 2015-05-13 15:41 +0000 [02c5130589] Jonathan Rose * Message.c: Clear message channel frames on cleanup The message channel is a special channel that doesn't actually process frames. However, certain actions can cause frames to be placed in the channel's read queue including the Hangup application which is called on the channel after each message is processed. Since the channel will continually be reused for many messages, it's necessary to flush these frames at some point. ASTERISK-25083 #close Reported by: Jonathan Rose Change-Id: Idf18df73ccd8c220be38743335b5c79c2a4c0d0f 2015-05-12 17:45 +0000 [d49d64b79c] Jonathan Rose * app_voicemail: fix moving when old messages full When completing voicemail playback of a message in the 'INBOX', the message gets moved to the 'Old' messages folder. Without this patch, if the 'Old' folder is already at its set limit, then the 'INBOX' message will simply be deleted. With this patch, the flag to delete the message will be removed if the save_to_folder function indicates that the message could not be moved due to a full folder. ASTERISK-25082 #close Reported by: Jonathan Rose Review: https://gerrit.asterisk.org/#/c/448/ Change-Id: I2be440a09f42e2d06d50975c40d1ad7f836ecb3f 2015-05-04 20:11 +0000 [9b13536fed] Rodrigo Ramírez Norambuena * main/manager.c: Bugfix sort action_manager by alphabetically Fix the alphabetic order added on ast_manager_register_struct. The order for struct manager_action added is not working, this change fixes the problem. Change-Id: I149da0cd06c3c4445d7516cc303358e9f26f8b4b 2015-05-08 18:01 +0000 [e67e8d5c7f] Alexandre Fournier * res_config_mysql: Fix broken column type checking MySQL configuration engine contains a bug in require_mysql(). This function is used for column type checking in tables. This bug only affects DATETIME, DATE and FLOAT types. It came from mixing the first condition (switch-case-like if/then/else), to check the expected column type, with the second condition, to check the actual column type against the expected column type. Both conditions must be checked separately in order to avoid the execution of the wrong block. ASTERISK-18252 #comment This patch might fix the issue Reported by: Gareth Blades ASTERISK-25041 #close Reported by: Alexandre Fournier Tested by: Alexandre Fournier Change-Id: I0b8bf7e68ab938be8e6525a249260cb648cb0bfa 2015-05-10 07:37 +0000 [16f602f5c2] Yousf Ateya * res_rtp_asterisk: Correction for the limit which detects that a packet is DTLS. First byte of DTLS packet shall be in range 20-63, not 20-64. Refer to RFC https://tools.ietf.org/html/rfc5764#section-5.1.2 for correct values. Change-Id: Iae6fa0d72b37c36a27fe40686e0ae6fba3afec31 2015-05-12 17:34 +0000 [c780b6e431] Richard Mudgett * chan_dahdi/sig_pri: Fix crash on ISDN call hangup collision. If an ISDN call is hungup by both sides at the same time a crash could happen. * Added missing NULL checks for the owner channel after calling pri_queue_pvt_cause_data() in two places. Code after those calls need to check the owner channel pointer for NULL before use because pri_queue_pvt_cause_data() needs to do deadlock avoidance to lock the owner and the owner may get hung up. ASTERISK-21893 #close Reported by: Alexandr Gordeev Change-Id: Ica3e266ebc7a894b41d762326f08653e1904bb9a 2015-05-10 02:26 +0000 [6627de830b] Sebastian Kemper * General: Fix recent menuselect-related cross compile regression MAKE_MENUSELECT currently sets CC to CC, which is the compiler for the target platform. But menuselect is to be run on the build system, so BUILD_CC needs to be used instead - like it was in the past, before the recent changes (https://reviewboard.asterisk.org/r/4370/). This is the patch for ASTERISK-25074. ASTERISK-25074 #close Reported by: Sebastian Kemper Tested by: Sebastian Kemper Change-Id: I8a2b1fc5deb6ad2b80f49baca35b1b13d468ebf8 2015-05-05 15:32 +0000 [637c8f065e] gtjoseph * sorcery: Add API to insert/remove a wizard to/from an object type's list Currently you can 'apply' a wizard to an object type but the wizard always goes at the end of the object type's wizard list. This patch adds a new ast_sorcery_insert_wizard_mapping function that allows you to insert a wizard anyplace in the list. I.E. You could add a caching wizard to an object type and place it before all wizards. ast_sorcery_get_wizard_mapping_count and ast_sorcery_get_wizard_mapping were added to allow examination of the mapping list. ast_sorcery_remove_mapping was added to remove a mapping by name. As part of this patch, the object type's wizard list was converted from an ao2_container to an AST_VECTOR_RW. A new test was added to test_sorcery for this capability. ASTERISK-25044 #close Change-Id: I9d2469a9296b2698082c0989e25e6848dc403b57 2015-05-12 01:31 +0000 [3cdb7950f0] Corey Farrell * Fix processing of asterisk.conf debug=yes. The code which reads asterisk.conf supports processing the debug option with ast_true, but ast_true returns -1. This causes debug to still be off, convert to 1 so debug will be on as requested. ASTERISK-25042 Reported by: Corey Farrell Change-Id: I3c898b7d082d914b057e111b9357fde46bad9ed6 2015-05-01 23:43 +0000 [6553a00770] Rodrigo Ramírez Norambuena * cdr_pgsql: Use PQescapeStringConn for escaping names. Use function PQescapeStringConn for escaping the name of the table and schema instead of doing it manually. Change-Id: I6709165e2d00463e9c813d24f17830ad4910b599 2015-05-09 16:58 +0000 [ea917fefaf] gtjoseph * vector: Add REMOVE, ADD_SORTED and RESET macros Based on feedback from Corey Farrell and Y Ateya, a few new macros have been added... AST_VECTOR_REMOVE which takes a parameter to indicate if order should be preserved. AST_VECTOR_ADD_SORTED which adds an element to a sorted vector. AST_VECTOR_RESET which cleans all elements from the vector leaving the storage intact. Change-Id: I41d32dbdf7137e0557134efeff9f9f1064b58d14 2015-05-11 07:07 +0000 [d5864a358c] Ivan Poddubny * pbx/pbx_spool: Fix issue when call files were executed too early pbx_spool used to delete/move the call file upon successful outgoing call completion, but did not delete it from in-memory list of files (dirlist, used only when compiled with inotify/kqueue support). That resulted in an extra attempt to process that filename after retrytime seconds. Then, if a new file with the same name appears that is scheduled in future further than the completed one plus its retrytime, then it gets executed earlier than expected. This patch fixes remove_from_queue function to also remove the entry from the dirlist. ASTERISK-17069 #close Reported by: Jeremy Kister ASTERISK-24442 #close Reported by: tootai Change-Id: If9ec9b88073661ce485d6b008fd0b2612e49a28b 2015-05-08 14:47 +0000 [4dbd4021c9] Rusty Newton * configs/basic-pbx: Modified main IVR to play new Allison prompt. The main IVR was playing demo-congrats. I've switched it over to the basic-pbx-ivr-main file that we added in core sounds 1.4.27. This prompt has Allison prompting the user with the actual IVR menu. ASTERISK-24892 #close Change-Id: Ifb749616ff8e156a1031ddaddfcc9244767a095d 2015-05-08 10:39 +0000 [613a461c3d] Sean Bright * res_rtp_asterisk: Issue ERROR if res_srtp is not found. While trying to get WebRTC working with chan_pjsip, I was running into the following error: Attempted to set an invalid DTLS-SRTP configuration on RTP instance... Josh helpfully pointed out that res_srtp.so might not be loaded, and sure enough, it wasn't. This patch adds a ERROR indiciating as much to hopefully help others having a similar problem. Change-Id: I13aa477b47b299876728a21b130998a0ea6cd19f 2015-05-07 17:49 +0000 [394fcb5eab] Rusty Newton * sounds: Add Swedish sounds to Makefile and XML Added the necessary lines to the Makefile and sounds.xml so we'll have the Swedish sounds in all available formats in menuselect. See also: Swedish sounds were added into the core sounds release 1.4.27. ASTERISK-24744 #close Reported by: Tove Hjelm Tested by: Rusty Newton Change-Id: Ib6f4fd177afd1667b2402735034001d4d055a908 2015-05-05 11:35 +0000 [2115f11b54] Alexander Traud (License 6520) * tcptls: Avoiding ERR_remove_state in OpenSSL. ERR_remove_state was deprecated with OpenSSL 1.0.0 and was replaced by ERR_remove_thread_state. ERR_load_SSL_strings and ERR_load_BIO_strings were called by SSL_load_error_strings already and got removed. These changes allow OpenSSL forks like BoringSSL to be used with Asterisk. ASTERISK-25043 #close Reported by: Alexander Traud patches: asterisk_with_BoringSSL.patch uploaded by Alexander Traud (License 6520) Change-Id: If1c0871ece21a7e0763fafbd2fa023ae49d4d629 (cherry picked from commit 247fef66537b59649e7571d64e2c574a106dbd65) 2015-05-07 14:54 +0000 [5392e970d0] gtjoseph * doc: Make progdocs play nice with git Moved contrib/asterisk-ng-doxygen to doc/asterisk-ng-doxygen.in Changed /Makefile to copy asterisk-ng-doxygen.in to asterisk-ng-doxygen then modify it with version instead of modifying asterisk-ng-doxygen directly. Updated clean targets as well. Updated /.gitignore and doc/.gitignore. Change-Id: I38712d3e334fa4baec19d30d05de8c6f28137622 2015-05-04 14:43 +0000 [608f0a94ee] Ivan Poddubny * contrib/editors: Fix vim syntax highlighting of comments in config files * Added a lookbehind to one-line comment matcher to skip escaped semicolons. * Added support for block comments. Change-Id: Id17dfaeda8ed4be572e8107a0c010066584aaee7 2015-05-06 13:24 +0000 [d649d682c4] Joshua Colp * res_pjsip_exten_state: Fix race condition between sending NOTIFY and termination The res_pjsip_exten_state module currently has a race condition between processing the extension state callback from the PBX core and processing the subscription shutdown callback from res_pjsip_pubsub. There is currently no synchronization between the two. This can present a problem as while the SIP subscription will remain valid the tree it points to may not. This is in particular a problem as a task to send a NOTIFY may get queued which will try to use the tree that may no longer be valid. This change does the following to fix this problem: 1. All access to the subscription tree is done within the task that sends the NOTIFY to ensure that no other thread is modifying or destroying the tree. This task executes on the serializer for the subscriptions. 2. A reference to the subscription serializer is kept to ensure it remains valid for the lifetime of the extension state subscription. 3. The NOTIFY task has been changed so it will no longer attempt to send a NOTIFY if the subscription has already been terminated. ASTERISK-25057 #close Reported by: Matt Jordan Change-Id: I0b3cd2fac5be8d9b3dc5e693aaa79846eeaf5643 2015-05-05 20:22 +0000 [5f9aea8e3c] gtjoseph * vector: Additional enhancements and fixes After using the new vector stuff for real I found... A bug in AST_VECTOR_INSERT_AT that could cause a seg fault. The callbacks needed to be closer to ao2_callback in behavior WRT to CMP_MATCH and CMP_STOP behavior and the ability to return a vector of matched entries. A pre-existing issue with APPEND and REPLACE was also fixed. I also added a new macro to test.h that acts like ast_test_validate but also accepts a return code variable and a cleanup label. As well as printing the error, it sets the rc variable to AST_TEST_FAIL and does a goto to the specified label on error. I had a local version of this in test_vector so I just moved it. ASTERISK-25045 Change-Id: I05e5e47fd02f61964be13b7e8942bab5d61b29cc 2015-05-04 17:28 +0000 [68513e00f7] Kevin Harwell * res_stasis_snoop: Spying on a single direction continually increases CPU Creating a snoop channel in ARI and spying only on a single direction (in or out) results in CPU utilization continually increasing until the CPU is fully consumed. This occurs because frames are being put in the opposing direction's slin factory queue, but not being removed. Fixed the problem by always reading and disposing of frames from the opposite queue of the direction selected. ASTERISK-24938 #closes Change-Id: I935bfd15f1db958f364d9d6b3b45582c0113dd60 2015-05-06 16:00 +0000 [904f5d98f6] Richard Mudgett * chan_dahdi: Improve force_restart_unavailable_chans option description. ASTERISK-25034 Reported by: Richard Mudgett Change-Id: I1ff8f02124d2f4abd632a050da52c64285bb7f30 2015-05-05 18:17 +0000 [be1260a35f] Richard Mudgett * features: Fix crash when transferee hangs up during DTMF attended transfer. A crash happens with this sequence of steps: 1) Party A is connected to party B. 2) Party B starts a DTMF attended transfer. 3) Party A hangs up while party B is dialing party C. When party A hangs up the bridge that party A and party B are in is dissolved and party B is kicked out of the bridge. When party B finishes dialing party C he attempts to move to the new bridge with party C. Since party B is no longer in a bridge the attempted move dereferences a NULL bridge_channel pointer and crashes. * Made the hold(), unhold(), ringing(), and the bridge_move() functions tolerant of the channel not being in a bridge. The assertion that party B is always in a bridge is not true if the bridged peer of party B hangs up and dissolves the bridge. Being tolerant of not being in a bridge allows the peer hangup stimulus to be processed by the FSM. * Made the bridge_move() function return void since where the return value for a failed move was checked generated a FSM coding ERROR message for a normal off-nominal condition. * Eliminated most uses of RAII_VAR in bridge_basic.c. ASTERISK-25003 #close Reported by: Artem Volodin Change-Id: Ie2c1b14e5e647d4ea6de300bf56d69805d7bcada 2015-05-05 15:40 +0000 [8b0f85ac06] gtjoseph * test_vector: Fix build breakage caused by ASTERISK_REGISTER_FILE My 13 version of test_vector had an ASTERISK_REGISTER_FILE() macro call at the top which is only supported in master. Once removed builds are successful. Change-Id: I7cac8b669bed6de543bbf4e2eec3cffc9741acdd 2015-05-05 14:48 +0000 [87263b47b5] Ivan Poddubny * app_queue: Fix queue_log EXITWITHTIMEOUT containing only 1 parameter This patch fixes EXITWITHTIMEOUT queue_log entry to always come with 3 parameters: position, original position and waiting time. ASTERISK-25038 #close Reported by: Etienne Lessard Change-Id: I0c62045922e26bee2125e93aee1dee17eee79618 2015-05-05 09:47 +0000 [366ea63438] Corey Farrell * res_ari_bridges: Add missing dependencies. Missed this module in the previous commit. res_ari_bridges uses symbols from res_stasis_playback and res_stasis_recording. ASTERISK-25027 #close Reported by: Corey Farrell Change-Id: I90bf756abd25adfc4920d2869ebe7feb636b8c5f 2015-05-05 09:27 +0000 [69ae8cf0a4] Corey Farrell * pbx_config: Register manager actions with module version of macro. Switch manager actions in pbx_config to use the registration macro that passes the module pointer, allowing pbx_config reference to be bumped while the manager actions run. ASTERISK-25061 #close Reported by: Corey Farrell Change-Id: I422c50dd74814616ac10c5e9c6598a0b1bc2c44e 2015-05-04 12:16 +0000 [181ae3b8d9] Joshua Colp * stasis: Fix dial masquerade datastore lifetime A recent change went into Asterisk which added reference counts to the channels stored in a dial masquerade datastore. Unfortunately this included a reference to the caller in a dialing operation. While all of the dialed targets have the datastore removed from them upon dialing completion this did not occur for the caller, causing it to have a reference to itself that could go never go away (as it depended on the destruction of the datastore which only happened when the channel was destroyed). This resulted in the caller channel remaining on the system despite it having hung up. This change does the following to fix this issue: 1. The dial masquerade datastore is now removed from the caller upon dialing completion, just like the dialed targets. 2. Upon destruction of the caller all the dialed targets are also removed from the dial masquerade datastore (just in case). 3. The reference to the caller has been removed as it should not be possible for the datastore to now be valid/useful after the lifetime of the caller has ended. ASTERISK-25025 #close Change-Id: I1ef4ca5ca04980028604cc2af5d2992ac3431b3f 2015-05-01 19:25 +0000 [7a7e9733c2] gtjoseph * vector: Traversal, retrieval, insert and locking enhancements Renamed AST_VECTOR_INSERT to AST_VECTOR_REPLACE because it really does replace not insert. The few users of AST_VECTOR_INSERT were refactored. Because these are macros, there should be no ABI compatibility issues. Added AST_VECTOR_INSERT_AT that actually inserts an element into the vector at a specific index pushing existing elements to the right. Added AST_VECTOR_GET_CMP that can retrieve from the vector based on a user-provided compare function. Added AST_VECTOR_CALLBACK function that will execute a function for each element in the vector. Similar to ao2_callback and ao2_callback_data functions although the vector callback can take a variable number of arguments. This should allow easy migration to a vector where a container might be too heavy. Added read/write locked vector and lock manipulation macros. Added unit tests. ASTERISK-25045 #close Change-Id: I2e07ecc709d2f5f91bcab8904e5e9340609b00e0 2015-05-03 13:55 +0000 [040d2f8558] Corey Farrell * main/test.c: Add test to verify there were no registration errors. This adds a test that will fail if any test failed to register. Also fail if any test registration produced a warning about missing a leading or trailing slash. ASTERISK-25053 #close Reported by: Corey Farrell Change-Id: I93e50b8fcbcfa7f1f5b41b2c44a51685c09529c3 2015-04-21 11:52 +0000 [3dcec04ab5] Martin Tomec * res_odbc: Use negative connection cache for all connections Apply the negative connection cache setting to all connections, even those that are not pooled. This ensures that the connection will not be re-established before the negative connection cache time is met. ASTERISK-22708 #close Change-Id: I431cc2e8584ab0b6908b3523d0a0e18c9a527271 2015-05-03 21:03 +0000 [f38066fcad] Corey Farrell * Format Interfaces: Prevent unload except by shutdown. Format interfaces cannot be unregistered, so the modules that provide them need to be held open except by shutdown. ASTERISK-25054 #close Reported by: Corey Farrell Change-Id: Iadbd9675bf0d30b8fded5a739b163db3ea2db8f3 2015-05-03 20:28 +0000 [e76a6a97bf] Matt Jordan * contrib/ast-db-manage: Add Postgres ENUM type support in auto DTMF mode update The upgrade script for auto DTMF mode (31cd4f4891ec) added in 88b0fa7755 failed to add ENUM support for Postgres databases. This requires a specific import from the sqlalchemy.dialects.postgresql package. This patch corrects this error, which allows for Postgres update scripts to be generated. ASTERISK-24706 Change-Id: I4742ac8efa533cd6f18e0bdd907b339a9aedf015 2015-05-01 19:50 +0000 [92120247e9] D Tucny * term: send proper reset sequence when black background is forced When using the force black background command-line option or configuration option an invalid reset sequence is sent following a coloured output item in the CLI, the result is that the colour is not 'turned off' and continues until the next non-default coloured text output. A reset sequence is already defined in term.c, but the ast_term_reset function doesn't use it, instead building it's own invalid sequence and returning that. This patch changes that behaviour, removing the building of a reset sequence and instead using the pre-built constant 'enddata' which is a suitable reset sequence for this purpose. ASTERISK-24896 #close Reported by: Dan Tucny Change-Id: I56323899123ae3264900389cae1f5b252aa3bf43 2015-05-02 18:58 +0000 [ad6ea29697] Corey Farrell * Remove unneeded uses of optional_api providers. A few cases exist where headers of optional_api provders are included but not needed. This causes unneeded calls to ast_optional_api_use. * Don't include optional_api.h from sip_api.h. * Move 'struct ast_channel_monitor' to channel.h. * Don't include monitor.h from chan_sip.c, channel.c or features.c. The move of struct ast_channel_monitor is needed since channel.c depends on it. This has no effect on users of monitor.h since channel.h is included from monitor.h. ASTERISK-25051 #close Reported by: Corey Farrell Change-Id: I53ea65a9fc9693c89f8bcfd6120649bfcfbc3478 2015-04-30 02:07 +0000 [525c8c8689] Rodrigo Ramírez Norambuena * include/asterisk/channel.h: Fix typo Change-Id: Ie584b85e16a94c255e60d0b1732ef9686464fef3 2015-05-02 02:15 +0000 [63196a8256] Corey Farrell * res_pjsip_dlg_options: Fix MODULEINFO section. Removed the extra space before "MODULEINFO" in res_pjsip_dlg_options. This extra space prevented any of the dependencies from being seen by menuselect, so building with default options would fail if PJSIP was not installed. This also makes the tool that extracts information for menuselect tolerant of multiple spaces in the future. ASTERISK-25033 #close Reported by: Peter Whisker Change-Id: Iccd54846f70c4a7a50cb5bf70b7bb5cb4bab3698 2015-04-29 03:03 +0000 [ac1f0090eb] Corey Farrell * Build System: Prevent unneeded changes to asterisk/buildopts.h. * Add AST_DEVMODE to BUILDOPTS * Remove CFLAGS that do not effect ABI from BUILDOPTS. * Use BUILDOPTS to generate AST_BUILDOPT_SUM. * Remove loop that defined AST_MODULE_* These changes ensure that only ABI effecting options are considered for AST_BUILDOPT_SUM. This also reduces unneeded full system rebuilds caused by enabling or disabling one module that another is dependent on. ASTERISK-25028 Reported by: Corey Farrell Change-Id: I2c516d93df9f6aaa09ae079a8168c887a6ff93a2 2015-05-01 13:22 +0000 [5875bf183c] Corey Farrell * Astobj2: Fix initialization order of refdebug and AO2_DEBUG. This ensures that refdebug is initialized before AO2_DEBUG if both are enabled, since AO2_DEBUG allocates a container. This change also makes AO2_DEBUG initialization critical, a failure will abort Asterisk startup. This is needed since the failure would be caused by reg_containers allocation failure, and that would result in a segmentation fault by ao2_container_register later in startup. ASTERISK-25048 #close Reported by: Corey Farrell Change-Id: I9a243ea3fc5653b48b931ba6d61971cb2e530244 2015-04-29 14:49 +0000 [1b19c15f17] Matt Jordan * main/pbx: Improve performance of dialplan reloads with a large number of hints The PBX core maintains two hash tables for hints: a container of the actual hints (hints), along with a container of devices that are watching that hint (hintdevices). When a dialplan reload occurs, each hint in the hints container is destroyed; this requires a lookup in the container of devices to find the device => hint mapping object. In the current code, this performs an ao2_callback, iterating over each of the device to hint objects in the hintdevices container. For a large number of hints, this is extremely expensive: dialplan reloads with 20000 hints could take several minutes in just this phase. This patch improves the performance of this step in the dialplan reloads by caching which devices are watching a hint on the hint object itself. Since we don't want to create a circular reference, we just cache the name of the device. This allows us to perform a smarter ao2_callback on the hintdevices container during hint removal, hashing on the name of the device and returning an iterator to the matching names. The overall performance improvement is rather large, taking this step down to a number of seconds as opposed to minutes. In addition, this patch also registers the hint containers in the PBX core with the astobj2 library. This allows for reasonable debugging to hash collisions in those containers. ASTERISK-25040 #close Reported by: Matt Jordan Change-Id: Iedfc97a69d21070c50fca42275d7b3e714e59360 2015-04-30 15:54 +0000 [3efe0df044] Corey Farrell * Sample Configs: Fix syntax error in pjsip.conf The sample pjsip.conf has a few comment lines that are missing the semicolons at the start of the comment, causing the config to fail load. Change-Id: I776a38c916a7df7ee3e072fd0b21dbf4cc457352 2015-04-30 15:20 +0000 [077979618b] Mark Michelson * Prevent potential crash on blond transfer. Scenario: Alice calls Bob. Bob performs a blond transfer to Carol. Carol rejects the incoming call (or some other immediate circumstance causes Carol not to answer the call) What occurs in this case is that when the bridge between Alice and Bob breaks, Alice is told to masquerade into Bob's channel that had placed the call to Carol. The actual masquerade goes down without a hitch. However, a channel fixup callback that attempts to publish dial events over Stasis has a crash. The reason for this crash is that the datastore on Bob's channel that placed the outbound call to Carol only had a bare pointer to Carol's channel. Since Carol rejected the incoming call, Carol's channel has been hung up and freed, meaning accessing her channel results in a crash. The fix here is simple. The dial fixup code has been altered to hold references to the involved channels and to drop those references when freeing data. ASTERISK-25025 #close Reported by Chet Stevens Change-Id: I54eedda207b8ec7a69263353b43abe5746aea197 2015-04-30 14:09 +0000 [4b8cddfb36] Mark Michelson * res_pjsip_outbound_authenticator_digest: Add missing outbound authenticator callback. The Asterisk 13 version of the fix for outbound registration was missing a key component that set the outbound authenticator's callback that creates an authenticated request based on an old request. This was picked up by some outbound registration tests failing in the testsuite. Change-Id: I5ca9379698c606da36bc38eaffccedaf64211ce3 2015-04-30 13:42 +0000 [415a0d0745] Joshua Colp * res_ari_device_states: Fix dependency on res_stasis_device_state. The res_ari_device_states module depends on res_stasis_device_state, not res_stasis_device_states. Change-Id: I26e02ad37f9e36bcc859867e2fad1b90452ec3de 2015-04-29 14:29 +0000 [d3c310a28c] Richard Mudgett * chan_dahdi: Add the chan_dahdi.conf force_restart_unavailable_chans option. Some telco switches occasionally ignore ISDN RESTART requests. The fix for ASTERISK-19608 added an escape clause for B channels in the restarting state if the telco ignores a RESTART request. If the telco fails to acknowledge the RESTART then Asterisk will assume the telco acknowledged the RESTART on the second call attempt requesting the B channel by the telco. The escape clause is good for dealing with RESTART requests in general but it does cause the next call for the restarting B channel to be rejected if the telco insists the call must go on that B channel. chan_dahdi doesn't really need to issue a RESTART request in response to receiving a cause 44 (Requested channel not available) code. Sending the RESTART in such a situation is not required (nor prohibited) by the standards. I think chan_dahdi does this for historical reasons to deal with buggy peers to get channels unstuck in a similar fashion as the chan_dahdi.conf resetinterval option. * Add the chan_dahdi.conf force_restart_unavailable_chans compatability option that when disabled will prevent chan_dahdi from trying to RESTART the channel in response to a cause 44 code. ASTERISK-25034 #close Reported by: Richard Mudgett Change-Id: Ib8b17a438799920f4a2038826ff99a1884042f65 2015-04-30 06:38 +0000 [7f611fa0e8] Rodrigo Ramírez Norambuena * cdr/cdr_csv.c: Add a new option to enable columns added in Asterisk 1.8 This patch adds a new option to cdr.conf, 'newcdrcolumns', that will handle CDR columns added in Asterisk 1.8. The columns are: * peeraccount * linkedid * sequence When enabled, the columns in the database entry will be populated with the data from the CDR. ASTERISK-24976 #close Change-Id: I51a57063f4ae5e194a9d933a8df45dc8a4534f0b 2015-04-30 06:04 +0000 [e332c7ed5e] Joshua Colp * res_pjsip_outbound_registration: Fix double unref on error return. When the PJSIP pjsip_regc_send function is invoked and an error status returned the caller currently decrements the reference count of the client state that it just incremented, assuming the registration callback would not have been invoked. In practice this is not correct. If the failure happens after the transaction has been set up the callback will still be invoked. This will cause the reference count to be incorrectly decremented twice, once by the registration callback and second by the caller of pjsip_regc_send. This change makes it so that whether the callback is invoked or not is known by the caller of pjsip_regc_send. Depending on this it can know whether it is responsible for decrementing the reference count of the client state or not. ASTERISK-25037 #close Reported by: Joshua Colp Change-Id: I749dc12f3a22115c49c5d7d95ff42a5fa45319de 2015-04-20 13:03 +0000 [9c3ed42875] Diederik de Groot * Update configure.ac/Makefile for clang Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which checks compiler requirements for RAII: gcc: -fnested-functions support clang: -fblocks (and if required -lBlocksRuntime) The original check was implemented in configure.ac and now has it's own file. This function also sets C_COMPILER_FAMILY to either gcc or clang for use by makefile Created autoconf/ast_check_strsep_array_bounds.m4 (contains AST_CHECK_STRSEP_ARRAY_BOUNDS): which checks if clang is able to handle the optimized strsep & strcmp functions (linux). If not, the standard libc implementation should be used instead. Clang + the optimized macro's work with: strsep(char *, char []), but not with strsepo(char *, char *). Instead of replacing all the occurences throughout the source code, not using the optimized macro version seemed easier See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h': llvm-comment: Normally, this array-bounds warning are suppressed for macros, so that unused paths like the one that accesses __s1[3] are not warned about. But if you preprocess manually, and feed the result to another instance of clang, it will warn about all the possible forks of this particular if statement. Instead of switching of this optimization, another solution would be to run the preproces- sing step with -frewrite-includes, which should preserve enough information so that clang should still be able to suppress the diag- nostic at the compile step later on. See also "https://llvm.org/bugs/show_bug.cgi?id=20144" See also "https://llvm.org/bugs/show_bug.cgi?id=11536" Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning suppressions: -Wno-unused-value -Wno-parentheses-equality In an earlier review (reviewboard: 4550 and 4554), they were deemed a nuisace and less than benefitial. configure.ac: Added AST_CHECK_RAII() see earlier Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier Removed moved content ASTERISK-24917 Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb 2015-04-29 16:15 +0000 [d4e207e27e] Matt Jordan * main/rtp_engine: Fix DTLS double-free introduced by 0b6410c4f8 The patch in 0b6410c4f8 did correctly fix a memory leak of the DTLS structures in the RTP engine. However, when a 'core reload' is issued, a double free of the memory pointed to by the char *'s in the DTLS configuration struct can occur, as ast_rtp_dtls_cfg_free does not set the pointers to NULL when they are freed. This patch sets those pointers to NULL, preventing a second call to ast_rtp_dtls_cfg_free from corrupting memory. ASTERISK-25022 Change-Id: I820471e6070a37e3c26f760118c86770e12f6115 2015-04-29 13:05 +0000 [3fb6daeb55] Kevin Harwell * res_fax: allow 2400 transmission rate according to v.27ter standard A previous set of patches (see: ASTERISK-22790 & ASTERISK-23231) made it so a v.27 modem was not allowed to have a minimum transmission rate of 2400 bits per second. This reverts all or some of those patches since according to the v.27ter standard a rate of 2400 bits per second is also supported. One of the original patches also added 9600 bits per second support for v.27. This patch also removes that since v.27ter only supports 2400/4800 bits per second. Also, since Asterisk specifically supports v.27ter the enum was renamed to better reflect this. ASTERISK-24955 #close Reported by: Matt Jordan Change-Id: I4b9dfb6bf7eff08463ab47ee1a74224f27cae733 2015-04-29 10:46 +0000 [49ef81c15c] Joshua Colp * res_sorcery_config: Fix build issue due to syntax error. Change-Id: Ic8322f04e37842848ad72cf2871bd0378f67c4ac 2015-04-28 00:29 +0000 [3278fe5327] Ashley Sanders * chan_pjsip: Creating Channel Causes Asterisk to Crash When Duplicate AOR Sections Exist in pjsip.conf This patch modifies the current loading strategy of the pjsip configuration. If duplicate sections (e.g. sections containing the same [id/type]) are defined in [pjsip.conf], the loader will consider the configuration for the given type as invalid when the duplicate section is encountered. The entire configuration (including what was previously loaded) for the duplicate [id/type] sections will be rejected and destroyed, an error message is logged and the load processing for the given stops. ASTERISK-24996 Reported By: Ashley Sanders Change-Id: I35090ca4cd40f1f34881dfe701a329145c347aef 2014-11-04 06:03 +0000 [89f6719f7a] Joshua Colp * res_pjsip_outbound_registration: Add virtual line support. Virtual line support establishes a relationship between messages related to an outbound registration and a local endpoint. This is accomplished by attaching a parameter to the Contact of the outbound registration and looking for it on any received requests. If the parameter exists and can be matched to an outbound registration the configured endpoint is associated with the request. ASTERISK-24949 #close Reported by: Joshua Colp Change-Id: I7df909d2625479110a83fdd354c21ac539e8615d 2015-04-29 06:39 +0000 [d61f03c4f9] Corey Farrell * ARI: Fix missing dependencies. ARI modules that are generated by 'make ari-stubs' are all dependent on res_ari_model. Additionally some of the same modules depend on one or more res_stasis_* modules. ASTERISK-25027 #close Reported by: Corey Farrell Change-Id: I8e07fe7e81fedacb87232f2b6f8b5f47927b4153 2015-04-29 06:26 +0000 [3e4624ad21] Corey Farrell * res_pjsip: Remove incorrect MODULEINFO from presence_xml.c. Remove incorrect MODULEINFO block and unneeded header includes from presence_xml.c. ASTERISK-25027 Reported by: Corey Farrell Change-Id: I977c609ab9d1fe05373027c4138900f6985990eb 2015-04-29 06:17 +0000 [fed9faab8d] Corey Farrell * Git Migration: Create doc/rest-api when needed. Create the directory './doc/rest-api' at the start of 'make ari-stubs' to prevent an error when documentation is generated. The directory is also added to git ignores. ASTERISK-25027 Reported by: Corey Farrell Change-Id: Iaccc7f0138501c23aa78feaca2f3cce9e68cbc1b 2015-04-29 05:17 +0000 [df23c8a86b] Joshua Colp * res_pjsip_outbound_registration: Fix build due to removal of transaction. Change-Id: I7a8a7beec3334cec304943f2dd7597eabe2e3150 2015-04-27 16:56 +0000 [e39bd6ba46] Mark Michelson * res_pjsip_outbound_registration: Don't fail on delayed processing: 13. This is the Asterisk 13 version of a change to master that allows for registration responses to be processed successfully potentially after the original transaction has timed out. The main difference between this and the master change is that the master version has API changes that are unacceptable for 13. For 13, this is worked around by adding a new API call that the outbound registration code uses instead. The following is the text from the master version of this commit: Odd behaviors have been observed during outbound registrations. The most common problem witnessed has been one where a request with authentication credentials cannot be created after receiving a 401 response. Other behaviors include apparently processing an incorrect SIP response. Inspecting the code led to an apparent issue with regards to how we handle transactions in outbound registration code. When a response to a REGISTER arrives, we save a pointer to the transaction and then push a task onto the registration serializer. Between the time that we save the pointer and push the task, it's possible for the transaction to be destroyed due to a timeout. It's also possible for the address to be reused by the transaction layer for a new transaction. To allow for authentication of a REGISTER request to be authenticated after the transaction has timed out, we now also hold a reference to the original REGISTER request instead of the transaction. The function for creating a request with authentication has been altered to take the original request instead of the transaction where the original request was sent. ASTERISK-25020 Reported by Mark Michelson Change-Id: If1ee5f601be839479a219424f0358a229f358f7c 2015-04-27 14:44 +0000 [1bf008fc76] Mark Michelson * res_pjsip_outbound_registration: Add debugging messages. When problems occur regarding outbound registrations, it currently is difficult to debug. Most off-nominal paths had warning messages, but sometimes we want to know what's going on before hitting the off-nominal path. This patch adds lots of debugging output that should give a clearer picture of what is happening with regards to outbound registrations. ASTERISK-25020 Reported by Mark Michelson Change-Id: I577bde7860be0a6c872b5bcb4d5047340bf45d45 2015-04-28 05:38 +0000 [0b6410c4f8] Steve Davies * res_rtp_asterisk: Resolve 2 discrete memory leaks in DTLS ao2 ref leak in res_rtp_asterisk.c when a DTLS policy is created. The resources are linked into a table, but the original alloc refs are never released. ast_strdup leak in rtp_engine.c. If ast_rtp_dtls_cfg_copy() is called twice on the same destination struct, a pointer to an alloc'd string is overwritten before the string is free'd. ASTERISK-25022 Reported by: one47 Change-Id: I62a8ceb8679709f6c3769136dc6aa9a68202ff9b 2015-04-27 12:11 +0000 [99fb87ae13] gtjoseph * res_pjsip: Fix SEGV on pending-qualify contacts Permanent contacts that hadn't been qualified yet were missing their contact_status entries causing SEGVs when running CLI commands. This patch makes sure that contact_statuses are created for both dynamic and permanent contacts when they are created. It also adds checks in the CLI code to make sure there's a contact_status, just in case. ASTERISK-25018 #close Reported-by: Ivan Poddubny Tested-by: Ivan Poddubny Tested-by: George Joseph Change-Id: I3cc13e5cedcafb24c400368b515b02d7fb81e029 2015-04-15 18:55 +0000 [d5dd43856e] Rodrigo Ramírez Norambuena * cdr/cdr_odbc.c: Added to record new columns add on CDR 1.8 Asterisk Version Add new column to INSERT new columns added in cdr 1.8 version. The columns are: * peeraccount * linkedid * sequence This feature is configurable in cdr_odbc.conf using a new configuration option, 'newcdrcolumns'. ASTERISK-24976 #close Change-Id: Ibe0c7540a88305c6012786f438a0813ad8b19127 2015-04-26 17:21 +0000 [e9788056e9] Matt Jordan * channels/chan_skinny: Fix compilation error introduced in f8e21a1adf A typo in commit f8e21a1adf resulted in a compilation error in chan_skinny. This patch fixes the typo. ASTERISK-24917 Change-Id: Id7f4ad1fe948eb2408622e80c27936ce4516c33c 2015-04-23 15:11 +0000 [7e5056b393] Kevin Harwell * app_confbridge: Default the template option to a compatible default profile. Confbridge dynamic profiles did not have a default profile unless you explicitly used Set(CONFBRIDGE(bridge,template)=default_bridge). If a template was not set prior to the bridge being created then some options were left with no default values set. This patch makes it so the default templates are set to the default bridge and user profiles. ASTERISK-24749 #close Reported by: philippebolduc Change-Id: I1bd6e94b38701ac2112d842db68de63d46f60e0a 2015-04-24 09:17 +0000 [1da9ec969d] Mark Michelson * res_pjsip_outbound_authenticator: Increase CSeq on authed requests. The way PJSIP generates an authenticated request is to use a previous request as a template. This means that the authenticated request will have the same Call-ID, From header (including tag), and CSeq as the original request. PJSIP generates a new branch on the Via header to indicate that this is a new transaction, though. There are some SIP implementations, though, that do not notice the change in the branch and therefore will match the authed request to the original request's transaction. Since the CSeq is the same, the server will repeat the response it sent to the original request. This patch aids interoperability by increasing the CSeq of the authed request by one. ASTERISK-24845 #close Reported by: Carl Fortin Tested by: Carl Fortin Change-Id: I39c4ca52e688a9f83bcc1878371334becdc5be01 2015-04-20 13:06 +0000 [cb318f3960] Diederik de Groot * Example script for scan-build (the llvm static analyzer) - Added Pre-amble (Options / Flags / Usage Example / GNU License) - Extended Configurability - Made Executable ASTERISK-24917 Change-Id: I70405fe54e4be7dbfbcb62e291690069b88617a8 2015-04-23 12:54 +0000 [eabf3b5a3c] Mark Michelson * res_pjsip_t38: Don't crash on authenticated reinvite after originated T.38 FAX. When Asterisk originates a channel to an application, the channel is hung up once the application finishes executing. When the application in question is SendFax, the Asterisk PJSIP code will attempt to reinvite the T.38 session to audio after the FAX completes. The hangup of the channel happens in the midst of this reinvite transaction. In most circumstances, this works out okay because the BYE is delayed until the reinvite transaction can complete. However, if the reinvite that Asterisk sends receives a 401/407 response, then Asterisk's attempt to re-send the reinvite with authentication will fail. This is because the session supplement in res_pjsip_t38 makes the assumption that the channel on the session will always be non-NULL. Since the channel has been hung up, though, the channel is now NULL. Attempting to operate on the channel causes a crash. This patch fixes the issue by ensuring that the channel on the session is not NULL before attempting to mess with the T.38 framehook. This patch also contains some corrections for comments that were incorrect and really confused me when I first started looking at the code. ASTERISK-25004 #close Reported by Mark Michelson Change-Id: Ic5a1230668369dda4bb13524098aed9306ab45a0 2015-04-23 09:16 +0000 [f70d21b2cf] gtjoseph * res_pjsip: Validate that contact uris start with sip: or sips: Currently we use pjsip_parse_hdr to validate contact uris but it appears that it allows uris without a scheme if there's a port supplied. I.E myexample.com will fail but myexample.com:5060 will pass even though it has no scheme. This causes SEGVs later on whenever the uri is used. To prevent this, permanent_contact_validate has been updated to check that the scheme is either 'sip' or 'sips'. 2 uses of possibly-null endpoint have also been fixed in create_out_of_dialog_request. ASTERISK-24999 Change-Id: Ifc17d16a4923e1045d37fe51e43bbe29fa556ca2 Reported-by: Brad Latus 2015-04-23 08:00 +0000 [1bb16bedc7] Diederik de Groot * Clang: change previous tautological-compare fixes. clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: I0557ae0154a0b7de68883848a609309cdf0aee6a 2015-04-22 16:22 +0000 [1474bb05f6] gtjoseph * res_corosync: Add check for config file before calling corosync apis On some systems, res_corosync isn't compatible with the installed version of corosync so corosync_cfg_initialize fails, load_module returns LOAD_FAILURE, and Asterisk terminates. The work around has been to remember to add res_corosync as a noload in modules.conf. A better solution though is to have res_corosync check for its config file before attempting to call corosync apis and return LOAD_DECLINE if there's no config file. This lets Asterisk loading continue. If you have a res_corosync.conf file and res_corosync fails, you get the same behavior as today and the fatal error tells you something is wrong with the install. ASTERISK-24998 Change-Id: Iaf94a9431a4922ec4ec994003f02135acfdd3889 2015-04-22 15:17 +0000 [73efb093b8] Corey Farrell * Astobj2: Ensure all calls to __adjust_lock pass a valid object. __adjust_lock doesn't check for invalid objects, and doesn't have an appropriate return value for invalid objects. Most callers of __adjust_lock pass objects that have already been confirmed valid, this change adds checks before the remaining calls. ASTERISK-24997 #close Reported by: Corey Farrell Change-Id: I669100f87937cc3f867cec56a27ae9c01292908f 2015-04-22 16:32 +0000 [b0e929219b] gtjoseph * .gitignore: Add .gcno and .gcda Products of --enable-coverage Change-Id: Ie20882d64b60692e2c941ea8872ab82a86ce77a3 2015-04-22 04:17 +0000 [d6dfc85666] Diederik de Groot * Clang: Fix some more tautological-compare warnings. clang can warn about a so called tautological-compare, when it finds comparisons which are logically always true, and are therefor deemed unnecessary. Exanple: unsigned int x = 4; if (x > 0) // x is always going to be bigger than 0 Enum Case: Each enumeration is its own type. Enums are an integer type but they do not have to be *signed*. C leaves it up to the compiler as an implementation option what to consider the integer type of a particu- lar enumeration is. Gcc treats an enum without negative values as an int while clang treats this enum as an unsigned int. rmudgett & mmichelson: cast the enum to (unsigned int) in assert. The cast does have an effect. For gcc, which seems to treat all enums as int, the cast to unsigned int will eliminate the possibility of negative values being allowed. For clang, which seems to treat enums without any negative members as unsigned int, the cast will have no effect. If for some reason in the future a negative value is ever added to the enum the assert will still catch the negative value. ASTERISK-24917 Change-Id: Ief23ef68916192b9b72dabe702b543ecfeca0b62 2015-04-14 14:04 +0000 [7b57116833] Joshua Colp * res_pjsip_mwi: Send unsolicited MWI NOTIFY on startup and when endpoint registers. Currently the res_pjsip_mwi module only sends an unsolicited MWI NOTIFY upon a mailbox state change (such as a new message being left, or one being deleted). In practice this is not sufficient to keep clients aware of the current MWI status. This change makes the module send unsolicited MWI NOTIFY on startup so that clients are guaranteed to have the most up to date MWI information. It also makes clients receive an unsolicited MWI NOTIFY upon registration so if they are unaware of the current MWI status they receive it. ASTERISK-24982 #close Reported by: Joshua Colp Change-Id: I043f20230227e91218f18a82c7d5bb2aa62b1d58 2015-04-21 15:17 +0000 [ad1a118632] Corey Farrell * Check for ao2_alloc failure in __ast_channel_internal_alloc. Fix a crash that could occur in __ast_channel_internal_alloc if ao2_alloc fails. ASTERISK-24991 #close Change-Id: I4ca89189eb22f907408cb87d0a1645cfe1314a90 2015-04-20 14:30 +0000 [3327560cb2] Mark Michelson * res_pjsip_pubsub: Set the endpoint on SUBSCRIBE dialogs. When SUBSCRIBE dialogs were established, we never associated the endpoint that created the subscription with the dialog we end up creating. In most cases, this ended up not causing any problems. The actual bug that was observed was that when a device that was behind NAT established a subscription with Asterisk, Asterisk would end up sending in-dialog NOTIFY requests to the device's private IP addres instead of the public address of the NAT router. When Asterisk receives the initial SUBSCRIBE from the device, res_pjsip_nat rewrites the contact to the public address on which the SUBSCRIBE was received. This allows for the dialog to have its target address set to the proper public address. Asterisk then would send a 200 OK response to the SUBSCRIBE, then a NOTIFY with the initial subscription state. The device would then send a 200 OK response to Asterisk's NOTIFY. Here's where things went wrong. When the 200 OK arrived, res_pjsip_nat did not rewrite the address in the Contact header. Then, when the PJSIP dialog layer processed the 200 OK, PJSIP would perform a comparison between the IP address in the Contact header and its saved target address for the dialog. Since they differed, PJSIP would update the target dialog address to be the address in the Contact header. From this point, if Asterisk needed to send a NOTIFY to the device, the result was that the NOTIFY would be sent to the private address that the device placed in the Contact header. The reason why res_pjsip_nat did not rewrite the address when it received the 200 OK response was that it could not associate the incoming response with a configured endpoint. This is because on a response, the only way to associate the response to an endpoint is by finding the dialog that the response is associated with and then finding the endpoint that is associated with that dialog. We do not perform endpoint lookups on responses. res_pjsip_pubsub skipped the step of associating the endpoint with the dialog we created, so res_pjsip_nat could not find the associated endpoint and therefore couldn't rewrite the contact. This commit message is like 50x longer than the actual fix. ASTERISK 24981 #close Reported by Mark Michelson Change-Id: I2b963c58c063bae293e038406f7d044a8a5377cd 2015-04-20 18:00 +0000 [d08446ec36] Richard Mudgett * chan_dahdi/sig_pri: Make post AMI HangupRequest events on PRI channels. The chan_dahdi channel driver is a very old driver. The ability for it to support ISDN was added well after the initial analog support. Setting the softhangup flags is a carry over from the original analog code. The driver was not updated to call ast_queue_hangup() which will post the AMI HangupRequest event. * Changed sig_pri.c to call ast_queue_hangup() instead of setting the softhangup flag when the remote party initiates a hangup. ASTERISK-24895 #close Reported by: Andrew Zherdin Change-Id: I5fe2e48556507785fd8ab8e1c960683fd5d20325 2015-04-20 13:01 +0000 [2be9cc2643] Diederik de Groot * Fix/Update clang-RAII macro implementation - When you need to refer to 'variable XXX' outside a block, it needs to be declared as '__block XXX', otherwise it will not be available with- in the block, making updating that variable hard to do, and ast_free lead to issues. - Removed the #error message because it creates complications when compiling external projects against asterisk For example when using a different compiler than the one used to compile asterisk. The warning/error should be generated during the configure process not the compilation process ASTERISK-24917 Change-Id: I12091228090e90831bf2b498293858f46ea7a8c2 2015-04-20 09:53 +0000 [b74b2cdcda] gtjoseph * pjsip_options: Fix format specifier for int64_t rtt. Contact status rtt is an int64_t and needs the PRId64 macro to properly create the format specifier on 32-bit systems. Change-Id: I4b8ab958fc1e9a179556a9b4ffa49673ba9fdec7 2015-04-18 13:36 +0000 [63169e00ff] gtjoseph * pjsip_options: Fix non-qualified contacts showing as unavailable The "Add qualify_timeout processing and eventing" patch introduced an issue where contacts that had qualify_frequency set to 0 were showing Unavailable instead Unknown. This patch checks for qualify_frequency=0 and create an "Unknown" contact_status with an RTT = 0. Previously, the lack of contact_status implied Unknown but since we're now changing endpoint state based on contact_status, I've had to add new UNKNOWN status so that changes could trigger the appropriate contact_status observers. ASTERISK-24977: #close Change-Id: Ifcbc01533ce57f0e4e584b89a395326e098b8fe7 2015-04-19 15:49 +0000 [f0c82a173a] Matt Jordan * main/pbx: Don't attempt to destroy a previously destroyed exten/priority tuple When a PBX registrar is unloaded, it will fail to remove its extension from the context root_table if a dialplan application used by that extension is still loaded. This can be the case for AGI, which can be unloaded after several of the standard PBX providers. Often, this is harmless; however, if the extension's priorities are removed during the failed unloading *and* the dialplan application later unregisters, it leaves a ticking timebomb for the next PBX provider that attempts to iterate over the extensions. When that occurs, the peer_table pointer on the extension will already be set to NULL. The current code does not check to see if the pointer is NULL before passing it to a hashtab function this is not NULL tolerant. Since it is possible for the peer_table to be NULL when we normally would not expect that to be the case, the solution in this patch is to simply skip over processing an extension's priorities if peer_table is NULL. Prior to this patch, the tests/pbx/callerid_match test would crash during module unload. With this patch, the test no longer crashes after running. ASTERISK-24774 #close Reported by: Corey Farrell Change-Id: I2bbeecb7e0f77bac303a1b9135e4cdb4db6d4c40 2015-04-17 18:05 +0000 [82bc0fd3ad] Richard Mudgett * res_fax: Fix latent bug exposed by ASTERISK-24841 changes. Three fax related tests started failing as a result of changes made for ASTERISK-24841: tests/fax/pjsip/gateway_t38_g711 tests/fax/sip/gateway_mix1 tests/fax/sip/gateway_mix3 Historically, ast_channel_make_compatible() did nothing if the channels were already "compatible" even if they had a sub-optimal translation path already setup. With the changes from ASTERISK-24841 this is no longer true in order to allow the best translation paths to always be picked. In res_fax.c:fax_gateway_framehook() code manually setup the channels to go through slin and then called ast_channel_make_compatible(). With the previous version of ast_channel_make_compatible() this was always a no-operation. * Remove call to ast_channel_make_compatible() in fax_gateway_framehook() that now undoes what was just setup when the framehook is attached. * Fixed locking around saving the channel formats in fax_gateway_framehook() to ensure that the formats that are saved are consistent. * Fix copy pasta errors in fax_gateway_framehook() that confuses read and write when dealing with saved channel formats. ASTERISK-24841 Reported by: Matt Jordan Change-Id: I6fda0877104a370af586a5e8cf9e161a484da78d 2015-04-17 16:19 +0000 [c59a800707] Corey Farrell * Fix issue with AST_THREADSTORAGE_RAW when DEBUG_THREADLOCALS is enabled. When DEBUG_THREADLOCALS is enabled it causes the threadlocal cleanup to be called as a function. This causes a compile error with raw threadstorage as it uses NULL for cleanup. This fix uses a macro that provides NULL when DEBUG_THREADLOCALS is disabled, and replaces the call to "c_cleanup(data);" with "{};" when DEBUG_THREADLOCALS is enabled. ASTERISK-24975 #close Reported by: Ashley Sanders Change-Id: I3ef7428ee402816d9fcefa1b3b95830c00d5c402 2015-04-15 10:38 +0000 [4f1a8dbe92] Mark Michelson * Detect potential forwarding loops based on count. A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23 2015-04-11 16:56 +0000 [674b18bdf0] gtjoseph * pjsip_options: Add qualify_timeout processing and eventing This is the second follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html The basic issues are that changes in contact status don't cause events to be emitted for the associated endpoint. Only dynamic contact add/delete actions update the endpoint. Also, the qualify timeout is fixed by pjsip at 32 seconds which is a long time. This patch makes use of the new transaction timeout feature in r4585 and provides the following capabilities... 1. A new aor/contact variable 'qualify_timeout' has been added that allows the user to specify the maximum time in milliseconds to wait for a response to an OPTIONS message. The default is 3000ms. When the timer expires, the contact is marked unavailable. 2. Contact status changes are now propagated up to the endpoint as follows... When any contact is 'Available', the endpoint is marked as 'Reachable'. When all contacts are 'Unavailable', the endpoint is marked as 'Unreachable'. The existing endpoint events are generated appropriately. ASTERISK-24863 #close Change-Id: Id0ce0528e58014da1324856ea537e7765466044a Tested-by: Dmitriy Serov Tested-by: George Joseph 2015-04-16 10:51 +0000 [b56c1914fa] Kevin Harwell * bridge.c: NULL app causes crash during attended transfer Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: Change-Id: I5134b84c4524906d8148817719d76ffb306488ac 2015-04-16 13:20 +0000 [8d4ce7cc2b] Scott Griepentrog * res_pjsip_pubsub: On notify fail deleted sub_tree is then referenced This change makes the send_notify of the sub_tree not happen when the sub_tree has been deleted due to the notify call failing, which avoids a crash. ASTERISK-24970 #close Change-Id: I1f20ffc08b192f59c457293b218025a693992cbf 2015-04-11 16:39 +0000 [bf46799f0e] gtjoseph * res_pjsip: Refactor endpt_send_request to include transaction timeout This is the first follow-on to https://reviewboard.asterisk.org/r/4572/ and the discussion at http://lists.digium.com/pipermail/asterisk-dev/2015-March/073921.html Since we currently have no control over pjproject transaction timeout, this patch pulls the pjsip_endpt_send_request function out of pjproject and into res_pjsip/endpt_send_transaction in order to implement that capability. Now when the transaction is initiated, we also schedule our own pj_timer with our own desired timeout. If the transaction completes before either timeout, pjproject cancels its timer, and calls our tsx callback where we cancel our timer and run the app callback. If the pjproject timer times out first, pjproject calls our tsx callback where we cancel our timer and run the app callback. If our timer times out first, we terminate the transaction which causes pjproject to cancel its timer and call our tsx callback where we run the app callback. Regardless of the scenario, pjproject is calling the tsx callback inside the group_lock and there are checks in the callback to make sure it doesn't run twice. As part of this patch ast_sip_send_out_of_dialog_request was created to replace its similarly named private function. It takes a new timeout argument in milliseconds (<= 0 to disable the timeout). ASTERISK-24863 #close Reported-by: George Joseph Tested-by: George Joseph Change-Id: I0778dc730d9689c5147a444a04aee3c1026bf747 2015-04-11 17:04 +0000 [1b6f6ff841] gtjoseph * res_pjsip: Add global option to limit the maximum time for initial qualifies Currently when Asterisk starts initial qualifies of contacts are spread out randomly between 0 and qualify_timeout to prevent network and system overload. If a contact's qualify_frequency is 5 minutes however, that contact may be unavailable to accept calls for the entire 5 minutes after startup. So while staggering the initial qualifies is a good idea, basing the time on qualify_timeout could leave contacts unavailable for too long. This patch adds a new global parameter "max_initial_qualify_time" that sets the maximum time for the initial qualifies. This way you could make sure that all your contacts are initialy, randomly qualified within say 30 seconds but still have the contact's ongoing qualifies at a 5 minute interval. If max_initial_qualify_time is > 0, the formula is initial_interval = min(max_initial_interval, qualify_timeout * random(). If not set, qualify_timeout is used. The default is "0" (disabled). ASTERISK-24863 #close Change-Id: Ib80498aa1ea9923277bef51d6a9015c9c79740f4 Tested-by: George Joseph 2015-04-15 16:08 +0000 [5d218cde87] gtjoseph * More .gitignore updates Added .pyc and .sha1 to the top-level .gitignore. Change-Id: I7dfc4f554d54d22947b38140d3305007503cc16a Tested-by: George Joseph 2015-04-14 13:16 +0000 [abd56db3e0] Rodrigo Ramírez Norambuena * cel_pgsql: Fix name string for log on unable allocate memory. The LOG_ERROR has reference to CDR instead of CEL for LENGTHEN_BUF1 and LENGTHEN_BUF2. ASTERISK-24965 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: Icc818697d7d66d34bfe3048cdd15ca2b06c89744 2015-04-14 13:48 +0000 [222fbe1d9a] Corey Farrell * Build System: Replace comment about setting menuselect defaults. The Makefile claims that you can set default menuselect options by creating ~/.asterisk.makeopts or /etc/asterisk.makeopts, but those files have never been respected in Asterisk 11 or 13. This changes the comment to accurately reflect that these files are not automatically used by the build system. ASTERISK-13721 #close Reported by: pj Change-Id: Ibde804ff196283def49ccb9432fbf224a22586e2 2015-04-12 09:08 +0000 [07e729cc7b] Rodrigo Ramírez Norambuena * cdr_pgsql: Fix CLI "cdr show pgsql status" command. The command always showed the usage information. * Fix the error in command validation for CLI_SHOWUSAGE. ASTERISK-24959 #close Reported by: Rodrigo Ramirez Norambuena Change-Id: I584f0936bb01001336a468a55c1d05d79fe795d5 (cherry picked from commit 23a180cade51e84b9def65b05759c3cb9feba225) 2015-04-13 19:06 +0000 [7d43d85bea] gtjoseph * .gitignore updates for master/13 Added products of ./bootstrap Added nmenuselect and gmenuselect to menuselect/ Change-Id: Ied658463958bafc04a9aff9ebc28e40c116a6e35 2015-04-13 14:41 +0000 [3d27c223a5] David M. Lee * Fixing extconf compile During the mass code deletion for clang support, a stray backslash was left behind that was causing utils to fail to compile. Change-Id: I60e5fa58c9a5b248bde23aaada79ff663f87a2a1 2015-04-13 09:54 +0000 [e996d8f728] Matt Jordan * build_tools/make_version: Update version parsing for Git migration External systems - such as the Asterisk Test Suite - require knowledge of the upstream branch. Unfortunately, after moving to Git, the Asterisk version currently consists of only a 'GIT" prefix followed by an object blob, e.g., GIT-as08d7. This makes it difficult for such systems to know what features are available in a particular check out of Asterisk. This patch fixes this by hardcoding the branch in a variable in the make_version script. Since the mainline branches are not changed often - typically only once a year - this is a reasonable approach to solving the problem, and is more reliable than parsing the output of 'git branch -vv'. Branches that track off of an upstream primary branch will then get the benefit of knowing which mainline branch they are currently based off of. ASTERISK-24954 #close Change-Id: I8090d5d548b6d19e917157ed530b914b7eaf9799 2015-04-12 12:59 +0000 [d1a6f1a9f9] Matt Jordan * git migration: Remove support for file versions Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Alter the "core show file version" CLI command such that it always reports the version of Asterisk. The file version is no longer available. * main/manager: The Version key now always reports the Asterisk version. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action. - Modification of the "core show file version" CLI command. Change-Id: Ia932d3c64cd18a14a3c894109baa657ec0a85d28 2015-04-13 06:19 +0000 [0e4b997cd7] Corey Farrell * res_monitor: Add dependency on func_periodic_hook. OPTIONAL_API has conditionals to define AST_OPTIONAL_API and AST_OPTIONAL_API_ATTR differently based on if AST_API_MODULE is defined. Unfortunately this is inside the include protection block, so only the first status of AST_API_MODULE is respected. For example res_monitor is an optional API provider, but uses func_periodic_hook. This makes func_periodic_hook non-optional to res_monitor. ASTERISK-17608 #close Reported by: Warren Selby Change-Id: I8fcf2a5e7b481893e17484ecde4f172c9ffb5679 2015-04-12 06:12 +0000 [a77c31b99c] Corey Farrell * main/editline: Add .gitignore. This patch adds a .gitignore for main/editline to ignore all build results. Change-Id: I68c7bf375ea46282689e5a706534b69fca233b5d 2015-04-11 23:22 +0000 [d918c3b78e] Matt Jordan * .gitignore: Ignore tarballs (*.gz) This patch updates the root .gitignore file to ignore files with a .gz extension. This will cause git to ignore downloaded sound tarballs in the the sounds/ directory. Change-Id: I1e42fbfa02a8884231507b683e8e49ac3e278aaa 2015-04-11 13:20 +0000 [555b5f5d30] gtjoseph * Add .gitignore and .gitreview files Add the .gitignore and .gitreview files to the asterisk repo. NB: You can add local ignores to the .git/info/exclude file without having to do a commit. Common ignore patterns are in the top-level .gitignore file. Subdirectory-specific ignore patterns are in their own .gitignore files. Change-Id: I4c8af3b8e3739957db545f7368ac53f38e99f696 Tested-by: George Joseph 2015-04-11 10:35 +0000 [5807ca519c] Matt Jordan * Blocked revisions 434708 ........ main/event: Remove unnecessary assignment of negative value to enum When cleaning up some clang compiler warnings, the comparison of a negative value to an unsigned enum was removed. However, the initial assignment of a negative value to said enum remained in the variable declaration. This patch removes that assignment. Thanks to ibercom in #asterisk-bugs for pointing it out. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434709 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-11 10:26 +0000 [d0d78d5732] Diederik de Groot (License 6600) * clang compiler warnings: Fix various warnings for tests This patch fixes a variety of clang compiler warnings for unit tests. This includes autological comparison issues, ignored return values, and interestingly enough, one embedded function. Fun! Review: https://reviewboard.asterisk.org/r/4555 ASTERISK-24917 Reported by: dkdegroot patches: rb4555.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434705 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434706 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-11 10:10 +0000 [4cf7d0bf01] Juergen Spies (License 6698) * res/res_pjsip_t38: Add missing initialization of t38faxmaxdatagram Prior to this patch, the far_max_datagram value on the UDPTL structure would remain -1 if the remote endpoint fails to provide the SDP media attribute T38FaxMaxDatagram. This can result in the INVITE request being rejected. With this patch, we will now properly initialize the value with either the default value or with the value provided by pjsip.conf's t38_udptl_maxdatagram parameter. Review: https://reviewboard.asterisk.org/r/4589 ASTERISK-24928 #close Reported by: Juergen Spies Tested by: Juergen Spies patches: pjsipT38patch20150331.txt submitted by Juergen Spies (License 6698) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 18:29 +0000 [13cd99682d] Richard Mudgett * chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. With this patch, chan_pjsip/res_pjsip now sets the native formats to the codecs negotiated by a call. * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native formats to include all the negotiated audio codecs instead of only the initial preferred audio codec and later the currently received audio codec. * The audio frame handling in channel.c:ast_read() is more streamlined and will automatically adjust to changes in received frame formats. The new policy is to remove translation and pass the new frame format to the receiver except if the translation was to a signed linear format. A more long winded version is commented in ast_read() along with some caveats. * The audio frame handling in channel.c:ast_write() is more streamlined and will automatically adjust any needed translation to changes in the frame formats sent. Frame formats sent can change for many reasons such as a recording is being played back or the bridged peer changed the format it sends. Since it is a normal expectation that sent formats can change, the codec mismatch warning message is demoted to a debug message. * Removed the short circuit check in channel.c:ast_channel_make_compatible_helper(). Two party bridges need to make channels compatible with each other. However, transfers and moving channels among bridges can result in otherwise compatible channels having sub-optimal translation paths if the make compatible check is short circuited. A result of forcing the reevaluation of channel compatibility is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc options take effect consistently now. It is unfortunate that these two options are enabled by default and negate some of the benefits to the changes in channel.c:ast_read() by forcing translation through signed linear on a two party bridge. * Improved the softmix bridge technology to better control the translation of frames to the bridge. All of the incoming translation is now normally handled by ast_read() instead of splitting any translation steps between ast_read() and the slin factory. If any frame comes in with an unexpected format then the translation path in ast_read() is updated for the next frame and the slin factory handles the current frame translation. This is the final patch in a series of patches aimed at improving translation path choices. The other patches are on the following reviews: https://reviewboard.asterisk.org/r/4600/ https://reviewboard.asterisk.org/r/4605/ ASTERISK-24841 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4609/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434671 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 16:03 +0000 [af458e2e60] Kevin Harwell * chan_sip: make progressinband default to no After the "progressinband" value setting of "never" was updated to never send a 183 this separated its use from the "no" value. Since "never" was the default, but most users probably expect "no" this patch updates the default for the "progressinband" setting to "no." ASTERISK-24835 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4606/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434654 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 12:53 +0000 [88b0fa7755] yaron nahum (License 6676) * res_pjsip: Add an 'auto' option for DTMF Mode This patch adds support for automatically detecting the type of DTMF that a PJSIP endpoint supports. When the 'dtmf_mode' endpoint option is set to 'auto', the channel created for an endpoint will attempt to determine if RFC 4733 DTMF is supported. If so, it will use that DTMF type. If not, the DTMF type for the channel will be set to inband. Review: https://reviewboard.asterisk.org/r/4438 ASTERISK-24706 #close Reported by: yaron nahum patches: yaron_patch_3_Feb.diff submitted by yaron nahum (License 6676) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434637 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 11:59 +0000 [16afee4651] gtjoseph * res_pjsip_config_wizard: Cleanup load unload While investigating other unload issues I realized that the load/unload process for the config wizard was pretty ugly so I've refactored it as follows... When the res_pjsip sorcery instance is created the config_wizard bumps it's own module reference to prevent it from unloading while the sorcery instance is still active. When res_pjsip unloads and it's sorcery instance is destroyed, the config wizard unrefs itself which then allows itself to unload cleanly. Since the config wizard now can't load after res_pjsip or unload before it (which should have been the correct behavior all along), I was able to remove the chunks of code in both load_module and unload_module that handled that case. Ran the testsuite tests to insure there were no functional changes and REF_DEBUG to insure that Asterisk was shutting down cleanly with no FRACKs or leaks. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4610/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 11:37 +0000 [125acc52fe] Richard Mudgett * bridge_softmix.c,channel.c: Minor code simplification and cleanup. * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats() and made some debug messages more helpful. * Made some debug and warning messages more helpful in channel.c:set_format(). Review: https://reviewboard.asterisk.org/r/4607/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434617 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 11:28 +0000 [a63f7ad04a] Richard Mudgett * translate.c: Only select audio codecs to determine the best translation choice. Given a source capability of h264 and ulaw, a destination capability of h264 and g722 then ast_translator_best_choice() would pick h264 as the best choice even though h264 is a video codec and Asterisk only supports translation of audio codecs. When the audio starts flowing, there are warnings about a codec mismatch when the channel tries to write a frame to the peer. * Made ast_translator_best_choice() only select audio codecs. * Restore a check in channel.c:set_format() lost after v1.8 to prevent trying to set a non-audio codec. This is an intermediate patch for a series of patches aimed at improving translation path choices for ASTERISK-24841. This patch is a complete enough fix for ASTERISK-21777 as the v11 version of ast_translator_best_choice() does the same thing. However, chan_sip.c still somehow tries to call ast_codec_choose() which then calls ast_best_codec() with a capability set that doesn't contain any audio formats for the incoming call. The remaining warning message seems to be a benign transient. ASTERISK-21777 #close Reported by: Nick Ruggles ASTERISK-24380 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4605/ ........ Merged revisions 434614 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434615 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 09:55 +0000 [c9791dba1f] Matt Jordan * res/ari: Fix model validation for ChannelHold event When the ChannelHold event was added, the 'musicclass' parameter was erroneously removed. This caused the ChannelHold events to be rejected as they failed model validation. This patch updates the Swagger schema such that it now properly reflects the event that is being created. Hooray for tests that catch things like this. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-10 07:39 +0000 [c39faa4729] Y Ateya (License 6693) * channels/chan_iax2: Improve POKE expiration time calculation for lossy networks POKE is used to check for peer availability; however, in networks with packet loss, the current calculations may result in POKE expiration times that are too short. This patch alters the expiration/retry time logic to take into account the last known qualify round trip time, as opposed to always using a static value for each peer. Review: https://reviewboard.asterisk.org/r/4536 ASTERISK-22352 #close Reported by: Frederic Van Espen ASTERISK-24894 #close Reported by: Y Ateya patches: poke_noanswer_duration.diff submitted by Y Ateya (License 6693) ........ Merged revisions 434564 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 17:35 +0000 [75c2c85962] gtjoseph * res_pjsip_phoneprov_provider: Fix reference leak on unload res_pjsip_phoneprov_provider was leaking references to phoneprov objects due to a missing OBJ_NODATA in an ao2_callback in load_users(). Rather than adding the OBJ_NODATA, I changed load_users to use a more straightforward ao2_iterator. This plugged the leak but exposed an unload order issue between res_pjsip_phoneprov_provider, res_phoneprov and res_pjsip. res_pjsip_phoneprov_provider unloads first, then res_phoneprov, then res_pjsip. Since res_pjsip_phoneprov_provider uses res_pjsip's sorcery instance, when it unloads, it's objects are still in the sorcery instance. When res_pjsip unloads, it destroys all its objects including res_pjsip_phoneprov_provider's. The phoneprov destructor then attempts to unregister the extension from res_phoneprov but because res_phoneprov is already cleaned up, its users container is gone and we get a FRACK. Simple solution, check for the NULL users container before attempting to remove the entry. Duh. Ran tests/res_phoneprov/res_phoneprov_provider. No leaks in res_pjsip_phoneprov_provider and no FRACKs. Reported-by: Corey Farrell Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4608/ ASTERISK-24935 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434545 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 17:31 +0000 [73c286a393] gtjoseph * loader/main: Don't set ast_fully_booted until deferred reloads are processed Until we have a true module management facility it's sometimes necessary for one module to force a reload on another before its own load is complete. If Asterisk isn't fully booted yet, these reloads are deferred. The problem is that asterisk reports fully booted before processing the deferred reloads which means Asterisk really isn't quite ready when it says it is. This patch moves the report of fully booted after the processing of the deferred reloads is complete. Since the pjsip stack has the most number of related modules, I ran the channels/pjsip testsuite to make sure there aren't any issues. All tests passed. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4604/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434544 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 17:03 +0000 [5737650a67] Kevin Harwell * res_pjsip: add CLI command to show global and system configuration Added a new CLI command for res_pjsip that shows both global and system configuration settings: pjsip show settings ASTERISK-24918 #close Reported by: Scott Griepentrog Review: https://reviewboard.asterisk.org/r/4597/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434527 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 11:07 +0000 [1695a5b85f] Richard Mudgett * chan_iax2.c: Fix ref leak in iax2_request(). * Increased warning message format capability string buffer size in iax2_request(). Review: https://reviewboard.asterisk.org/r/4601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434510 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 10:54 +0000 [92c1688edb] Richard Mudgett * bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible(). Review: https://reviewboard.asterisk.org/r/4601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 10:42 +0000 [2679d0100a] yaron nahum (License 6676) * res/res_pjsip_dlg_options: Add a module to handle in-dialog OPTIONS requests This patch adds a new session supplement that handles in-dialog OPTIONS requests. Said OPTIONS requests are sent a 200 OK, as an endpoint lookup for the OPTIONS request would already have been done by the time the session supplement receives the inbound request. ASTERISK-24862 #close Reported by: yaron nahum patches: res_pjsip_dlg_options.c submitted by yaron nahum (License 6676) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434506 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-09 07:56 +0000 [6ba6e3dffd] Diederik de Groot (License 6600) * clang compiler warnings: Fix autological comparisons This fixes autological comparison warnings in the following: * chan_skinny: letohl may return a signed or unsigned value, depending on the macro chosen * func_curl: Provide a specific cast to CURLoption to prevent mismatch * cel: Fix enum comparisons where the enum can never be negative * enum: Fix comparison of return result of dn_expand, which returns a signed int value * event: Fix enum comparisons where the enum can never be negative * indications: tone_data.freq1 and freq2 are unsigned, and hence can never be negative * presencestate: Use the actual enum value for INVALID state * security_events: Fix enum comparisons where the enum can never be negative * udptl: Don't bother to check if the return value from encode_length is less than 0, as it returns an unsigned int * translate: Since the parameters are unsigned int, don't bother checking to see if they are negative. The cast to unsigned int would already blow past the matrix bounds. * res_pjsip_exten_state: Use a temporary value to cache the return of ast_hint_presence_state * res_stasis_playback: Fix enum comparisons where the enum can never be negative * res_stasis_recording: Add an enum value for the case where the recording operation is in error; fix enum comparisons * resource_bridges: Use enum value as opposed to -1 * resource_channels: Use enum value as opposed to -1 Review: https://reviewboard.asterisk.org/r/4533 ASTERISK-24917 Reported by: dkdegroot patches: rb4533.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434469 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 21:05 +0000 [e05c8ae68e] Stefan Engström (License 6691) * apps/app_queue: Prevent possible crash when evaluating queue penalty rules Although it only occurred once, a crash occurred when a queue attempted to evaluate a queue penalty rule that appeared to have already been destroyed. In many locations in app_queue, a test is done to see if qe->pr is NULL; however, when we dispose of a queue's penalty rules, we don't set the pointer to NULL after free'ing it. This patch does that to prevent any dangling pointers from lingering on the queue object. Review: https://reviewboard.asterisk.org/r/4522 ASTERISK-23319 #close Reported by: Vadim patches: rb4552.patch submitted by Stefan Engström (License 6691) ........ Merged revisions 434448 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434449 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 13:15 +0000 [f21b45db49] Jonathan Rose * res_pjsip_t38: Fix FAX failures when using PJSIP with authentication Without this patch, if a PJSIP endpoint with udptl enabled and authentication set attempted to use sendFax, the FAX session would fail during setup. This was because the invite issued in response to being auth challenged would cause the PJSIP channel performing the FAX to receive a second T38 framehook and this would cause frames to be consumed in an inappropriate manner. ASTERISK-24933 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4577/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434425 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 13:14 +0000 [4441bb6a25] Richard Mudgett * Bridging: Eliminate the unnecessary make channel compatible with bridge operation. When a channel enters the bridging system it is first made compatible with the bridge and then the bridge technology makes the channel compatible with the technology. For all but the DAHDI native and softmix bridge technologies the make channel compatible with the bridge step is an effective noop because the other technologies allow all audio formats. For the DAHDI native bridge technology it doesn't matter because it is not an initial bridge technology and chan_dahdi allows only one native format per channel. For the softmix bridge technology, it is a noop at best and harmful at worst because the wrong translation path could be setup if the channel's native formats allow more than one audio format. This is an intermediate patch for a series of patches aimed at improving translation path choices. * Removed code dealing with the unnecessary step of making the channel compatible with the bridge. ASTERISK-24841 Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 11:40 +0000 [f767440906] Maciej Szmigiero (license 6085) * Security/tcptls: MitM Attack potential from certificate with NULL byte in CN. When registering to a SIP server with TLS, Asterisk will accept CA signed certificates with a common name that was signed for a domain other than the one requested if it contains a null character in the common name portion of the cert. This patch fixes that by checking that the common name length matches the the length of the content we actually read from the common name segment. Some certificate authorities automatically sign CA requests when the requesting CN isn't already taken, so an attacker could potentially register a CN with something like www.google.com\x00www.secretlyevil.net and have their certificate signed and Asterisk would accept that certificate as though it had been for www.google.com - this is a security fix and is noted in AST-2015-003. ASTERISK-24847 #close Reported by: Maciej Szmigiero Patches: asterisk-null-in-cn.patch submitted by mhej (license 6085) ........ Merged revisions 434337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 434338 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434384 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 11:23 +0000 [1712d16825] Richard Mudgett * format_cache.c: Add missing slin12 format to ast_format_cache_is_slinear(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434357 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 07:33 +0000 [ae39dd1f46] Matt Jordan * chan_iax2: Fix compilation issue due to funky merge Don't mix declarations and code git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 07:00 +0000 [05397ad01e] Jaco Kroon (License 5671) * chan_iax2: Fix crash caused by unprotected access to iaxs[peer->callno] This patch fixes an access to the peer callnumber that is unprotected by a corresponding mutex. The peer->callno value can be changed by multiple threads, and all data inside the iaxs array must be procted by a corresponding lock of iaxsl. The patch moves the unprotected access to a location where the mutex is safely obtained. Review: https://reviewboard.asterisk.org/r/4599/ ASTERISK-21211 #close Reported by: Jaco Kroon patches: asterisk-11.2.1-iax2_poke-segfault.diff submitted by Jaco Kroon (License 5671) ........ Merged revisions 434291 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434292 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 06:53 +0000 [be13c72142] Valentin Vidić (License 6697) * chan_sip: Handle IPv4 mapped IPv6 clients when NAT is enabled When udpbindaddr is set to the IPv6 bind all address of '::', Asterisk will attempt to handle both IPv4 and IPv6 addresses, although the information will be stored in a struct with an AF_INET6 address type. However, the current NAT handling code won't handle the IPv4 mapped IPv6 addresses correctly. This patch adds an additional check for the mapped address case, allowing the NAT code to handle clients even when the address is IPv6. Review: https://reviewboard.asterisk.org/r/4563/ ASTERISK-18032 #close Reported by: Christoph Timm patches: nat_with_ipv6.diff submitted by Valentin Vidić (License 6697) ........ Merged revisions 434288 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434289 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-08 06:44 +0000 [f324870dab] Diederik de Groot (License 6600) * clang compiler warnings: Fix pointer-bool-converesion warnings This patch fixes several warnings pointed out by the clang compiler. * chan_pjsip: Removed check for data->text, as it will always be non-NULL. * app_minivm: Fixed evaluation of etemplate->locale, which will always evaluate to 'true'. This patch changes the evaluation to use ast_strlen_zero. * app_queue: - Fixed evaluation of qe->parent->monfmt, which always evaluates to true. Instead, we just check to see if the dereferenced pointer evaluates to true. - Fixed evaluation of mem->state_interface, wrapping it with a call to ast_strlen_zero. * res_smdi: Wrapped search_msg->mesg_desk_term with calls to ast_strlen_zero. Review: https://reviewboard.asterisk.org/r/4541 ASTERISK-24917 Reported by: dkdegroot patches: rb4541.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434285 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434286 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-07 14:38 +0000 [a6aed7f6f6] Scott Griepentrog * Revert accidental change in r434261 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434262 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-07 14:35 +0000 [0584e29300] Scott Griepentrog * pjsip: resolve compatibility problem with ast_sip_session A change in r430179 inserted a variable near the top of a structure caused a problem when running DPMA in a version of Asterisk compiled across the change. This patch moves the new variable to the end of the structure, eliminating the problem. Review: https://reviewboard.asterisk.org/r/4574/ ........ Merged revisions 433944 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434261 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-07 11:40 +0000 [d754f70239] Kevin Harwell * bridge.c: Hangup attended transfer target after it has been swapped out After completing an attended transfer the transfer target channel (the one that gets swapped out) was not being hung up after leaving the bridge. This resulted in a channel possibly being left around. Added an explicit softhangup for the channel in question after the transfer is successfully completed in order to make sure the channel is hung up. ASTERISK-24782 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4575/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434240 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-07 10:33 +0000 [c516981dc7] Mark Michelson * Do not queue message requests that we do not respond to. If we receive a MESSAGE request that we cannot send a response to, we should not send the incoming MESSAGE to the dialplan. This commit should help the bouncing message_retrans test to pass consistently. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434218 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-07 10:21 +0000 [ab803ec342] Matt Jordan * ARI: Add the ability to intercept hold and raise an event For some applications - such as SLA - a phone pressing hold should not behave in the fashion that the Asterisk core would like it to. Instead, the hold action has some application specific behaviour associated with it - such as disconnecting the channel that initiated the hold; only playing MoH to channels in the bridge if the channels are of a particular type, etc. One way of accomplishing this is to use a framehook to intercept the hold/unhold frames, raise an event, and eat the frame. Tasty. This patch accomplishes that using a new dialplan function, HOLD_INTERCEPT. In addition, some general cleanup of raising hold/unhold Stasis messages was done, including removing some RAII_VAR usage. Review: https://reviewboard.asterisk.org/r/4549/ ASTERISK-24922 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 21:09 +0000 [488f093e97] Diederik de Groot (License 6600) * clang compiler warnings: Fix sometimes-initialized warning in func_math This patch fixes a bug in a unit test in func_math where a variable could be passed to ast_free that wasn't allocated. This patch corrects the issue and ensures that we only attempt to free a variable if we previously allocated it. Review: https://reviewboard.asterisk.org/r/4552 ASTERISK-24917 Reported by: dkdegroot patches: rb4552.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434190 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434191 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 21:03 +0000 [c027133f6d] Diederik de Groot (License 6600) * clang compiler warnings: Fix non-literal-null-conversion warnings Clang will flag errors when a char pointer is set to '\0', as opposed to a value that the char pointer points to. This patch fixes this warning in a variety of locations. Review: https://reviewboard.asterisk.org/r/4551 ASTERISK-24917 Reported by: dkdegroot patches: rb4551.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434187 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434188 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 14:23 +0000 [2270c40d33] Kevin Harwell * res_pjsip: config option 'timers' can't be set to 'no' When setting the configuration option 'timers' equal to 'no' the bit flag was not properly negated. This patch clears all associated flags and only sets the specified one. pjsip will handle any necessary flag combinations. Also went ahead and did similar for the '100rel' option. ASTERISK-24910 #close Reported by: Ray Crumrine Review: https://reviewboard.asterisk.org/r/4582/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434131 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 14:02 +0000 [95de71f247] gtjoseph * build: Fixes for gcc 5 compilation These are fixes for compilation under gcc 5.0... chan_sip.c: In parse_request needed to make 'lim' unsigned. inline_api.h: Needed to add a check for '__GNUC_STDC_INLINE__' to detect C99 inline semantics (same as clang). ccss.c: In ast_cc_set_parm, needed to fix weird comparison. dsp.c: Needed to work around a possible compiler bug. It was throwing an array-bounds error but neither sgriepentrog, rmudgett nor I could figure out why. manager.c: In action_atxfer, needed to correct an array allocation. This patch will go to 11, 13, trunk. Review: https://reviewboard.asterisk.org/r/4581/ Reported-by: Jeffrey Ollie Tested-by: George Joseph ASTERISK-24932 #close ........ Merged revisions 434113 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 13:18 +0000 [d54ccda3b1] Diederik de Groot (License 6600) * clang compiler warnings: Remove large chunks of unused code from extconf This patch fixes a warning caught by clang, in which it detected that large chunks of extconf were unused. Frankly, I wish we could pretend that all of extconf was unused, but alas, that is not yet the case. A few extraneous functions in the parking tests were removed as well, for the same reason. Review: https://reviewboard.asterisk.org/r/4553 ASTERISK-24917 Reported by: dkdegroot patches: rb4553.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434093 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434097 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 13:03 +0000 [0ecd472e4f] Diederik de Groot (License 6600) * clang compiler warnings: Fix sometimes-uninitialized warning in pbx_config This patch fixes a warning caught by clang, in which a char pointer could be assigned to before it was initialized. The patch re-organizes the code to ensure that the pointer is always initialized, even on off nominal paths. Review: https://reviewboard.asterisk.org/r/4529 ASTERISK-24917 Reported by: dkdegroot patches: rb4529.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434090 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 12:52 +0000 [4e7be5b2dc] Diederik de Groot (License 6600) * clang compiler warnings: Fix format specified in framehook This patch fixes an invalid format specifier used in the formatting of an ERROR message in the framehook code. The format specifier specifies a type of 'unsigned short', but the argument passed to it is of type 'int'. The patch changes the format specifier to 'i'. Review: https://reviewboard.asterisk.org/r/4540 ASTERISK-24917 Reported by: dkdegroot patches: rb4535.patch submitted by dkdegroot (License 6600) ........ Merged revisions 434087 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 11:02 +0000 [2443b40341] Mark Michelson * Ensure that a non-zero sample rate is returned for all formats. Versions of Asterisk prior to 12 defaulted to 8000 as a sample rate if one was not provided by a format. In Asterisk 13, this was removed. The result was that some calculations which involve dividing by the sample rate resulted in dividing by 0. The fix being put in place here is to have the same default fallback that was present in previous versions of Asterisk. Asterisk-24914 #close Reported by Marcello Ceschia git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434046 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 10:16 +0000 [b1102cd642] Corey Farrell * res_pjsip_phoneprov_provider: Revert 433996 / 433997. res_pjsip_phoneprov_provider is using ao2_callback with OBJ_MULTIPLE, then ignoring the return. OBJ_NODATA flag was to prevent a reference leak, but this caused the module to FRACK on unload. Revert change until this can be investigated further. ASTERISK-24935 Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4578/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434025 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-06 09:50 +0000 [0f25076f67] Mark Michelson (license #5049) * ParkedCall: Don't allow dialplan fallthrough after retrieving parked call. This is a change to align behavior with that of Asterisk 11 and previous versions. In those versions, if a parked call were retrieved, and the call ended, the parked call retriever would be hung up after the ParkedCall application ran. Prior to this patch, in Asterisk 13, the same situation would result in the parked call retriever falling through to additional priorities in the extension where the ParkedCall application was called. With this patch, the behavior between Asterisk 11 and 13 aligns. ASTERISK-24899 #close Reported by Malcolm Davenport Patches: ASTERISK-24899.patch uploaded by Mark Michelson(license #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434022 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-05 07:53 +0000 [709fa14b44] Corey Farrell * res_pjsip_phoneprov_provider: Fix leaked OBJ_MULTIPLE iterator. res_pjsip_phoneprov_provider was using ao2_callback with OBJ_MULTIPLE, then ignoring the return. Added OBJ_NODATA flag to prevent a reference leak. ASTERISK-24935 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4578/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433996 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-03 16:53 +0000 [1ee8424f27] Mark Michelson * res_pjsip_messaging: Serialize outbound SIP MESSAGEs Outbound SIP MESSAGEs had the potential to be sent out of order from how they were specified in a set of dialplan steps. This change creates a serializer for sending outbound MESSAGE requests on. This ensures that the MESSAGEs are sent by Asterisk in the same order that they were sent from the dialplan. ASTERISK-24937 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4579 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433968 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-02 09:56 +0000 [169e57d2e0] Scott Griepentrog * pjsip: resolve compatibility problem with ast_sip_session A change in r430179 inserted a variable near the top of a structure caused a problem when running DPMA in a version of Asterisk compiled across the change. This patch moves the new variable to the end of the structure, eliminating the problem. Review: https://reviewboard.asterisk.org/r/4574/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433944 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-02 05:31 +0000 [1eb0c5f4e8] Corey Farrell * Tell menuselect that MALLOC_DEBUG conflicts with DEBUG_CHAOS. DEBUG_CHAOS was marked as conflicting with MALLOC_DEBUG, but for this to work correctly MALLOC_DEBUG must also be marked as conflicting with DEBUG_CHAOS. Review: https://reviewboard.asterisk.org/r/4557/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433923 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-04-01 11:25 +0000 [e301185983] Ashley Sanders * stasis: set a channel variable on websocket disconnect error Resolve compile errors caused by r433863 by fixing the documentation xml to comply with the schema. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-31 22:26 +0000 [a1f12d9231] Ashley Sanders * stasis: set a channel variable on websocket disconnect error Resolve compile errors caused by r433839 by included the missing header file, pbx.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433863 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-31 17:00 +0000 [7293ecd90b] Ashley Sanders * stasis: set a channel variable on websocket disconnect error When an error occurs while writing to a web socket, the web socket is disconnected and the event is logged. A side-effect of this, however, is that any application on the other side waiting for a response from Stasis is left hanging indefinitely (as there is no mechanism presently available for notifying interested parties about web socket error states in Stasis). To remedy this scenario, this patch introduces a new channel variable: STASISSTATUS. The possible values for STASISSTATUS are: SUCCESS - The channel has exited Stasis without any failures FAILED - Something caused Stasis to croak. Some (not all) possible reasons for this: - The app registry is not instantiated; - The app requested is not registered; - The app requested is not active; - Stasis couldn't send a start message ASTERISK-24802 Reported By: Kevin Harwell Review: https://reviewboard.asterisk.org/r/4519/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433839 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-31 11:55 +0000 [94949e7f2f] Richard Mudgett * chan_sip: Fix expression in unit test /channels/chan_sip/test_sip_rtpqos. Fix misplaced parentheses in original fabs() expression. ........ Merged revisions 433816 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433817 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-31 06:47 +0000 [9967739669] Corey Farrell * Re-add _ast_mem_backtrace_buffer variable for ABI compatibility. Modules built prior to commit of r4502 expect to link at runtime to the variable _ast_mem_backtrace_buffer. This change re-adds the variable to the C file only. Review: https://reviewboard.asterisk.org/r/4558/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433795 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-30 06:42 +0000 [2d39bc5528] Corey Farrell * Fix an ABI compatibility issue with ast_log_safe for modules. Binary modules are sometimes built against the latest release of Asterisk in each branch, and need to be compatible with all releases of that branch. This change ensures that utils.h only uses ast_log_safe from the core. For modules and utilities ast_log is used instead. Review: https://reviewboard.asterisk.org/r/4548/ ........ Merged revisions 433772 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433773 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-29 21:44 +0000 [5f8faf16af] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Wabsolute-value warnings This patch fixes several warnings caught by clang - in this case, usage of the abs function on non-integer values. This patch uses labs and fabs, as appropriate, in the various affected files. Review: https://reviewboard.asterisk.org/r/4525 ASTERISK-24917 Reported by: dkdegroot patches: rb4525.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433749 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-29 21:39 +0000 [09b681e344] Diederik de Groot (License 6600) * clang compiler warnings: Fix invalid enum conversion This patch fixes some invalid enum conversion warnings caught by clang. In particular: * chan_sip: Several functions mixed usage of the st_refresher_param enum and st_refresher enum. This patch corrects the functions to use the right enum. * chan_pjsip: Fixed mixed usage of ast_sip_session_t38state and ast_t38_state. * strings: Fixed incorrect usage of AO2 flags with strings container. * res_stasis: Change a return enumeration to stasis_app_user_event_res. Review: https://reviewboard.asterisk.org/r/4535 ASTERISK-24917 Reported by: dkdegroot patches: rb4535.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433746 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-29 21:29 +0000 [7f33abb827] Matt Jordan * main/stdtime/localtime: Fix warning introduced in r433720 The patch in r433720 caused a warning to be kicked back by gcc. It occurred due to this check in unistd.h: if (__nbytes > __bos0 (__buf)) return __read_chk_warn (__fd, __buf, __nbytes, __bos0 (__buf)); That is, if __nbytes is greater than the result of GCC's built-in object size for the struct, we'll kick back a warning. As it turns out, this is because there is an error in the code in the patch. We are passing the address of the pointer to the struct, not iev, which is a pointer to the struct. Hence, the number of bytes is probably going to be lot larger than the number of bytes that make up a pointer! This patch changes the code just read from the pointer to the struct - which fixes the warning. ASTERISK-24917 ........ Merged revisions 433743 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433744 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-29 20:56 +0000 [47eeb67e14] Diederik de Groot (License 6600) * clang compiler warnings: Ignore -Wunused-command-line-argument Asterisk's build system has a tendency to pass include directives for libraries to everything compiled within a particular group of source files. This means we pass the header for libxml2 to things that don't necessarily need it. As a result, we ignore this particular warning. Review: https://reviewboard.asterisk.org/r/4545/ ASTERISK-24917 Reported by: dkdegroot patches: rb4545.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433720 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-29 20:52 +0000 [dbb4d6f9e7] Diederik de Groot (License 6600) * clang compiler warnings: Fix warning for -Wgnu-variable-sized-type-not-at-end This patch fixes a warning caught by clang, wherein a variable sized struct is not located at the end of a struct. While the code in question actually expected this, this is a good warning to watch for. Hence, this patch refactors the code in question to not have two variable length elements in the same struct. Review: https://reviewboard.asterisk.org/r/4530/ ASTERISK-24917 Reported by: dkdegroot patches: rb4530.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433717 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433718 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:56 +0000 [e126ab9eeb] Diederik de Groot (License 6600) * clang compiler warnings: Fix a variety of "unused" warnings This patch fixes the -Wunused-value -Wunused-variable -Wunused-const-variable errors caught by clang. Specifically: * apps/app_queue.c: removed unused qpm_cmd_usage[], qum_cmd_usage[], qsmp_cmd_usage[] * cel/cel_sqlite3_custom.c: removed unused name[] = "cel_sqlite3_custom" * channels/chan_pjsip.c: removed unused desc[] = "PJSIP Channel" * codecs/gsm/src/gsm_create.c: removed unused ident[] = "$Header$" * funcs/func_env.c:729: Fixed ast_str_append_substr. * main/editline/np/strlcat.c: removed unused rcsid variable * main/editline/np/strlcpy.c: removed unused rcsid variable * main/security_events.c: removed unused TIMESTAMP_STR_LEN * utils/conf2ael.c: removed unused cfextension_states * utils/extconf.c: removed unused cfextension_states Review: https://reviewboard.asterisk.org/r/4526 ASTERISK-24917 Reported by: dkdegroot patches: rb4526.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433693 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433694 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:48 +0000 [2f6534527d] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Wself-assign Assigning a variable to itself isn't super useful. However, the WAV format modules make use of this in order to perform byte endian checks. This patch works around the warning by only performing the self assignment if we are going to do more than just assign it to ourselves. Which is odd, but true. Review: https://reviewboard.asterisk.org/r/4544/ ASTERISK-24917 Reported by: dkdegroot patches: rb4544.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433690 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433691 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:40 +0000 [eb70993a50] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Wparantheses-equality warnings Clang will treat ((a == b)) as a warning, as it reasonably expects that the developer may have intended to write (a == b) or ((a = b)). This patch cleans up all instances where equality, not assignment, was intended between two parantheses. Review: https://reviewboard.asterisk.org/r/4531/ ASTERISK-24917 Repoted by: dkdegroot patches: rb4531.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433687 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433688 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:31 +0000 [c0ff16036a] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Wbitfield-constant-conversion warning In chan_iax2, we attempt to assign a -1 to a bitfield. This gets caught by clang, as it will truncate the -1 to a 1 implicitly. Instead, we just assign the value a '1'. Review: https://reviewboard.asterisk.org/r/4537/ ASTERISK-24917 Reported by: dkdegroot patches: rb4537.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433683 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433684 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:27 +0000 [844bc76bef] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Winitializer-overrides This patch fixes clange compiler warnings for initializer overrides. Specifically: res_pjsip/config_transport maps PJSIP_TLSV1_METHOD to the same enumeration value as PJSIP_SSL_DEFAULT_METHOD. When initializing an array containing those enum values, we therefore initialize the value twice to two different values, "tlsv1" and "default". This patch changes it to just initialize the index in the array to "tlsv1". Review: https://reviewboard.asterisk.org/r/4539/ ASTERISK-24917 Reported by: dkdegroot patches: rb4539.patch submitted by dkdegroot (License 6600) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433682 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-28 07:19 +0000 [5e204042d9] Diederik de Groot (License 6600) * clang compiler warnings: Fix -Wunused-function; make inline function static This patch fixes clang compilers warnings for unused functions. Specifically: * channels/chan_iax2: removed user_ref function * main/dsp.c: removed goertzel_update function * main/config.c: made variable_list_switch static Review: https://reviewboard.asterisk.org/r/4527 ASTERISK-24917 Reported by: dkdegroot patches: rb4527.patch submitted by dkdegroot (License 6600) ........ Merged revisions 433678 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433680 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 17:34 +0000 [cfbf5fbe91] Jonathan Rose * SAC: Add a few basic queues Review: https://reviewboard.asterisk.org/r/4503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433658 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 17:25 +0000 [1a50d8d4c2] Jonathan Rose * SAC: Add conferencing extensions and configuration Review: https://reviewboard.asterisk.org/r/4504/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433656 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 16:15 +0000 [c6c08d755d] Rusty Newton * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 2 Example configuration files for a "basic PBX" deployment for the fictitious Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4488/ and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company Patch 4488 includes all functionality needed for SAC's outside connectivity and some externally accessed features, as well as outbound dialing. Reported by: Malcolm Davenport Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/4488/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 16:04 +0000 [13557675d4] Richard Mudgett * res_pjsip_registrar_expire.c: Made use ao2 container template routines and eliminated some RAII_VAR() usage. * Converted the contact_autoexpire container to use the ao2 template hash and cmp functions. Also made use the OBJ_SEARCH_xxx names instead of the deprecated names. * Eliminates several unnecessary uses of RAII_VAR(). Review: https://reviewboard.asterisk.org/r/4524/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 15:30 +0000 [85feac857c] Mark Michelson * Add stateful PJSIP response API call, and use it for out-of-dialog responses. Asterisk had an issue where retransmissions of MESSAGE requests resulted in Asterisk processing the retransmission as if it were a new MESSAGE request. This patch fixes the issue by creating a transaction in PJSIP on the incoming request. This way, if a retransmission arrives, the PJSIP transaction layer will resend the response and Asterisk will not ever see the retransmission. ASTERISK-24920 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4532/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433619 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 12:50 +0000 [dc2cf21144] Richard Mudgett * res_pjsip_registrar_expire.c: Cleanup scheduler leaks on unload/shutdown. Contact expiration object refs were leaked when the module was unloaded. * Made empty the scheduler of entries before destroying it to release the object ref held by the scheduler entry. Review: https://reviewboard.asterisk.org/r/4523/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 09:41 +0000 [6e6f5b3a1f] Justin T. Gibbs (License 6692) * res/res_timing_kqueue: Update the module to conform to current timer API This patch updates the kqueue timing module to conform to current timer API. This fixes issues with using the kqueue timing source on Asterisk 13 on FreeBSD 10. These issues include: - Remove support for kevent64(). The values used to support Asterisk timers fit within 32bits and so can be handled on all platforms via kevent(). - Provide debug logging for, but do not track, unacked events. This matches the behavior of all other timer implementations. - Implement continuous mode by triggering and leaving active, a user event. This ensures that the file descriptor for the timer returns immediately from poll(), without placing the load of a high speed timer on the kernel. - In kqueue_timer_get_max_rate(), don't overstate the capability of the timer. On some platforms, UINT_MAX is greater than INTPTR_MAX, the largest integer type kqueue supports for timers. - In kqueue_timer_get_event(), assume the caller woke up from poll() and just return the mode the timer is currently in. This matches all other timer implementations. - Adjust the test code now that unacked events are not tracked. Review: https://reviewboard.asterisk.org/r/4465/ ASTERISK-24857 #close Reported by: scsiguy Tested by: Ed Hynan patches: rb4465.patch submitted by scsiguy (License 6692) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 07:26 +0000 [b0df413fb2] Corey Farrell * Fix link error for utils/aelparse. Use the standard ast_log instead of ast_log_safe for STANDALONE programs. Review: https://reviewboard.asterisk.org/r/4538/ ........ Merged revisions 433549 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433550 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-27 02:09 +0000 [d01706ce1e] Corey Farrell * Improved and portable ast_log recursion avoidance This introduces a new logger routine ast_log_safe. This routine should be used for all error messages in code that can be run as a result of ast_log. ast_log_safe does nothing if run recursively. All error logging in astobj2.c, strings.c and utils.h have been switched to ast_log_safe. This required adding support for raw threadstorage. This provides direct access to the void* pointer in threadstorage. In ast_log_safe, NULL is used to signify that this thread is not already running ast_log_safe, (void*)1 when it is already running. This was done since it's critical that ast_log_safe do nothing that could log during recursion checking. ASTERISK-24155 #close Reported by: Timo Teräs Review: https://reviewboard.asterisk.org/r/4502/ ........ Merged revisions 433522 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433523 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-26 18:07 +0000 [4b225e2104] Corey Farrell * Fix compile errors caused by r4500 / r4501. * Add ast_register_cleanup to utils/clicompat.c to deal with any utils that copy sources from main. * Asterisk 13+: remove unused variables from core_local.c. Review: https://reviewboard.asterisk.org/r/4534/ ........ Merged revisions 433499 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-26 17:19 +0000 [6adf26f14d] Corey Farrell * Replace most uses of ast_register_atexit with ast_register_cleanup. Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-26 12:46 +0000 [d0df545a44] Corey Farrell * res_pjsip: Enable unload of all modules at shutdown. * Move most of res_pjsip:module_unload to unload_pjsip to resolve crashes caused by running PJSIP functions from non-PJSIP threads. * Remove call to pjsip_endpt_destroy(ast_pjsip_endpoint), it was causing crashes in some cases. In theory pj_shutdown() should take care of this. * Mark res_pjsip_keepalive and res_pjsip_session as allowed to unload at shutdown. * Resolve leaked config global in res_pjsip_notify. * Unregister pubsub pjsip service module. * Implement cleanup for res_pjsip_session. ASTERISK-24731 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4498/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433469 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-26 12:04 +0000 [fd434a210f] Kevin Harwell * app_confbridge: file playback blocks dtmf Attempting to execute DTMF in a confbridge while file playback (prompt, announcement, etc) is occurring is not allowed. You have to wait until the sound file has completed before entering DTMF. This patch fixes it so that app_confbridge now monitors for dtmf key presses during menu driven file playback. If a key is pressed playback stops and it executes the matched menu option. ASTERISK-24864 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4510/ ........ Merged revisions 433445 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433446 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-25 13:37 +0000 [dea885a607] Richard Mudgett * A couple minor cleanup tweaks. * In res/res_sorcery_realtime.c: Broke long line. * In main/bucket.c: Eliminated unnecessary NULL check as ast_sorcery_unref() is NULL tolerant and set the global object to NULL after unref in the system shutdown bucket_cleanup(). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433420 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-25 10:30 +0000 [05de9082a5] Simon Arlott (License 5756) * res_xmpp: Buddies are always auto-registered when processing the roster Due to a quirk in the configuration handling of res_xmpp, the 'autoregister' setting was never actually processed. This was due to not properly copying over the global settings to the client settings when applying the configuration to the run-time object. Review: https://reviewboard.asterisk.org/r/4496/ ASTERISK-14233 ASTERISK-24780 #close Reported by: Simon Arlott patches: asterisk-13.1.0-24780 uploaded by Simon Arlott (License 5756) ........ Merged revisions 433395 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433396 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-24 14:26 +0000 [b1e9552b08] Richard Mudgett * chan_pjsip: Add "rpid_immediate" option to prevent unnecessary "180 Ringing" messages. Incoming PJSIP call legs that have not been answered yet send unnecessary "180 Ringing" or "183 Progress" messages every time a connected line update happens. If the outgoing channel is also PJSIP then the incoming channel will always send a "180 Ringing" or "183 Progress" message when the outgoing channel sends the INVITE. Consequences of these unnecessary messages: * The caller can start hearing ringback before the far end even gets the call. * Many phones tend to grab the first connected line information and refuse to update the display if it changes. The first information is not likely to be correct if the call goes to an endpoint not under the control of the first Asterisk box. When connected line first went into Asterisk in v1.8, chan_sip received an undocumented option "rpid_immediate" that defaults to disabled. When enabled, the option immediately passes connected line update information to the caller in "180 Ringing" or "183 Progress" messages as described above. * Added "rpid_immediate" option to prevent unnecessary "180 Ringing" or "183 Progress" messages. The default is "no" to disable sending the unnecessary messages. ASTERISK-24781 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4473/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433338 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-22 18:58 +0000 [a3fe43fbdc] snuffy (License 5024) * Fix compilations errors on 64-bit OpenBSD systems In versiong 5.5, OpenBSD went to 64-bit time values. This requires a cast to (long) when printing members of certain time structs. Review: https://reviewboard.asterisk.org/r/4507 ASTERISK-24879 #close Reported by: snuffy Tested by: snuffy patches: openbsd-time64.diff uploaded by snuffy (License 5024) ........ Merged revisions 433268 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433269 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-22 18:04 +0000 [08a88aab15] snuffy (License 5024) * Fix compilation issues for OpenBSD This patch addresses compilation issues for OpenBSD. Specifically, it addresses: * It allows including in asterisk.c * Provides a needed (size_t) cast in xmldoc.c In 13+, it also addresses a conditional inclusion in loader.c. Review: https://reviewboard.asterisk.org/r/4506 ASTERISK-24880 #close Reported by: snuffy Tested by: snuffy patches: misc-openbsd.diff uploaded by snuffy (License 5024) ........ Merged revisions 433245 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433247 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-20 14:52 +0000 [6ca98524bf] Richard Mudgett * Audit ast_pjsip_rdata_get_endpoint() usage for ref leaks. Valgrind found some memory leaks associated with ast_pjsip_rdata_get_endpoint(). The leaks would manifest when sending responses to OPTIONS requests, processing MESSAGE requests, and res_pjsip supplements implementing the incoming_request callback. * Fix ast_pjsip_rdata_get_endpoint() endpoint ref leaks in res/res_pjsip.c:supplement_on_rx_request(), res/res_pjsip/pjsip_options.c:send_options_response(), res/res_pjsip_messaging.c:rx_data_to_ast_msg(), and res/res_pjsip_messaging.c:send_response(). * Eliminated RAII_VAR() use with ast_pjsip_rdata_get_endpoint() in res/res_pjsip_nat.c:nat_on_rx_message(). * Fixed inconsistent but benign return value in res/res_pjsip/pjsip_options.c:options_on_rx_request(). Review: https://reviewboard.asterisk.org/r/4511/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433222 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-20 13:23 +0000 [1c09028171] Richard Mudgett * res_pjsip_sdp_rtp,sorcery: Fix invalid access and memory leak respectively. Valgrind found a memory leak and invalid access. * Fix invalid access by sscanf() being fed a non-nul terminated string of digits in res/res_pjsip_sdp_rtp.c:get_codecs(). * Fix memory leak in main/sorcery.c:sorcery_object_field_destructor(). * Fix potential NULL pointer dereference in main/xmldoc.c:xmldoc_get_syntax_config_option(). Review: https://reviewboard.asterisk.org/r/4513/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-19 14:19 +0000 [73dcea59bd] Matt Jordan * funcs/func_env: Fix regression caused in FILE read operation When r432935 was merged, it did correctly fix a situation where a FILE read operation on the middle of a file buffer would not read the requested length in the parameters passed to the FILE function. Unfortunately, it would also allow the FILE function to append more bytes than what was available in the buffer if the length exceeded the end of the buffer length. This patch takes the minimum of the remaining bytes in the buffer along with the calculated length to append provided by the original patch, and uses that as the length to append in the return result. This patch also updates the unit tests with the scenarios that were originally pointed out in ASTERISK-21765 that the original implementation treated incorrectly. ASTERISK-21765 ........ Merged revisions 433173 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433174 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-19 05:20 +0000 [4c84dca2d8] Corey Farrell * logger: Apply default console logging when configuration cannot be loaded. When logger.conf is missing or invalid enable console logging and display an error message. ASTERISK-24817 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4497/ ........ Merged revisions 433122 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433126 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-19 04:53 +0000 [958bc84caf] Corey Farrell * chan_sip: Simplify dialog/peer references, improve REF_DEBUG output. * Replace functions for ref/undef of dialogs and peers with macro's to call ao2_t_bump/ao2_t_cleanup. * Enable passthough of REF_DEBUG caller information to sip_alloc and find_call. ASTERISK-24882 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4189/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433115 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-19 04:44 +0000 [7fddae99dd] Corey Farrell * chan_sip: Fix dialog reference leaked to scheduler for reinvite_timeout. Release the scheduler reference to the dialog for reinvite timeout during dialog_unlink_all. ASTERISK-24876 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4491/ ........ Merged revisions 433112 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433113 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 21:34 +0000 [dba0f1ad67] Richard Mudgett * res_pjsip_session: Fix off-nominal extra unref of session. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433088 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 17:15 +0000 [2c7b945149] Scott Griepentrog * Various: bugfixes found via chaos Using DEBUG_CHAOS several instances of a null pointer crash, and one uninitialized variable were uncovered and fixed. Also added details on why Asterisk failed to initialize. Review: https://reviewboard.asterisk.org/r/4468/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433064 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 16:57 +0000 [1fb1c81923] Scott Griepentrog * core: Introduce chaos into memory allocations Locate potential crashes by exercising seldom used code paths. This patch introduces a new define DEBUG_CHAOS, and mechanism to randomly return an error condition from functions that will seldom do so. Functions that handle the allocation of memory get the first treatment. Review: https://reviewboard.asterisk.org/r/4463/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 16:49 +0000 [2122c205e6] Richard Mudgett * Audit ast_sockaddr_resolve() usage for memory leaks. Valgrind found some memory leaks associated with ast_sockaddr_resolve(). Most of the leaks had already been fixed by earlier memory leak hunt patches. This patch performs an audit of ast_sockaddr_resolve() and found one more. * Fix ast_sockaddr_resolve() memory leak in apps/app_externalivr.c:app_exec(). * Made main/netsock2.c:ast_sockaddr_resolve() always set the addrs parameter for safety so the pointer will never be uninitialized on return. The same goes for res/res_pjsip_acl.c:extract_contact_addr(). * Made functions that call ast_sockaddr_resolve() with RAII_VAR() controlling the addrs variable use ast_free instead of ast_free_ptr to provide better MALLOC_DEBUG information. Review: https://reviewboard.asterisk.org/r/4509/ ........ Merged revisions 433056 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433057 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 13:34 +0000 [94fe4a9178] Kevin Harwell * res_pjsip: Allow configuration of endpoint identifier query order Updated some documentation stating that endpoint identifiers registered without a name are place at the front of the lookup list. Also renamed register method 'ast_sip_register_endpoint_identifier_by_name' to 'ast_sip_register_endpoint_identifier_with_name' ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 13:20 +0000 [1f428f25f0] Kevin Harwell * res_pjsip: Allow configuration of endpoint identifier query order This patch fixes previously reverted code that caused binary incompatibility problems with some modules. And like the original patch it makes sure that no matter what order the endpoint identifier modules were loaded, priority is given based on the ones specified in the new global 'endpoint_identifier_order' option. ASTERISK-24840 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433028 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-17 11:10 +0000 [522f063186] Richard Mudgett * res_pjsip: Add reason comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433005 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 21:28 +0000 [5c03a5f2e7] Matt Jordan * main/frame: Don't report empty disallow values as an error In realtime, it is normal to have a database with both 'allow' and 'disallow' columns in the schema. It is perfectly valid to have an 'allow' value of '!all,g722,ulaw,alaw' and no 'disallow' value. Unlike in static conf files, you can't *not* provide the disallow value. Thus, the empty disallow value causes a spurious WARNING message, which is kind of annoying. This patch makes it so that a 'disallow' value with no ... value ... is ignored. Granted, you can still screw this up as well, as technically specifying 'disallow=all,!ulaw' allows only ulaw, and then you would have no 'allow' value in your database. But really, why would you do that? WHY? ASTERISK-16779 #close Reported by: Atis Lezdins ........ Merged revisions 432970 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432971 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 21:00 +0000 [f7c6bedb06] Joshua Colp * func_curl: Don't hold exclusive lock when performing HTTP request. This code originally kept a lock held when performing the HTTP request to ensure that the options provided to curl remain valid. This doesn't seem to be necessary these days and holding the lock caused requests to happen sequentially instead of in parallel. ASTERISK-18708 #close Reported by: Dave Cabot ........ Merged revisions 432948 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432949 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 20:36 +0000 [287a22435f] Joshua Colp * core: Fix tab completion of "core set debug channel" CLI command. The "core set debug channel" CLI command mistakenly had source filenames added to its tab completion. This occurred because the CLI generator fell back to the "core set debug" command which permits setting debug at a source filename level. ASTERISK-21038 #close Reported by: Richard Kenner ........ Merged revisions 432944 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432945 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 20:21 +0000 [37d33ed997] Di-Shi Sun (License 5076) * FILE: fix retrieval of file contents when offset is specified The loop that reads in a file was not correctly using the offset when determining what bytes to append to the output. This patch corrects the logic such that the correct portion of the file is extracted when an offset is specified. ASTERISK-21765 Reported by: John Zhong Tested by: Matt Jordan, Di-Shi Sun patches: file_read_390821.patch uploaded by Di-Shi Sun (License 5076) ........ Merged revisions 432935 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432938 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 19:18 +0000 [a4c27baf47] Matt Jordan * apps/app_amd: Document maximum_word_length option; fix AMDCAUSE documentation This patch corrects the documentation for the AMD application. Specifically: * It documents the maximum_word_length option, which limits the maximum allowed length of a single utterance. * It clarifies the AMDCAUSE values MAXWORDS and MAXWORDLENGTH. MAXWORDLENGTH was documented as MAXWORDS, while MAXWORDS was undocumented. Thanks to the issue reporter, Frank DiGennaro, for pointing out the issues. ASTERISK-19470 #close Reported by: Frank DiGennaro ........ Merged revisions 432918 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432920 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 12:04 +0000 [a3292230b8] Richard Mudgett * chan_pjsip: AMI action PJSIPShowEndpoint closes AMI connection on error. Also fixed similar problem with AMI action PJSIPShowEndpoints. ASTERISK-24872 #close Reported by: Dmitriy Serov Review: https://reviewboard.asterisk.org/r/4487/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432894 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 11:26 +0000 [34aa0214eb] Richard Mudgett * chan_pjsip/res_pjsip_callerid: Make Party ID handling simpler and consistent. The res_pjsip modules were manually checking both name and number presentation values when there is a function that determines the combined presentation for a party ID struct. The function takes into account if the name or number components are valid while the manual code rarely checked if the data was even valid. * Made use ast_party_id_presentation() rather than manually checking party ID presentation values. * Ensure that set_id_from_pai() and set_id_from_rpid() will not return presentation values other than what is pulled out of the SIP headers. It is best if the code doesn't assume that AST_PRES_ALLOWED and AST_PRES_USER_NUMBER_UNSCREENED are zero. * Fixed copy paste error in add_privacy_params() dealing with RPID privacy. * Pulled the id->number.valid test from add_privacy_header() and add_privacy_params() up into the parent function add_id_headers() to skip adding PAI/RPID headers earlier. * Made update_connected_line_information() not send out connected line updates if the connected line number is invalid. Lower level code would not add the party ID information and thus the sent message would be unnecessary. * Eliminated RAII_VAR usage in send_direct_media_request(). Review: https://reviewboard.asterisk.org/r/4472/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432892 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-13 09:48 +0000 [0497b7b155] Kevin Harwell * Revert - res_pjsip: Allow configuration of endpoint identifier query order Due to a break in binary compatibility with some other modules these changes are being reverted until the issue can be resolved. ASTERISK-24840 Reported by: Mark Michelson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-12 07:58 +0000 [b9fd61f2c7] Matt Jordan * main/audiohook: Update internal sample rate on reads When an audiohook is created (which is used by the various Spy applications and Snoop channel in Asterisk 13+), it initially is given a sample rate of 8kHz. It is expected, however, that this rate may change based on the media that passes through the audiohook. However, the read/write operations on the audiohook behave very differently. When a frame is written to the audiohook, the format of the frame is checked against the internal sample rate. If the rate of the format does not match the internal sample rate, the internal sample rate is updated and a new SLIN format is chosen based on that sample rate. This works just fine. When a frame is read, however, we do something quite different. If the format rate matches the internal sample rate, all is fine. However, if the rates don't match, the audiohook attempts to "fix up" the number of samples that were requested. This can result in some seriously large number of samples being requested from the read/write factories. Consider the worst case - 192kHz SLIN. If we attempt to read 20ms worth of audio produced at that rate, we'd request 3840 samples (192000 / (1000 / 20)). However, if the audiohook is still expecting an internal sample rate of 8000, we'll attempt to "fix up" the requested samples to: samples_converted = samples * (ast_format_get_sample_rate(format) / (float) audiohook->hook_internal_samp_rate); which is: 92160 = 3840 * (192000 / 8000) This results in us attempting to read 92160 samples from our factories, as opposed to the 3840 that we actually wanted. On a 64-bit machine, this miraculously survives - despite allocating up to two buffers of length 92160 on the stack. The 32-bit machines aren't quite so lucky. Even in the case where this works, we will either (a) get way more samples than we wanted; or (b) get about 3840 samples, assuming the timing is pretty good on the machine. Either way, the calculation being performed is wrong, based on the API users expectations. My first inclination was to allocate the buffers on the heap. As it is, however, there's at least two drawbacks with doing this: (1) It's a bit complicated, as the size of the buffers may change during the lifetime of the audiohook (ew). (2) The stack is faster (yay); the heap is slower (boo). Since our calculation is flat out wrong in the first place, this patch fixes this issue by instead updating the internal sample rate based on the format passed into the read operation. This causes us to read the correct number of samples, and has the added benefit of setting the audihook with the right SLIN format. Note that this issue was caught by the Asterisk Test Suite as a result of r432195 in the 13 branch. Because this issue is also theoretically possible in Asterisk 11, the change is being made here as well. Review: https://reviewboard.asterisk.org/r/4475/ ........ Merged revisions 432810 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432811 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-12 07:39 +0000 [f5bc032567] Diederik de Groot (License 6600) * Add support for the clang compiler; update RAII_VAR to use BlocksRuntime RAII_VAR, which is used extensively in Asterisk to manage reference counted resources, uses a GCC extension to automatically invoke a cleanup function when a variable loses scope. While this functionality is incredibly useful and has prevented a large number of memory leaks, it also prevents Asterisk from being compiled with clang. This patch updates the RAII_VAR macro such that it can be compiled with clang. It makes use of the BlocksRuntime, which allows for a closure to be created that performs the actual cleanup. Note that this does not attempt to address the numerous warnings that the clang compiler catches in Asterisk. Much thanks for this patch goes to: * The folks on StackOverflow who asked this question and Leushenko for providing the answer that formed the basis of this code: http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang * Diederik de Groot, who has been extremely patient in working on getting this patch into Asterisk. Review: https://reviewboard.asterisk.org/r/4370/ ASTERISK-24133 ASTERISK-23666 ASTERISK-20399 ASTERISK-20850 #close Reported by: Diederik de Groot patches: RAII_CLANG.patch uploaded by Diederik de Groot (License 6600) ........ Merged revisions 432807 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432808 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-11 11:38 +0000 [bd029688cd] Richard Mudgett * res_pjsip: Move internal init/destroy prototypes to private header file. Done as a separate commit from a finding in https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432787 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-11 10:24 +0000 [c24a294f0b] Richard Mudgett * res_pjsip: Fix pjsip.conf type=global object default value handling. When a type=global section is not defined in pjsip.conf the global defaults are not applied. As a result the mandatory Max-Forwards header is not added to SIP messages for res_pjsip/chan_pjsip. The handling of pjsip.conf type=global objects has several problems: 1) If the global object is missing the defaults are not applied. 2) If the global object is missing the default_outbound_endpoint's default value is not returned by ast_sip_global_default_outbound_endpoint(). 3) Defines are needed so default values only need to be changed in one place. * Added a sorcery instance observer callback to check if there were any type=global sections loaded. If there were more than one then issue an error message. If there were none then apply the global defaults. * Fixed ast_sip_global_default_outbound_endpoint() to return the documented default when no type=global object is defined. * Made defines for the global default values. * Increased the default_useragent[] size because SVN version strings can get lengthy and 128 characters may not be enough. * Fixed an off-nominal code path ref leak in global_alloc() if the string fields fail to initialize. * Eliminated RAII_VAR in get_global_cfg() and ast_sip_global_default_outbound_endpoint(). ASTERISK-24807 #close Reported by: Anatoli Review: https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432766 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-11 10:18 +0000 [737064bfa4] Richard Mudgett * res_pjsip: Fixed invalid empty Server and User-Agent SIP headers. Setting pjsip.conf useragent to an empty string results in an empty SIP header being sent. * Made not add an empty SIP header item to the global SIP headers list. Review: https://reviewboard.asterisk.org/r/4467/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432764 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-10 18:09 +0000 [bc357c1d7e] Joshua Colp * core: Don't create snapshots with locks. Snapshots are immutable and are never changed. Allocating them with a lock is wasteful. Review: https://reviewboard.asterisk.org/r/4469/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432742 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-10 16:33 +0000 [afea98dc73] Javier Acosta (License 6690) * res/res_config_odbc: Fix improper escaping of backslashes with MySQL When escaping backslashes with MySQL, the proper way to escape the characters in a LIKE clause is to escape the '\' four times, i.e., '\\\\'. To quote the MySQL manual: "Because MySQL uses C escape syntax in strings (for example, “\n” to represent a newline character), you must double any “\” that you use in LIKE strings. For example, to search for “\n”, specify it as “\\n”. To search for “\”, specify it as “\\\\”; this is because the backslashes are stripped once by the parser and again when the pattern match is made, leaving a single backslash to be matched against." ASTERISK-24808 #close Reported by: Javier Acosta patches: res_config_odbc.diff uploaded by Javier Acosta (License 6690) ........ Merged revisions 432720 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432721 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-10 13:13 +0000 [055001716c] Graham Barnett (License 6685) * app_voicemail: Fix crash with IMAP backends when greetings aren't present When an IMAP backend is in use and greetings are set to be used, but aren't present for a user in their IMAP folder, Asterisk will crash. This occurs due to the mailstream being set to the 'greetings' folder and being left in that particular state, regardless of the success/failure of the attempt to access the folder the mailstream points to. Later access of the mailstream assumes that it points to the 'INBOX' (or some other folder), resulting in either a crash (if the greetings folder didn't exist and the mailstream is invalid) or an inability to read messages from the 'INBOX' folder. This patch restores the mailstream to its correct state after accessing the greetings. This fixes the crash, and sets the mailstream to the state that VoiceMailMain expects. Note that while ASTERISK-23390 also contained a patch for this issue, the patch on ASTERISK-24786 is the one being merged here. Review: https://reviewboard.asterisk.org/r/4459/ ASTERISK-23390 #close Reported by: Ben Smithurst ASTERISK-24786 #close Reported by: Graham Barnett Tested by: Graham Barnett patches: app_voicemail.c.patch.SIGSEGV3rev2 uploaded by Graham Barnett (License 6685) ........ Merged revisions 432695 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432696 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-10 12:47 +0000 [92178247ee] Ed Hynan (Licnese 6680) * localtime: Fix file descriptor leak on kqueue(2) systems The localtime management in the Asterisk core contains a thread that watches for changes in the local timezone. On systems where the directory containing /etc/localtime is modified frequently, the thread monitoring the changes will be woken up to determine if any changes in timezone have occurred. When using kqueue(2), this can cause a leak of file descriptors due to some improper management of resources. This patch updates the kqueue(2) handling in localtime, such that is no longer leaks resources. Review: https://reviewboard.asterisk.org/r/4450/ ASTERISK-24739 #close Reported by: Ed Hynan patches: 11.15.0-u.diff uploaded by Ed Hynan (Licnese 6680) 11.7.0-u.diff uploaded by Ed Hynan (License 6680) svn-trunk-Jan-26-2015-u.diff uploaded by Ed Hynan (License 6680) ........ Merged revisions 432691 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-10 11:04 +0000 [cae712d986] Richard Mudgett * res_pjsip_refer: Fix occasional unexpected BYE sent after receiving a REFER. A race condition happened between initiating a transfer and requesting that a dialog termination be delayed. Occasionally, the transferrer channels would exit the bridge and hangup before the dialog termination delay was requested. * Made request dialog termination delay before initiating the transfer action. If the transfer fails then cancel the delayed dialog termination request. ASTERISK-24755 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4460/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432668 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-09 11:12 +0000 [110b99646c] Kevin Harwell * res_pjsip: Allow configuration of endpoint identifier query order It's possible to have a scenario that will create a conflict between endpoint identifiers. For instance an incoming call could be identified by two different endpoint identifiers and the one chosen depended upon which identifier module loaded first. This of course causes problems when, for example, the incoming call is expected to be identified by username, but instead is identified by ip. This patch adds a new 'global' option to res_pjsip called 'endpoint_identifier_order'. It is a comma separated list of endpoint identifier names that specifies the order by which identifiers are processed and checked. ASTERISK-24840 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4455/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-07 19:46 +0000 [714cb27000] Joshua Colp * res_rtp_asterisk: Fix wrongful use of USE_PJPROJECT define. As pjproject is now used as a shared library a different define, HAVE_PJPROJECT, is used to specify if pjproject is present. ASTERISK-24830 #close Reported by: Stefan Engström git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432614 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-06 16:50 +0000 [e158517a9c] Richard Mudgett * res_pjsip_refer: Make safely get the context for a blind transfer. Made safely get the TRANSFER_CONTEXT channel value while the channel is locked in refer_incoming_attended_request() and refer_incoming_blind_request(). The pointer returned by pbx_builtin_getvar_helper() is only valid while the channel is locked. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-06 16:12 +0000 [5d16d80b59] Richard Mudgett * res_pjsip_refer: Made refer_attended_alloc() not create the ao2 object with a lock. The lock is unused. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-06 15:11 +0000 [772793f18e] Jonathan Rose * app: Add functions to swap voicemail function table for testing purposes git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432556 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-06 14:18 +0000 [8cced7767c] Richard Mudgett * chan_dahdi/sig_analog: Fix distinctive ring detection to suck less. The distinctive ring feature interferes with detecting Caller ID and appears to have been broken for years. What happens is if you have a ring-ring cadence as used in the UK you get too many DAHDI events for the distinctive ring pattern array and Caller ID detection is aborted. I think when Zapata/DAHDI added the ring begin event it broke distinctive ring. More events happen than before and the code does no filtering of which event times are recorded in the pattern array. * Made distinctive ring only record the ringt count when the ring ends instead of on just any DAHDI event. Distinctive ring can be ring, ring-ring, ring-ring-ring, or different ring durations for the up to three rings. * Fixed the distinctive ring detection enable (chan_dahdi.conf option usedistinctiveringdetection) to be per port instead of somewhat per port and somewhat global. This has been broken since v1.8. * Fixed using the default distinctive ring context when the detected pattern does not match any configured dringX patterns. The default context did not get set when the previous call was a matched distinctive ring pattern and the current call is not matched. This has been broken since v1.8. * Made distinctive ring have no effect on Caller ID detection when it is disabled. Caller ID detection just monitors for 10 seconds before giving up. * Fixed leak of struct callerid_state memory when a polarity reversal during Caller ID detection causes the incoming call to be aborted. DAHDI-1143 AST-1545 ASTERISK-24825 #close Reported by: Richard Mudgett ASTERISK-17588 Reported by: Daniel Flounders Review: https://reviewboard.asterisk.org/r/4444/ ........ Merged revisions 432530 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432534 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-06 13:31 +0000 [13e715b30c] Richard Mudgett * chan_sip: Fix realtime locking inversion when poking a just built peer. When a realtime peer is built it can cause a locking inversion when the just built peer is poked. If the CLI command "sip show channels" is periodically executed then a deadlock can happen because of the locking inversion. * Push the peer poke off onto the scheduler thread to avoid the locking inversion of the just built realtime peer. AST-1540 ASTERISK-24838 #close Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/4454/ ........ Merged revisions 432526 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432528 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-05 10:38 +0000 [06fa8db864] gtjoseph * app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE. There is a leftover "assert" in app_voicemail/__messagecount that references variables that don't exist. This causes the compile to fail when --enable-dev-mode and IMAP_STORAGE are selected. This patch removes the assert. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4461/ ........ Merged revisions 432484 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432485 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-04 12:52 +0000 [999d96d405] Matt Jordan * translate: Prevent invalid memory accesses on fast shutdown When a 'core restart now' or 'core stop now' is executed and a channel is currently in a media operation, the translator matrix can be destroyed while a channel is currently blocked on getting the best translation choice (see ast_translator_best_choice). When the channel gets the mutex, the translation matrix now has invalid memory, and Asterisk crashes. This patch does two things: (1) We now only clean up the translation matrix on a graceful shutdown. In that case, there are no channels, and so there is no risk of this occurring. (2) We also now set the __matrix and __indextable to NULL. In some initial backtraces when this occurred, it looked as if there was a memory corruption occurring, and it wasn't until we determined that something had restarted Asterisk that the issue became clear. By setting these to NULL on shutdown, it becomes a bit easier to determine why a crash is occurring. Note that we could litter the code with NULL checks on the __matrix, but the act of making the translation matrix cleaned up on shutdown should preclude this issue from occurring in the first place, and this part of the code needs to be as fast as possible. Review: https://reviewboard.asterisk.org/r/4457/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432453 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-03-02 13:14 +0000 [9cdadc168c] Matt Jordan * res/res_pjsip_sdp_rtp: Revert portion of r432195 Unfortunately, while initial testing with ConfBridge did not reproduce the audio problem alluded to in the comment in res_pjsip_sdp_rtp, further testing did show that bridge_softmix and/or ConfBridge has a severe problem bridging two or more participants at different sampling rates. Sometimes, it even picks odd sampling rates that cause hideous audio problems. This patch backs out the offending portion of the code until the issues in the affected bridging modules can be more properly analyzed. ASTERISK-24841 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432423 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-27 12:23 +0000 [9d85e855de] Richard Mudgett * ARI: Fix crash if integer values used in JSON payload 'variables' object. Sending the following ARI commands caused Asterisk to crash if the JSON body 'variables' object passes values of types other than strings. POST /ari/channels POST /ari/channels/{channelid} PUT /ari/endpoints/sendMessage PUT /ari/endpoints/{tech}/{resource}/sendMessage * Eliminated RAII_VAR usage in ast_ari_channels_originate_with_id(), ast_ari_channels_originate(), ast_ari_endpoints_send_message(), and ast_ari_endpoints_send_message_to_endpoint(). ASTERISK-24751 #close Reported by: jeffrey putnam Review: https://reviewboard.asterisk.org/r/4447/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-26 12:52 +0000 [c33c5183a5] Scott Griepentrog * Dial API: add self destruct option when complete This patch adds a self-destruction option to the dial api. The usefulness of this is mostly when using async mode to spawn a separate thread used to handle the new call, while the calling thread is allowed to go on about other business. The only alternative to this option would be the calling thread spawning a new thread, or hanging around itself waiting to destroy the dial struct after completion. Example of use (minus error checking): struct ast_dial *dial = ast_dial_create(); ast_dial_append(dial, "PJSIP", "200", NULL); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_ANSWER_EXEC, "Echo"); ast_dial_option_global_enable(dial, AST_DIAL_OPTION_SELF_DESTROY, NULL); ast_dial_run(dial, NULL, 1); The dial_run call will return almost immediately after spawning the new thread to run and monitor the dial. If the call is answered, it is placed into the echo app. When completed, it will call ast_dial_destroy() on the dial structure. Note that any allocations made to pass values to ast_dial_set_user_data() or dial options must be free'd in a state callback function on any of: AST_DIAL_RESULT_UNASWERED, AST_DIAL_RESULT_ANSWERED, AST_DIAL_RESULT_HANGUP, or AST_DIAL_RESULT_TIMEOUT. Review: https://reviewboard.asterisk.org/r/4443/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432385 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-26 11:07 +0000 [169058e73f] Kevin Harwell * app_chanspy, channel: fix frame leaks Fixed a couple of frame leaks that were found during testing. ASTERISK-24828 #close Reported by: John Hardin Review: https://reviewboard.asterisk.org/r/4445/ ........ Merged revisions 432362 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 22:58 +0000 [de86b30dba] Matt Jordan * make: Remove 'res_features' from libraries to link against with cygwin/mingw32 Both the apps and channels Makefiles still listed 'res_features' as modules to link against when compiling for cygwin or mingw32. This module hasn't existed for quite some time. ASTERISK-18105 #close Reported by: feyfre ........ Merged revisions 432341 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432342 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 21:03 +0000 [34989bd9c8] Makoto Dei (License 5027) * channels/chan_sip: Don't send a BYE after final response when PBX thread fails When Asterisk fails to start a PBX thread for a new channel - for example, when the maxcalls setting in asterisk.conf is exceeded - we currently send a final response, and then attempt to send a BYE request to the UA. Since that's all sorts of wrong, this patch fixes that by setting sipalreadygone on the sip_pvt such that we don't get stuck sending BYE requests to something that does not want it. Note that this patch is a slight modification of the one on ASTERISK-15434. For clarity, it explicitly calls sipalreadygone with the calls to transmit a final response. ASTERISK-21845 ASTERISK-15434 #close Reported by: Makoto Dei Tested by: Matt Jordan patches: sip-pbxstart-failed.patch uploaded by Makoto Dei (License 5027) ........ Merged revisions 432320 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432321 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 17:48 +0000 [53aec7a969] Rusty Newton * configs/basic-pbx - Super Awesome Company example configs Phase 1, Patch 1 Example configuration files for a "basic PBX" deployment for the fictitious Super Awesome Company. Details at https://reviewboard.asterisk.org/r/4379/ and https://wiki.asterisk.org/wiki/display/AST/Super+Awesome+Company Reported by: Malcolm Davenport Tested by: Rusty Newton Review: https://reviewboard.asterisk.org/r/4379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432301 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 17:09 +0000 [474fec4f92] Matt Jordan * configure: Promote SQLite3 "not installed" warning to error Since Asterisk won't build without the library, not having it is definitely an error. Thanks to Kyle Kurz for pointing this out. ........ Merged revisions 432280 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432281 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 17:02 +0000 [ddff640f94] Matt Jordan * channels/chan_sip: Clarify WARNING message in mismatched SRTP scenario When we receive an SDP as part of an offer/answer for a peer/friend has been configured to require encryption, and that SDP offer/answer failed to provide acceptable crypto attributes, we currently issue a WARNING that uses the phrase "we" and "requested". In this case, both of those terms are ambiguous - the user will probably think "we" is Asterisk (it most likely isn't) and it may not be a "request", so much as an SDP that was received in some fashion. This patch makes the WARNING messages slightly less bad and a bit more accurate as well. ASTERISK-23214 #close Reported by: Rusty Newton ........ Merged revisions 432277 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 15:42 +0000 [dd8ac00f24] Olle Johansson (License 5267) * channels/sip/sdp_crypto: Handle SRTP keys negotiated with key lifetime/MKI Prior to this patch, SDP offers negotiating SDES-SRTP crypto attributes would be rejected if those crypto attributes contained either a key lifetime or a MKI parameter. While from a theoretical point of view this was defensible - Asterisk does not support key lifetimes or multiple crypto keys - from a practical point of view, this is quite a problem. A large number of endpoints offer lifetimes/MKI, which Asterisk can tolerate so long as it doesn't actually have to support anything more than a single key or refresh the key. In reality, this is (so far as we've seen) always the case. This patch is a forward port of Olle's work in the lingon-srtp-key-lifetime-1.8 branch. To quote Olle from ASTERISK-17721, it handles lifetime/MKI parameters in the following fashion: > The Lingon branch now handle lifetime and MKI parameters. > > We only accept lifetimes up to max for the crypto and higher than 10 hours > for packetization of 20 ms (50 pps). > > We only handle MKI with index 1. > > We do not really bother with counting packets and reinviting at end of > lifetime, so the min of 10 hours kind of takes care of most calls. If there > are longer ones, we rely on the other side for re-invites. > > It's still not perfect, but I personally think this is an improvement. A > configuration option for minimum lifetime accepted could be added. When the patch was ported forward, I decided against adding a configuration option as Olle's handling was more than sufficient for every case I've seen come through the issue tracker or through interoperability testing. We can revisit that decision if it proves to be false. A few small other tweaks were made to the surrounding code to reduce indentation and provide better type safety for the 'tag' parameter. Review: https://reviewboard.asterisk.org/r/4419/ Review: https://reviewboard.asterisk.org/r/4418/ ASTERISK-17721 #close Reported by: Terry Wilson ASTERISK-17899 #close Reported by: Dwayne Hubbard patches: lingon-srtp-key-lifetime-1.8.diff uploaded by oej (License 5267) ASTERISK-20233 Reported by: tootai ASTERISK-22748 Reported by: Alejandro Mejia ........ Merged revisions 432239 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432258 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-25 14:44 +0000 [43a3e80be1] David M. Lee * Increase WebSocket frame size and improve large read handling Some WebSocket applications, like [chan_respoke][], require a larger frame size than the default 8k; this patch bumps the default to 16k. This patch also fixes some problems exacerbated by large frames. The sanity counter was decremented on every fread attempt in ws_safe_read(), regardless of whether data was read from the socket or not. For large frames, this could result in loss of sanity prior to reading the entire frame. (16k frame / 1448 bytes per segment = 12 segments). This patch changes the sanity counter so that it only decrements when fread() doesn't read any bytes. This more closely matches the original intention of ws_safe_read(), given that the error message is "Websocket seems unresponsive". This patch also properly logs EOF conditions, so disconnects are no longer confused with unresponsive connections. [chan_respoke]: https://github.com/respoke/chan_respoke Review: https://reviewboard.asterisk.org/r/4431/ ........ Merged revisions 432236 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432237 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-24 16:14 +0000 [978649a568] Matt Jordan * channels/chan_sip: Fix crash when transmitting packet after thread shutdown When the monitor thread is stopped, its pthread ID is set to a specific value (AST_PTHREADT_STOP) so that later portions of the code can determine whether or not it is safe to manipulate the thread. Unfortunately, __sip_reliable_xmit failed to check for that value, checking instead only for AST_PTHREAD_STOP. Passing the invalid yet very specific value to pthread_kill causes a crash. This patch adds a check for AST_PTHREADT_STOP in __sip_reliable_xmit such that it doesn't attempt to poke the thread if the thread has already been stopped. ASTERISK-24800 #close Reported by: JoshE ........ Merged revisions 432198 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432199 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-24 15:58 +0000 [3d1a1533bf] Matt Jordan * ARI/PJSIP: Apply requesting channel's format cap to created channels This patch addresses the following problems: * ari/resource_channels: In ARI, we currently create a format capability structure of SLIN and apply it to the new channel being created. This was originally done when the PBX core was used to create the channel, as there was a condition where a newly created channel could be created without any formats. Unfortunately, now that the Dial API is being used, this has two drawbacks: (a) SLIN, while it will ensure audio will flows, can cause a lot of needless transcodings to occur, particularly when a Local channel is created to the dialplan. When no format capabilities are available, the Dial API handles this better by handing all audio formats to the requsted channels. As such, we defer to that API to provide the format capabilities. (b) If a channel (requester) is causing this channel to be created, we currently don't use its format capabilities as we are passing in our own. However, the Dial API will use the requester channel's formats if none are passed into it, and the requester channel exists and has format capabilities. This is the "best" scenario, as it is the most likely to create a media path that minimizes transcoding. Fixing this simply entails removing the providing of the format capabilities structure to the Dial API. * chan_pjsip: Rather than blindly picking the first format in the format capability structure - which actually *can* be a video or text format - we select an audio format, and only pick the first format if that fails. That minimizes the weird scenario where we attempt to transcode between video/audio. * res_pjsip_sdp_rtp: Applied the joint capapbilites to the format structure. Since ast_request already limits us down to one format capability once the format capabilities are passed along, there's no reason to squelch it here. * channel: Fixed a comment. The reason we have to minimize our requested format capabilities down to a single format is due to Asterisk's inability to convey the format to be used back "up" a channel chain. Consider the following: PJSIP/A => L;1 <=> L;2 => PJSIP/B g,u,a g,u,a g,u,a u That is, we have PJSIP/A dialing a Local channel, where the Local;2 dials PJSIP/B. PJSIP/A has native format capabilities g722,ulaw,alaw; the Local channel has inherited those format capabilities down the line; PJSIP/B supports only ulaw. According to these format capabilities, ulaw is acceptable and should be selected across all the channels, and no transcoding should occur. However, there is no way to convey this: when L;2 and PJSIP/B are put into a bridge, we will select ulaw, but that is not conveyed to PJSIP/A and L;1. Thus, we end up with: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B g g X u u Which causes g722 to be written to PJSIP/B. Even if we can convey the 'ulaw' choice back up the chain (which through some severe hacking in Local channels was accomplished), such that the chain looks like: PJSIP/A <=> L;1 <=> L;2 <=> PJSIP/B u u u u We have no way to tell PJSIP/A's *channel driver* to Answer in the SDP back with only 'ulaw'. This results in all the channel structures being set up correctly, but PJSIP/A *still* sending g722 and causing the chain to fall apart. There's a lot of difficulty just in setting this up, as there are numerous race conditions in the act of bridging, and no clean mechanism to pass the selected format backwards down an established channel chain. As such, the best that can be done at this point in time is clarifying the comment. Review: https://reviewboard.asterisk.org/r/4434/ ASTERISK-24812 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432195 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-24 12:32 +0000 [5b73246a9d] Kevin Harwell * bridge_softmix: G.729 codec license held When more than one call using the same codec type enters into a softmix bridge and no audio is present for a channel the bridge optimizes the out frame by using the same one for all channels with the same codec type. Unfortunately, when that number (channels with same codec type) dropped to <= 1 the codec was not dereferenced. At least not until all parties left the bridge. Thus in the case of G.729 the license was not released. This patch ensures that the codec is dereferenced immediately when the optimization no longer applies. ASTERISK-24797 #close Reported by: Luke Hulsey Review: https://reviewboard.asterisk.org/r/4429/ ........ Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432175 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-21 14:47 +0000 [f726304283] Joshua Colp * res_ari_channels: Return a 404 response when a requested channel variable does not exist. This change makes it so that if a channel variable is requested and it does not exist a 404 response will be returned instead of an allocation failed response. This makes it easier to debug and figure out what is going on for a user. ASTERISK-24677 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432154 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-21 13:26 +0000 [7a507ae31a] Joshua Colp * res_pjsip_registrar: Add Expires header to 200 OK if present in REGISTER. Some implementations don't pay attention to the expires for individual contacts. In this case they may consider the lack of an Expires header in the 200 OK as unregistered. This change makes it so if an Expires header is present in the REGISTER we will add one in the 200 OK. ASTERISK-24785 #close Reported by: Ross Beer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432136 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-21 12:51 +0000 [f0d018e249] Joshua Colp * res_pjsip: Add a log message when creating a UAC dialog to a target URI that is invalid. ASTERISK-24499 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432118 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-21 11:35 +0000 [c40d78c31e] Graham Barnett (License 6685) * apps/app_voicemail: Demote an ERROR message to a WARNING message When using IMAP voicemail with FreePBX, you will often get ERROR messages complaining about not being able to find a mailbox. This is due to how FreePBX handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this a configuration error, as in any other system it would be indicative of someone misconfiguring their system. Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch demotes the message so that system administrators can hopefully reduce some of the noise in their log files. Note that in the original patch this was made into a NOTICE, but that's a too forgiving. ASTERISK-24790 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) ........ Merged revisions 432098 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432099 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-21 08:05 +0000 [bf9d416536] Joshua Colp * http: Add missing html tag to 'httpstatus' functionality. ASTERISK-24724 #close Reported by: Ashley Sanders ........ Merged revisions 432078 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432079 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-20 20:56 +0000 [93c9c3af2f] Corey Farrell * Allow shutdown to unload modules that register bucket scheme's or codec's. * Change __ast_module_shutdown_ref to be NULL safe (11+). * Allow modules that call ast_bucket_scheme_register or ast_codec_register to be unloaded during graceful shutdown only (13+ only). ASTERISK-24796 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4428/ ........ Merged revisions 432058 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432059 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-20 20:46 +0000 [54a699fb64] Corey Farrell * asterisk/lock.h: Fix syntax errors for non-gcc OSX with 64-bit integers. Add a couple of missing closing brackets / parenthesis. ASTERISK-24814 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4436/ ........ Merged revisions 432054 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432055 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-20 11:51 +0000 [89b48af3e5] Richard Mudgett * chan_dahdi/sig_analog: Put log message strings on one line. With the log messages on one line, you can search for the log message seen in the log and expect to find it. ........ Merged revisions 432032 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432034 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-20 11:46 +0000 [8e806f9e12] Matt Hoskins (license 6688) * ASTERISK-24811: Add ast_sorcery_apply_config() to res_pjsip_publish_asterisk. Matt Hoskins reported that res_pjsip_publish_asterisk wouldn't pull config from realtime. Turns out it was just missing a call ast_sorcery_apply_config(). res_pjsip_acl was missing it as well, so I added it. The other pjsip modules looked OK. ASTERISK-24811 #close Reported-by: Matt Hoskins Tested-by: George Joseph Tested-by: Matt Hoskins patches: res_pjsip_publish_asterisk.c.patch submitted by Matt Hoskins (license 6688) Review: https://reviewboard.asterisk.org/r/4433/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432033 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-20 09:47 +0000 [c7bdf62a95] Graham Barnett (License 6685) * apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange When interfacing with Microsoft Exchange, custom headers will be returned as all lower case. Currently, the IMAP header code will fail to parse the returned custom headers, as it will be performing a case sensitive comparison. This can cause playback of messages to fail, as needed information - such as origtime - will not be present. This patch updates app_voicemail's header parsing code to perform a case insensitive lookup for the requested custom headers. Since the headers are specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be unique in an IMAP message, this should cause no issues with other systems. ASTERISK-24787 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685) ........ Merged revisions 432012 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@432013 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-19 15:25 +0000 [e0ff83c272] Richard Mudgett * chan_dahdi: Remove some dead code. ........ Merged revisions 431992 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431993 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-19 12:25 +0000 [40547e7210] Richard Mudgett * ISDN AOC: Fix crash from an AOC-E message that doesn't have a channel association. Processing an AOC-E event that does not or no longer has a channel association causes a crash. The problem with posting AOC events to the channel topic is that AOC-E events don't always have a channel association and posting the event to the all channels topic is just wrong. AOC-E events do however have their own charging association method to refer to the agreement with the charging entity. * Changed the AOC events to post to the AMI manager topic instead of the channel topics. If a channel is associated with the event then channel snapshot information is supplied with the AMI event. * Eliminated RAII_VAR() usage in aoc_to_ami() and ast_aoc_manager_event(). This patch supercedes the patch on Review: https://reviewboard.asterisk.org/r/4427/ ASTERISK-22670 #close Reported by: klaus3000 ASTERISK-24689 #close Reported by: Marcel Manz ASTERISK-24740 #close Reported by: Panos Gkikakis Review: https://reviewboard.asterisk.org/r/4430/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431974 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-19 11:30 +0000 [2181c9443f] Richard Mudgett * res_pjsip_refer: Handle INVITE with Replaces failure after answer. * Fixed hangup handling of the session->channel after answer if the ast_channel_move() or ast_bridge_impart() fails. We are still the thread controlling the session->channel so we need to call ast_hangup() to kill the channel. * Fixed debug messages in refer_incoming_invite_request() referencing incorrect channnels on success. Code comments now say why the session->channel cannot be used. Review: https://reviewboard.asterisk.org/r/4422/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431956 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-19 09:28 +0000 [374013d817] Alexander Traud (License 6520) * tcptls: Handle new OpenSSL compile time option to disable SSLv3 Some distributions are going to disable SSLv3 at compile time. This option can be checked using the directive OPENSSL_NO_SSL3_METHOD. This patch updates the TCP/TLS handling in Asterisk to look for that directive before attempting to use the SSLv3 specific methods. ASTERISK-24799 #close Reported by: Alexander Traud patches: no-ssl3-method.patch uploaded by Alexander Traud (License 6520) ........ Merged revisions 431936 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431937 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-18 20:01 +0000 [eb9448a1ae] Corey Farrell * Create work around for scheduler leaks during shutdown. * Added ast_sched_clean_by_callback for cleanup of scheduled events that have not yet fired. * Run all pending peercnt_remove_cb and replace_callno events in chan_iax2. Cleanup of replace_callno events is only run 11, since it no longer releases any references or allocations in 13+. ASTERISK-24451 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4425/ ........ Merged revisions 431916 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431917 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-17 09:31 +0000 [6d3fcfc3c2] Richard Mudgett * res_pjsip_refer: Fix crash from a REFER and BYE collision. Analyzing a one-off crash on a busy system showed that processing a REFER request had a NULL session channel pointer. The only way I can think of that could cause this is if an outgoing BYE transaction overlapped the incoming REFER transaction in a collision. Asterisk sends a BYE while the phone sends a REFER to complete an attended transfer. * Made check the session channel pointer before processing an incoming REFER request in res_pjsip_refer. * Fixed similar crash potential for res_pjsip supplement incoming request processing for res_pjsip_sdp_rtp INFO, res_pjsip_caller_id INVITE/UPDATE, res_pjsip_messaging MESSAGE, and res_pjsip_send_to_voicemail REFER messages. * Made res_pjsip_messaging respond to a message body too large with a 413 instead of ignoring it. ASTERISK-24700 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4417/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431898 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-16 15:29 +0000 [562b7bf6f0] Matt Jordan * res/res_rtp_asterisk: Fix crash in debug from RTCP reports without report block When RTCP debugging was enabled, an RTCP report without a report block would cause a crash. This was due to the verbose output not checking to see if the report_block pointer was NULl before dereferencing it. This patch adds the necessary check to prevent printing any verbose output if the far side hasn't provided us the information they should have. ASTERISK-24791 #close Reported by: JoshE Tested by: JoshE git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431879 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-15 13:00 +0000 [7890d0ad07] Joshua Colp * pjsip: Remove "contact" type from pjsip.conf.sample The "contact" object is not meant to be configured from the pjsip.conf configuration file. It is meant to be created as a result of a registration and stored elsewhere. ASTERISK-24085 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431860 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-15 11:59 +0000 [cbe63ab283] Joshua Colp * install_prereq: Tweak flags when configuring pjproject. This change does two things: 1. Disables debugging so assertions which can return an error do, instead of asserting. 2. Enables IPv6 support. ASTERISK-24632 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431843 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-15 11:42 +0000 [c8f3074cc4] Joshua Colp * res_sorcery_config: Improve object lookup times. The res_sorcery_config module currently uses a fixed bucket size of 53. This means that depending on the number of objects you either end up with excess buckets or a lot of collisions. Due to the way that res_sorcery_config is implemented it's actually possible to make the bucket size dynamic based on the number of objects. This is due to the fact that each loading of the config file produces a new container and does not modify the existing one. This change uses the number of expected objects and finds a prime number near it. In practice depending on the number of objects this can speed up lookups anywhere from 2X to 15X. This change also removes the lock from the container as it is not needed. Review: https://reviewboard.asterisk.org/r/4423/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431841 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-15 10:00 +0000 [a3044cbf02] Joshua Colp * res_pjsip: Add "pjsip show version" CLI command. When debugging things it can be useful to know absolutely what version of pjproject res_pjsip is running against. This change adds a "pjsip show version" CLI command which can be used to query for this. ASTERISK-24685 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-15 06:39 +0000 [ce70587ba6] Matthias Urlichs (license 5508) * res_timing_pthread: Fix leaky pipes. During some refactoring the way private information for timers was stored was changed. As a result of this the action which normally removed the timer upon closure in res_timing_pthread was also removed causing the timer to remain after it should using up resources. This change ensures that the timer is removed upon closure. ASTERISK-24768 #close Reported by: Matthias Urlichs patches: timer.patch submitted by Matthias Urlichs (license 5508) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431807 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-14 18:32 +0000 [4f4d03fdd1] Matt Jordan * apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes The Test Event for MIXMONITOR_END - which signals that a MixMonitor has completed - technically fired before the filestream was closed. If a test used this to trigger a condition to verify that the file was written, it could result in a race condition where the file size would not be what the test expected. Luckily, no tests were using this (although they should have been). Since the test event needed to be moved after the point where the MixMonitor autochan has been destroyed, the test event no longer emits the channel name. Luckily, nothing needs it. ........ Merged revisions 431788 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431789 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-14 13:45 +0000 [758a897876] Joshua Colp * sorcery: Output an error message if a wizard is specified for an object type and it isn't found. ASTERISK-24612 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431771 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-14 12:30 +0000 [8c6e3ad3b4] Joshua Colp * res_pjsip_exten_state: Improve log message when a subscription is attempted to a non-existent extension. ASTERISK-24716 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431754 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-14 12:20 +0000 [3543a36362] Joshua Colp * 'information' ends with an 'n'. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431752 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-14 12:19 +0000 [5d26236758] Joshua Colp * chan_pjsip: Fix crash when CHANNEL dialplan function is invoked with pjsip argument and no type. ASTERISK-24771 #close Reported by: Niklas Larsson git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431751 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-13 11:21 +0000 [4d797f17c5] Richard Mudgett * res_pjsip_session: Fix double re-INVITE collision crash. A multi-asterisk box setup with direct media enabled would occasionally crash when two re-INVITE collisions on a call leg happen in a row. The re-INVITE logic only had one timer struct to defer the re-INVITE. When the second collision happens the timer struct is overwritten and put into the timer heap again. Resources for the first timer are leaked and the heap has two positions occupied by the same timer struct. Now the heap ordering is potentially corrupted, the timer will fire twice, and any resources allocated for the second timer will be released twice. * The solution is to put the collided re-INVITE into the delayed requests queue with all the other delayed requests and cherry pick the next request that can come off the queue when an event happens. * Changed to put delayed BYE requests at the head of the delayed queue. There is no sense in processing delayed UPDATEs and re-INVITEs when a BYE has been requested. * Made the start of a BYE request flush the delayed requests queue to prevent a delayed request from overlapping the BYE transaction. I saw a few cases where a delayed re-INVITE got started after the BYE transaction started. * Changed the delayed_request struct to use an enum instead of a string for the request method. Cherry picking the queue is easier with an enum than string comparisons and the compiler can warn if a switch statement does not cover all defined enum values. * Improved the debug output to give more information. It helps to know which channel is involved with an endpoint. Trunks can have many channels associated with the endpoint at the same time. ASTERISK-24727 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4414/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431734 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-12 14:32 +0000 [1995baad71] Matt Jordan * ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 12:02 +0000 [e8ec15a9ef] Kevin Harwell * res_pjsip: dtls_handler causes Asterisk to crash There have been a couple of times where a crash occurred in the dtls_handler section of the code for res_pjsip. Unfortunately, in working this issue the problem was unable to be reproduced. After looking at the backtraces and through the code the current best guess as to why this happened might be due to a reentrance problem and the strtok function. So, the current fix is to convert the strtok function into the reentrant version of the function, strtok_r. ASTERISK-24741 #close Reported by: Zane Conkle Review: https://reviewboard.asterisk.org/r/4409/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431698 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 11:36 +0000 [e64d151fae] Kevin Harwell * ari_websockets: removed extra check on websocket session read When merging the websocket timeout issue (ASTERISK-24701) an extra, almost duplicate, check was left in the code that should not have been. This removes it. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431693 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 11:28 +0000 [feddab7944] Richard Mudgett * HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 11:12 +0000 [29f3ff0b61] Richard Miller (License 5685) * channels/chan_sip: Fix RealTime error during SIP unregistration with MariaDB When a SIP device that has its registration stored in RealTime unregisters, the entry for that device is updated with blank values, i.e., "", indicating that it is no longer registered. Unfortunately, one of those values that is 'blanked' is the device's port. If the column type for the port is not a string datatype (the recommended type is integer), an ODBC or database error will be thrown. MariaDB does not coerce empty strings to a valid integer value. This patch updates the query run from chan_sip such that it replaces the port value with a value of '0', as opposed to a blank value. This is the value that other database backends coerce the empty string ("") to already, and the handling of reading a RealTime registration value from a backend already anticipates receiving a port of '0' from the backends. ASTERISK-24772 #close Reported by: Richard Miller patches: chan_sip.diff uploaded by Richard Miller (License 5685) ........ Merged revisions 431673 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431674 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 10:51 +0000 [72e5ba2ce8] Kevin Harwell * res_http_websocket: websocket write timeout fails to fully disconnect When writing to a websocket if a timeout occurred the underlying socket did not get closed/disconnected. This patch makes sure the websocket gets disconnected on a write timeout. Also a notice is logged stating that the websocket was disconnected. ASTERISK-24701 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4412/ ........ Merged revisions 431669 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-11 09:51 +0000 [2531f75057] Corey Farrell * Enable REF_DEBUG for ast_module_ref / ast_module_unref. Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431663 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-10 17:16 +0000 [4d8ab20a8a] gtjoseph * res_pjsip_config_wizard: Add ability to auto-create hints. Looking at the Super Awesome Company sample reminded me that creating hints is just plain gruntwork. So you can now have the pjsip conifg wizard auto-create them for you. Specifying 'hint_exten' in the wizard will create 'exten => ,hint/PJSIP/' in whatever is specified for 'hint_context'. Specifying 'hint_application' in the wizard will create 'exten => ,1,' in whatever is specified for 'hint_context'. The default for 'hint_context' is the endpoint's context. There's no default for 'hint_application'. If not specified, no app is added. There's no default for 'hint_exten'. If not specified, neither the hint itself nor the application will be created. Some may think this is the slippery slope to users.conf but hints are a basic necessity for phones unlike voicemail, manager, etc that users.conf creates. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4383/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431643 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-08 21:10 +0000 [32e42e50cc] Ben Merrills (License 6678) * res/ari/resource_channels: Add missing 'no_answer' reason to DELETE /channels One of the canonical reasons for hanging up a channel is because the far end failed to answer - or because someone else answered, and we want to get rid of this channel. This patch adds the missing value to the 'reason' query parameter for the DELETE /channels operation. Review: https://reviewboard.asterisk.org/r/4400 ASTERISK-24745 #close Reported by: Ben Merrills patches: add_no_answer_ari_hangup_cause.diff uploaded by Ben Merrills (License 6678) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431622 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-08 21:01 +0000 [03445a147e] Jeremiah Gowdy (License 6358) * Blocked revisions 431620 While it may not be obvious, r431620 should not occur in Asterisk 13. * We no longer set the SIP_DEFER_BYE_ON_TRANSFER flag during the handling of the INVITE with Replaces. This is now set and handled explicitly in the attended transfer and blind transfer code. * An INVITE with Replaces replacing a channel in a Bridge will now safely eject the channel being replaced. No masquerade occurs. * An INVITE with Replaces replacing a channel not in a Bridge will masquerade, but will do so in such a fashion that we can ensure that we are hanging up the channel when completed. Since the code the patch fixes no longer exists due to core framework changes, we should send a BYE naturally without the need for the flag. ........ channels/chan_sip: Ensure that a BYE is sent during INVITE w/Replaces transfer Consider a scenario where Alice and Bob have an established dialog with each other external to Asterisk. Bob decides to perform an attended transfer of Alice to Asterisk. In this case, Alice will send an INVITE with Replaces to Asterisk, where the Replaces specifies Bob's dialog with Asterisk. In this particular scenario, Asterisk will complete the transfer, but - since Bob's channel has had Alice masqueraded into it and is now a Zombie - a BYE request will not be sent. This patch fixes that issue by adding a new flag to chan_sip that tracks whether or not we have an INVITE with Replaces. If we do, the flag is used on the sip_pvt to ensure that a BYE request is sent, even if the channel has been masqueraded away. Review: https://reviewboard.asterisk.org/r/4362/ ASTERISK-22436 #close Reported by: Eelco Brolman Tested by: Jeremiah Gowdy, Kristian Høgh patches: asterisk-11-hangup-replaced-3.diff uploaded by Jeremiah Gowdy (License 6358) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-08 20:34 +0000 [8582411344] ibercom (License 6599) * res/res_odbc: Remove unneeded queries when determining if a table exists This patch modifies the ast_odbc_find_table function such that it only performs a lookup of the requested table if the table is not already known. Prior to this patch, a queries would be executed against the database even if the table was already known and cached. Review: https://reviewboard.asterisk.org/r/4405/ ASTERISK-24742 #close Reported by: ibercom patches: patch.diff uploaded by ibercom (License 6599) ........ Merged revisions 431617 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431618 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-08 11:24 +0000 [675b2b8103] Matt Jordan * res/res_pjsip_sdp_rtp: Fix leak of local ICE candidates when applying to SDP When an SDP is created for an outgoing request/response, the ICE candidates obtained from the RTP instance are currently leaked. This causes the ao2 container that holds the candidates to never properly be reclaimed when the RTP instance is destroyed. This patch properly decrements the ICE candidates' container if it is successfully obtained. ASTERISK-24769 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431600 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-06 15:26 +0000 [323c0927ac] Scott Griepentrog * various: cleanup issues found during leak hunt In this collection of small patches to prevent Valgrind errors are: fixes for reference leaks in config hooks, evaluating a parameter beyond bounds, and accessing a structure after a lock where it could have been already free'd. Review: https://reviewboard.asterisk.org/r/4407/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431583 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-03 19:27 +0000 [18c8c1bae3] Joshua Colp * res_pjsip_keepalive: Don't crash if PJSIP module is not loaded. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431555 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-02-03 18:58 +0000 [2f2eb1931a] Joshua Colp * sorcery: Don't try to load object types which haven't been defined. The act of defining wizards for an object type in sorcery.conf will create a minimal object type. This can cause a problem when a module has multiple sorcery instances (which all get the wizards from sorcery.conf applied) but the sorcery instances do not all contain full information about the object types. Upon loading errors will occur stating that the objects can not be created. This is confusing and is actually perfectly fine. This change makes it so that only object types which have been fully defined will be loaded. ASTERISK-24748 #close Reported by: Joshua Colp git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431538 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-31 10:27 +0000 [f67402a52a] Joshua Colp * res_format_attr_h264: Fix crash when determining joint capability. The res_format_attr_h264 module currently incorrectly attempts to copy SPS and PPS information from the wrong attribute. This change fixes that. ASTERISK-24616 #close Reported by: Yura Kocyuba Review: https://reviewboard.asterisk.org/r/4392/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-30 11:44 +0000 [05e2832b35] Richard Mudgett * app_agent_pool: Fix initial module load agent device state reporting. When the app_agent_pool module initially loads there is a race condition between the thread loading agents.conf and the device state internal processing thread. If the device state internal processing thread handles the agent creation state updates before the thread that loaded agents.conf registers the device state provider callback then the cached agent state is "Invalid". When a consumer module like app_queue asks for the agent state it gets the cached "Invalid" state instead of the real state from the provider. * Moved loading the agents.conf configuration to the last thing setup by app_agent_pool in load_module(). Now the device state provider callback is registered before the config is loaded so the agent creation state updates are guaranteed to get the initial device state. * Removed some now redundant config cleanup on error in load_config(). * Added lock protection when accessing the device state in agent_pvt_devstate_get() and eliminated the RAII_VAR() usage. ASTERISK-24737 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4390/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431492 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-30 11:38 +0000 [6583b4de98] Kevin Harwell * res_pjsip_outbound_publish: eventually crashes when no response is ever received When Asterisk attempts to send SIP outbound publish information and no response is ever received (no 200 okay, 412, 423) the system eventually crashes. A response is never received because the system Asterisk is attempting to send publish information to is not available. The underlying pjsip framework attempts to send publish information. After several attempts it calls back into the Asterisk outbound publish code. At this point if the "client->queue" is empty Asterisk attempts to schedule a refresh which utilizes "rdata" and since no response was received the given "rdata" struture is NULL. Attempting to dereference a NULL object of course results in a crash. The fix here removes the dependency on rdata for schedule_publish_refresh. Instead param->expiration is now passed to it as this is set to -1 if no response is received. Also added a notification when no response is received. ASTERISK-24635 #close Reported by: Marco Paland Review: https://reviewboard.asterisk.org/r/4384/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-30 10:52 +0000 [112d23c73e] Ashley Sanders * HTTP: For httpd server, need option to define server name for security purposes Added a new config property [servername] to the http.conf file; updated the http server to use the new property when sending responses, for showing http status through the CLI and when reporting status through the 'httpstatus' webpage. ASTERISK-24316 #close Reported By: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431471 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-30 10:47 +0000 [43dd42d8ae] Mark Michelson * Fix some memory leaks. These memory leaks were found and fixed by John Hardin. I'm just committing them for him. ASTERISK-24736 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/4389 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431468 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-29 17:02 +0000 [f7d23dfcc6] Scott Griepentrog * stasis transfer: fix stasis bridge push race part two When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431450 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-29 14:58 +0000 [e8896ac008] Mark Michelson * Use SIPS URIs in Contact headers when appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in res_pjsip/chan_pjsip.c Review: https://reviewboard.asterisk.org/r/4345 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431426 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-29 14:44 +0000 [22fc3359da] Mark Michelson * Use SIPS URIs in Contact headers when appropriate. RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific scenarios when we are required to use SIPS URIs in Contact headers. Asterisk's non-compliance with this could actually cause calls to get dropped when communicating with clients that are strict about checking the Contact header. Both of the SIP stacks in Asterisk suffered from this issue. This changeset corrects the behavior in chan_sip. ASTERISK-24646 #close Reported by Stephan Eisvogel Review: https://reviewboard.asterisk.org/r/4346 ........ Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 2015-01-29 13:52 +0000 [b8ea23b0d1] Mark Michelson * Allow disabling of 100rel support on PJSIP endpoints. Due to an inversion error, setting 100rel=no would not actually change the current value of the setting (which defaulted to "yes"). With this fix, the inversion is cor