32989 Commits (93870e7bb4d5ef2cf5f9f06c110d7d45221cf067)

 

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Author	SHA1	Message	Date
Alexander Traud	620d9f4782	chan_sip: Set up calls without audio (text+video), again. ... The previous commit 6d980de fixed this issue in the core of Asterisk. With that, each channel technology can be used without audio theoretically. Practically, the channel-technology driver chan_sip turned out to have an invalid check preventing that. chan_sip tested whether there is at least one audio format. However, chan_sip has to test whether there is at least one format. More cannot be tested while requesting chan_sip because only the [general] capabilities but not the [peer] caps are known yet. And the [peer] caps might not be a subset or show any intersection with the [general] caps. This change here fixes this. The original commit f04d5fb, thirteen years ago, contained a software bug as it passed ANY audio capability to the channel-technology driver. Instead, it should have passed NO audio format. Therefore, this addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. Then, Asterisk 10 changed that from ANY to NO, but nobody reported since then. ASTERISK-29265 Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f	4 years ago
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	4 years ago
Alexander Traud	6d980de282	channel: Set up calls without audio (text+video), again. ... ASTERISK-29259 Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5	4 years ago
roadkill	9b5d20e3d5	res/res_pjsip.c: allow user=phone when number contain *# ... if From number contain * or # asterisk will not add user=phone Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY this is a problem when you want to send call to ISUP as they will disregard the From header and either replace From with anonymous or with p-asserted-identity ASTERISK-29261 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4	4 years ago
Alexander Traud	4aff42b274	chan_sip: SDP: Reject audio streams correctly. ... This completes the fix for ASTERISK_24543. Only when the call is an outgoing call, consult and append the configured format capabilities (p->caps). When all audio formats got rejected the negotiated format capabilities (p->jointcaps) contain no audio formats for incoming calls. This is required when there are other accepted media streams. ASTERISK-29258 Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9	4 years ago
Ivan Poddubnyi	05472da92b	main/frame: Add missing control frame names to ast_frame_subclass2str ... Log proper control frame names instead of "Unknown control '14'", etc. Change-Id: I1724f2f4d1b064b25a5c93a7da0cb03be5143935	4 years ago
Boris P. Korzun	92f5cf7f2d	res_musiconhold: Add support of various URL-schemes by MoH. ... Provided a support of variuos URL-schemes for res_musiconhold, registered by ast_bucket_scheme_register(). ASTERISK-29262 #close Change-Id: If0ea8697587353dce358a70035d82649fd4632b6	4 years ago
Jaco Kroon	060ce10163	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>	4 years ago
Alexander Traud	10a0a0c59b	pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. ... Otherwise, Clang 10 warned because of logical-not-parentheses. Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79	4 years ago
Alexander Traud	df6afadf26	res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. ... ASTERISK-29248 Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556	4 years ago
Sean Bright	6d2bec7028	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	4 years ago
Ben Ford	948ceb1228	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	4 years ago
Robert Cripps	24e678b9bb	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	4 years ago
Jean Aunis	c559667868	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	4 years ago
Alexander Traud	1c05667cfc	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	4 years ago
Ivan Poddubnyi	f2aa6c7017	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	4 years ago
Ivan Poddubnyi	134d2e729d	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	4 years ago
Jaco Kroon	2d3441772b	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	4 years ago
George Joseph	9a4486e9fb	Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" ... This reverts commit 2fe76dd816. Reason for revert: Too many issues reported. Need to research and correct. ASTERISK-29230 ASTERISK-29231 Reported by: Michael Maier Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb	4 years ago
Jaco Kroon	c797500956	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>	4 years ago
Jaco Kroon	4e038c1eaa	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>	4 years ago
Kevin Harwell	3bcf483373	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	4 years ago
Sean Bright	44d68bd56b	app_voicemail: Prevent deadlocks when out of ODBC database connections ... ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	4 years ago
Dan Cropp	ffa87ecade	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	4 years ago
Kevin Harwell	4274a4a7dd	pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type ... A prior patch segmented channel snapshots, and changed the underlying data object type associated with ast_channel_snapshot_type stasis messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but now it type ast_channel_snapshot_update. When publishing ast_channel_snapshot_type in pbx_realtime the ast_channel_snapshot was being passed in as the message data object. When a handler, expecting a data object type of ast_channel_snapshot_update, dereferenced this value a crash would occur. This patch makes it so pbx_realtime now uses the expected type, and channel snapshot publish method when publishing. ASTERISK-29168 #close Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896	4 years ago
Sean Bright	1b74555fcf	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	4 years ago
Nick French	505939c9ed	res_pjsip: Prevent segfault in UDP registration with flow transports ... Segfault occurs during outbound UDP registration when all transport states are being iterated over. The transport object in the transport is accessed, but flow transports have a NULL transport object. Modify to not iterate over any flow transport ASTERISK-29210 #close Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a	4 years ago
Alexander Traud	80c14f74bc	codecs: Remove test-law. ... This was dead code, test code introduced with Asterisk 13. This was found while analyzing ASTERISK_28416 and ASTERISK_29185. This change partly fixes, not closes those two issues. Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940	4 years ago
Torrey Searle	51e2187a14	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 (cherry picked from commit a7aea71e60)	4 years ago
Richard Mudgett	058bc0d593	chan_vpb.cc: Fix compile errors. ... Fix the usual compile problem when someone adds a new callback to struct ast_channel_tech. Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32	4 years ago
Richard Mudgett	6d7af72559	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	4 years ago
Sungtae Kim	02c4b2ac60	res_pjsip_session: Fixed NULL active media topology handle ... Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	4 years ago
Sean Bright	357510cec3	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	4 years ago
Sungtae Kim	91fc57f56b	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	4 years ago
George Joseph	7d4ae7dc18	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	4 years ago
Pirmin Walthert	0b10995811	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	4 years ago
Nathan Bruning	5e426987c2	res_musiconhold: Don't crash when real-time doesn't return any entries ... ASTERISK-29211 #close Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877	4 years ago
Joshua C. Colp	9ee1f7154f	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	4 years ago
Joshua C. Colp	6475fe3dd7	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	4 years ago
Sean Bright	90fd1fd96a	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	4 years ago
lvl	b08427134f	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, and changes the default location from /tmp to /var/cache/asterisk. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	4 years ago
Sean Bright	c8b6340023	media_cache: Fix reference leak with bucket file metadata ... Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6	4 years ago
Stanislav	ab7a08b4ef	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	4 years ago
Joshua C. Colp	eda3679c1c	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	4 years ago
Alexander Traud	b91fb3c396	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	4 years ago
Sean Bright	d04b5903d1	CHANGES: Remove already applied CHANGES update ... Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	4 years ago
Alexander Greiner-Baer	fba10fb54c	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	4 years ago
Alexander Traud	103d7da3bb	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	4 years ago
Boris P. Korzun	8cb439f7e4	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	5 years ago
Alexander Traud	7c355d78cb	modules.conf: Align the comments for more conclusiveness. ... Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306	5 years ago