32763 Commits (ff037a135cc7501215268a2da8689541baf4e40c)

 

All Branches   

Search

Author	SHA1	Message	Date
Robert Cripps	ff037a135c	res/res_pjsip_session.c: Check that media type matches in ... function ast_sip_session_media_state_add. Check ast_media_type matches when a ast_sip_session_media is found otherwise when transitioning from say image to audio, the wrong session is returned in the first if statement. ASTERISK-29220 #close Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d	6 years ago
Jaco Kroon	aa03d07a08	AC_HEADER_STDC causes a compile failure with autoconf 2.70 ... From https://www.mail-archive.com/bug-autoconf@gnu.org/msg04408.html > ... the long-obsolete AC_HEADER_STDC, previously used internally by > AC_INCLUDES_DEFAULT, used AC_EGREP_HEADER. The AC_HEADER_STDC macro > is now a no-op (and is not used at all within Autoconf anymore), so > that change is likely what made the first use of AC_EGREP_HEADER the > one inside the if condition, causing the observed results. The implication is that the test does nothing anyway, and due to it being a no-op from 2.70 onwards, results in the required not being set to yes, resulting in ./configure to fail. Change-Id: Ic1ff38d87f791fbf1f2a80512f81bb7110392460 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Alexander Traud	6915106cc1	pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. ... Otherwise, Clang 10 warned because of logical-not-parentheses. Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79	6 years ago
Alexander Traud	3b1cab26a5	res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. ... ASTERISK-29248 Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556	6 years ago
Ben Ford	0520f8320c	chan_pjsip.c: Add parameters to frame in indicate. ... There are a couple of parameters (datalen and data) that do not get set in chan_pjsip_indicate which could cause an Invalid message to pop up for things such as fax. This patch adds them to the frame. Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8	6 years ago
Sean Bright	2c533dc76f	res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet ... The last argument to ast_copy_string() is the buffer size, not the number of characters, so we add 1 to avoid stamping out the final \n in the persisted SUBSCRIBE message. Change-Id: I019b78942836f57965299af15d173911fcead5b2	6 years ago
Asterisk Development Team	6056818467	Update CHANGES and UPGRADE.txt for 16.16.0	6 years ago
Jean Aunis	45e1d89135	Stasis/messaging: tech subscriptions conflict with endpoint subscriptions. ... When both a tech subscription and an endpoint subscription exist for a given endpoint, TextMessageReceived events are dispatched to the tech subscription only. ASTERISK-29229 Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8	6 years ago
Alexander Traud	34b0960310	chan_sip: SDP: Sidestep stream parsing when its media is disabled. ... Previously, chan_sip parsed all known media streams in an SDP offer like video (and text) even when videosupport=no (and textsupport=no). This wasted processor power. Furthermore, chan_sip accepted SDP offers, including no audio but just video (or text) streams although videosupport=no (or textsupport=no). Finally, chan_sip denied the whole offer instead of individual streams when they had encryption (SDES-sRTP) unexpectedly enabled. ASTERISK-29238 ASTERISK-29237 ASTERISK-29222 Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755	6 years ago
Ivan Poddubnyi	97afc9055f	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	6 years ago
Ivan Poddubnyi	17fa9c93d0	chan_pjsip: Stop queueing control frames twice on outgoing channels ... The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. This resulted in extra noise in logs (for example, "is making progress" and "is ringing" get logged twice by app_dial), as well as in noise in signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. This change splits the response handler into 2 functions: - one for updating HANGUPCAUSE, which is still called twice, - another that does the rest, which is called only once as before. ASTERISK-28016 Reported-by: Alex Hermann ASTERISK-28549 Reported-by: Gant Liu ASTERISK-28185 Reported-by: Julien Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940	6 years ago
Jaco Kroon	7df88c98d0	contrib/systemd: Added note on common issues with systemd and asterisk ... With newer version of linux /var/run/ is a symlink to /run/ that has been turned into tmpfs. Added note that if asterisk has to bind to a specific IP that systemd has to wait until the network is up. Added note on how to make sure that the environment variable HOSTNAME is included. ASTERISK-29216 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: Ib3e560655befd3e99eec743687144f5569533379	6 years ago
George Joseph	7ed20b9d3b	Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" ... This reverts commit e1fd51cd2c. Reason for revert: Too many issues reported. Need to research and correct. ASTERISK-29230 ASTERISK-29231 Reported by: Michael Maier Change-Id: Ie2356c1cd74436603600599885286b98055b8a17	6 years ago
Jaco Kroon	aca435dfe7	pbx_lua: Add LUA_VERSIONS environment variable to ./configure. ... On Gentoo it's possible to have multiple lua versions installed, all with a path of /usr, so it's not possible to use the current --with-lua option to determisticly pin to a specific version as is required by the Gentoo PMS standards. This environment variable allows to lock to specific versions, unversioned check will be skipped if this variable is supplied. Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Jaco Kroon	32e36144c7	func_lock: fix multiple-channel-grant problems. ... Under contention it becomes possible that multiple channels will be told they successfully obtained the lock, which is a bug. Please refer ASTERISK-29217 This introduces a couple of changes. 1. Replaces requesters ao2 container with simple counter (we don't really care who is waiting for the lock, only how many). This is updated undex ->mutex to prevent memory access races. 2. Correct semantics for ast_cond_timedwait() as described in pthread_cond_broadcast(3P) is used (multiple threads can be released on a single _signal()). 3. Module unload races are taken care of and memory properly cleaned up. Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Dan Cropp	e127a57761	chan_pjsip: Incorporate channel reference count into transfer_refer(). ... Add channel reference count for PJSIP REFER. The call could be terminated prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY occurred several minutes later, it would attempt to access a session which was no longer valid. Terminate event subscription if pjsip_xfer_initiate() or pjsip_xfer_send_request() fails in transfer_refer(). ASTERISK-29201 #close Reported-by: Dan Cropp Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435	6 years ago
Kevin Harwell	e96f744816	app_mixmonitor: cleanup datastore when monitor thread fails to launch ... launch_monitor_thread is responsible for creating and initializing the mixmonitor, and dependent data structures. There was one off nominal path after the datastore gets created that triggers when the channel being monitored is hung up prior to monitor starting itself. If this happened the monitor thread would not "launch", and the mixmonitor object and associated objects are freed, including the underlying datastore data object. However, the datastore itself was not removed from the channel, so when the channel eventually gets destroyed it tries to access the previously freed datastore data and crashes. This patch removes and frees datastore object itself from the channel before freeing the mixmonitor object thus ensuring the channel does not call it when destroyed. ASTERISK-28947 #close Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8	6 years ago
Sean Bright	2b7af3eb27	app_voicemail: Prevent deadlocks when out of ODBC database connections ... ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	6 years ago
Sean Bright	4c5bffb217	asterisk: Export additional manager functions ... Rename check_manager_enabled() and check_webmanager_enabled() to begin with ast_ so that the symbols are automatically exported by the linker. ASTERISK~29184 Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9	6 years ago
Richard Mudgett	89cf7899be	res_pjsip_session.c: Fix compiler warnings. ... AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an unsigned long on all machines. Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338	6 years ago
Sungtae Kim	ab3f57d88f	res_pjsip_session: Fixed NULL active media topology handle ... Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	6 years ago
Torrey Searle	9196e0d1d5	res/res_pjsip_diversion: prevent crash on tel: uri in History-Info ... Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter. ASTERISK-29191 ASTERISK-29219 Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37	6 years ago
Sean Bright	0a23296834	app_chanspy: Spyee information missing in ChanSpyStop AMI Event ... The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	6 years ago
Sungtae Kim	a47e6965b3	res_ari: Fix wrong media uri handle for channel play ... Fixed wrong null object handle in /channels/<channel_id>/play request handler. ASTERISK-29188 Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe	6 years ago
George Joseph	5a2867efa9	logger.c: Automatically add a newline to formats that don't have one ... Scope tracing allows you to not specify a format string or variable, in which case it just prints the indent, file, function, and line number. The trace output automatically adds a newline to the end in this case. If you also have debugging turned on for the module, a debug message is also printed but the standard log functionality which prints it doesn't add the newline so you have messages that don't break correctly. * format_log_message_ap(), which is the common log message formatter for all channels, now adds a newline to the end of format strings that don't already have a newline. ASTERISK-29209 Reported by: Alexander Traud Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da	6 years ago
Pirmin Walthert	11def974a8	res_pjsip_nat.c: Create deep copies of strings when appropriate ... In rewrite_uri asterisk was not making deep copies of strings when changing the uri. This was in some cases causing garbage in the route header and in other cases even crashing asterisk when receiving a message with a record-route header set. Thanks to Ralf Kubis for pointing out why this happens. A similar problem was found in res_pjsip_transport_websocket.c. Pjproject needs as well to be patched to avoid garbage in CANCEL messages. ASTERISK-29024 #close Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b	6 years ago
Joshua C. Colp	7e4bb4ed11	res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent. ... This adds support for both Digium and Sangoma user agent strings for the Sangoma specific body supplement. Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482	6 years ago
Nathan Bruning	bb46595799	res_musiconhold: Don't crash when real-time doesn't return any entries ... ASTERISK-29211 #close Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877	6 years ago
lvl	8d2558209b	Introduce astcachedir, to be used for temporary bucket files ... As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, but leaves it at /tmp by default, to ensure backwards compatibility. A future commit that only targets master could change the default location to something more sensible such as /var/tmp/asterisk. At that point, the cachedir could be created and cleaned up during uninstall by the Makefile script. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	6 years ago
Joshua C. Colp	ea744ca7c2	pjsip: Match lifetime of INVITE session to our session. ... In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	6 years ago
Sean Bright	0c185c9e21	res_http_media_cache.c: Set reasonable number of redirects ... By default libcurl does not follow redirects, so we explicitly enable it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl will follow up to CURLOPT_MAXREDIRS redirects, which by default is configured to be unlimited. This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If we determine at some point that this needs to be increased on configurable it is a trivial change. ASTERISK-29173 #close Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30	6 years ago
Sean Bright	ddbf3a7f73	media_cache: Fix reference leak with bucket file metadata ... Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6	6 years ago
Stanislav	159522003a	res_pjsip_stir_shaken: Fix module description ... the 'J' is missing in module description. "PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk" ASTERISK-29175 #close Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a	6 years ago
Joshua C. Colp	15566494f9	voicemail: add option 'e' to play greetings as early media ... When using this option, answering the channel is deferred until all prompts/greetings have been played and the caller is about to leave their message. ASTERISK-29118 #close Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb	6 years ago
Alexander Traud	4c79bc19d1	loader: Sync load- and build-time deps. ... In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO. ASTERISK-29148 Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2	6 years ago
Sean Bright	a360150ee0	CHANGES: Remove already applied CHANGES update ... Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	6 years ago
Alexander Traud	f667c5a781	chan_sip: Remove unused sip_socket->port. ... 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port vanished. However, the struct member itself and all seven set/uses remained as dead code. ASTERISK-28798 Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59	6 years ago
Alexander Greiner-Baer	a8f6238cc8	res_pjsip: set Accept-Encoding to identity in OPTIONS response ... RFC 3261 says that the Accept-Encoding header should be present in an options response. Permitted values according to RFC 2616 are only compression algorithms like gzip or the default identity encoding. Therefore "text/plain" is not a correct value here. As long as the header is hard coded, it should be set to "identity". Without this fix an Alcatel OmniPCX periodically logs warnings like "[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed" on a SIP Trunk. ASTERISK-29165 #close Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840	6 years ago
Boris P. Korzun	89d3de37ca	bridge_basic: Fixed setup of recall channels ... Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641. common_recall_channel_setup() setups common things on the recalled transfer target, but used same target as source instead trasfered. ASTERISK-29161 #close Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f	6 years ago
Alexander Traud	d1a78e047d	modules.conf: Align the comments for more conclusiveness. ... Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306	6 years ago
George Joseph	8d8c9db618	app_queue: Fix deadlock between update and show queues ... Operations that update queues when shared_lastcall is set lock the queue in question, then have to lock the queues container to find the other queues with the same member. On the other hand, __queues_show (which is called by both the CLI and AMI) does the reverse. It locks the queues container, then iterates over the queues locking each in turn to display them. This creates a deadlock. * Moved queue print logic from __queues_show to a separate function that can be called for a single queue. * Updated __queues_show so it doesn't need to lock or traverse the queues container to show a single queue. * Updated __queues_show to snap a copy of the queues container and iterate over that instead of locking the queues container and iterating over it while locked. This prevents us from having to hold both the container lock and the queue locks at the same time. This also allows us to sort the queue entries. ASTERISK-29155 Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41	6 years ago
Asterisk Development Team	3366956139	Update CHANGES and UPGRADE.txt for 16.15.0	6 years ago
George Joseph	e1fd51cd2c	res_pjsip_outbound_registration.c: Use our own scheduler and other stuff ... * Instead of using the pjproject timer heap, we now use our own pjsip_scheduler. This allows us to more easily debug and allows us to see times in "pjsip show/list registrations" as well as being able to see the registrations in "pjsip show scheduled_tasks". * Added the last registration time, registration interval, and the next registration time to the CLI output. * Removed calls to pjsip_regc_info() except where absolutely necessary. Most of the calls were just to get the server and client URIs for log messages so we now just save them on the client_state object when we create it. * Added log messages where needed and updated most of the existong ones to include the registration object name at the start of the message. Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2	6 years ago
George Joseph	80f116c156	pjsip_scheduler.c: Add type ONESHOT and enhance cli show command ... * Added a ONESHOT type that never reschedules. * Added "like" capability to "pjsip show scheduled_tasks" so you can do the following: CLI> pjsip show scheduled_tasks like outreg PJSIP Scheduled Tasks: Task Name Interval Times Run ... ============================================= ========= ========= ... pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ... pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ... * Fixed incorrect display of "Next Start". * Compacted the displays of times in the CLI. * Added two new functions (ast_sip_sched_task_get_times2, ast_sip_sched_task_get_times_by_name2) that retrieve the interval, next start time, and next run time in addition to the times already returned by ast_sip_sched_task_get_times(). Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3	6 years ago
Alexei Gradinari	728cd55cde	sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data ... The data can be freed if the old object '_data' is the same object as new 'data'. Because at first the object is unreferenced which can lead to destroying it. This could happened in res_pjsip_pubsub when the publication is updated which could lead to segfault in function publish_expire. Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da	6 years ago
Alexander Traud	ddfb76a864	res_pjsip/config_transport: Load and run without OpenSSL. ... ASTERISK-28933 Reported-by: Walter Doekes Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f	6 years ago
Alexander Traud	277aa0ced6	res_stir_shaken: Include OpenSSL headers where used actually. ... This avoids the inclusion of the OpenSSL headers in the public header, which avoids one external library dependency in res_pjsip_stir_shaken. Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4	6 years ago
Dovid Bender	5046e1fb06	func_curl.c: Allow user to set what return codes constitute a failure. ... Currently any response from res_curl where we get an answer from the web server, regardless of what the response is (404, 403 etc.) Asterisk currently treats it as a success. This patch allows you to set which codes should be considered as a failure by Asterisk. If say we set failurecodes=404,403 then when using curl in realtime if a server gives a 404 error Asterisk will try to failover to the next option set in extconfig.conf ASTERISK-28825 Reported by: Dovid Bender Code by: Gobinda Paul Change-Id: I94443e508343e0a3e535e51ea6e0562767639987	6 years ago
Kevin Harwell	8973fe5cf3	AST-2020-001 - res_pjsip: Return dialog locked and referenced ... pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	6 years ago
Ben Ford	58aa6a7057	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. ... If Asterisk sends out an INVITE and receives a challenge with a different nonce value each time, it will continuously send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	6 years ago