asterisk

Commit Graph

Author	SHA1	Message	Date
sungtae kim	08ba504e99	chan_pjsip: Changed to continued after invalid media for pjsip show channelstats Currently, the pjsip show channelstats cli does not show channel's stats after hits the invalid channel info. This makes hard to use this cli. Changed to keep iterate after hits the invalid channel info. ASTERISK-28292 Change-Id: I3efdff1c9e1b1efd3c971fb82ae77aa133a6f43c	6 years ago
Sungtae Kim	3fe40c0c05	res_pjsip_session Added rtcp stats result vector into the session Currently, the Asterisk's pjsip_session module does not keeping the rtcp's stats info after it was removed. But by adding the results vector and keeping it until session is destroying, it can give more useful information for other modules. ASTERISK-28253 Change-Id: Ib25c2d3fc4da084aecfde2a82c1b1d733bd64fa5	6 years ago
Joshua Colp	3a2db0ee04	CI: Use tmpfs option to Docker instead of mount. Some tests require Asterisk to execute scripts which are stored in /tmp. When mount is used for tmpfs there is no ability to allow scripts to be executed from that location. This change switches to using tmpfs which can be told to allow executables to be run from /tmp. Change-Id: I0e598ca2b76af1f7f2d29f0da7b1731a214a291a	6 years ago
Kevin Harwell	e85f92f37a	json.c/strings.c - Add a couple of utility functions Added 'ast_json_object_string_get' to the JSON wrapper in order to make it a little easier to retrieve a string field from the JSON object. Also added an 'ast_strings_equal' function that safely checks (checks for NULLs) for equality between two strings. Change-Id: I26f0a16d61537505eb41b4b05ef2e6d67fc2541b	6 years ago
Torrey Searle	01b781b4dc	res/res_rtp_asterisk: clear smoother when local bridging p2p_write updates txformat but doesn't require a smoother. If a smoother was created by another bridge type the smoother could fall out of date causing one way audio issues. To prevent this the smoother is now destroyed on the start of native bridge. ASTERISK-28284 #close Change-Id: I84e67f144963787fff9b4d79ac500514fb40cdc6	6 years ago
Chris Savinovich	b327b0bbda	Revert "Test_cel: Fails when DONT_OPTIMIZE is off" This reverts commit `05b79d16ab`. Change-Id: I2e64fc22c33b10cfd592681732fc67ae0669a301	6 years ago
George Joseph	aaee80adfe	Merge "ci: Rerun unit tests when non-code changes occur." into 16	6 years ago
Joshua C. Colp	2070f62bb6	Merge "res_odbc: Add basic query logging." into 16	6 years ago
Joshua Colp	1084b6e1a1	ci: Rerun unit tests when non-code changes occur. This change makes it so that even if non-code changes occur (such as commit message changing) unit tests will still be run and result in a verification. ASTERISK-28251 Change-Id: I6491fff7c93e5d5cd8e41054486968bf66c4f608	6 years ago
Kevin Harwell	2cf3931379	res_pjsip_registrar: lock transport monitor when setting 'removing' flag A previous patch attempt to mitigate blocked threads on transport shutdown for a given contact. It was thought that a second lock could be avoided by checking the 'removing' flag on the transport monitor twice (once before and once after the normal named aor locking). However as with usual threading issues if the timing was right the original problem still occured. This patch adds locking around the first 'removing' flag check and set, thus nullifying the secondary check, so it was removed. ASTERISK-28213 Change-Id: Iaa8e36e5311789549b76d8de42dfcea96013b2ed	6 years ago
Joshua Colp	44254d548b	res_odbc: Add basic query logging. When Asterisk is connected and used with a database the response time of the database can cause problems in Asterisk if it is long. Normally the only way to see this problem would be to retrieve a backtrace from Asterisk and examine where things are blocked, or examine the database to see if there is any indication of a problem. This change adds some basic query logging to make it easier to investigate such a problem. When logging is enabled res_odbc will now keep track of the number of queries executed, as well as the query that has taken the longest time to execute. There is also an option which will cause a WARNING message to be output if a query takes longer than a configurable amount of time to execute. This makes it easier and clearer for users that their database may be experiencing a problem that could impact Asterisk. ASTERISK-28277 Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6	6 years ago
George Joseph	c0b7b60ba1	Merge "sounds: Sort 'core show sounds' output" into 16	6 years ago
George Joseph	7e366e4d4b	Merge "main/cdr: Fixed cdr start overwriting" into 16	6 years ago
Sungtae Kim	e3e3027092	main/cdr: Fixed cdr start overwriting The CDR was overwriting the start time when the call continued the dialplan from the ARI stasis or a Local channel was originated. This change fixes this by no longer reinitializing the CDR when transitioning out of the dialed pending state to the single state. ASTERISK-28181 Change-Id: I921bc04064b6cff1deb2eea56a94d86489561cdc	6 years ago
Giuseppe Sucameli	419db481d1	Fix deadlock handling subscribe req during res_parking reload Split destroy_hint method to separate hint removal and extension hint state changed callback, the latter now called via stasis. This avoids deadlock between res_parking reload that is removing the parking lot and the related hint and subscribe requests coming for the same parking lot. ASTERISK-28173 Change-Id: I5b03c3455b3b12b6f83cea4cc34f4b4b20444f7e	6 years ago
George Joseph	3d54ef5978	Merge "pjsip/config_global: regcontext context not created" into 16	6 years ago
George Joseph	29a77b337d	Merge "Added ARI resource /ari/asterisk/ping" into 16	6 years ago
George Joseph	c925742e8b	Merge "res_stasis: Auto-create context and extens on Stasis app launch." into 16	6 years ago
sungtae kim	19fc99a2fb	Added ARI resource /ari/asterisk/ping Added ARI resource. GET /ari/asterisk/ping : It returns "pong" message with timestamp and asterisk id. It would be useful for simple heath check. Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29	6 years ago
Sean Bright	f29da72969	sounds: Sort 'core show sounds' output Change-Id: Ib39052a745040f75eb635f15a042da15b20e22ab	6 years ago
Joshua C. Colp	ffe6ecb40f	Merge "bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64" into 16	6 years ago
Ben Ford	1051e1dd18	res_stasis: Auto-create context and extens on Stasis app launch. At AstriCon, there was a strong desire for the ability to completely bypass dialplan when using ARI. This is possible through the automatic creation of a context and a couple of extensions whenever an application is started. For example, if you have an application named 'ari-example', a context named 'stasis-ari-example' will be automatically created whenever this application is started as long as one does not already exist. Two extensions (a match-all extension for Stasis and a 'h' extension) are created within this context. Any endpoint that registers to Asterisk within this context will send all calls to the corresponding Stasis application. When the application is destroyed, the context is removed. ASTERISK-28104 #close Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac	6 years ago
Joshua C. Colp	6ccf1155d1	Merge "media_index.c: Refactored so it doesn't cache the index" into 16	6 years ago
George Joseph	70fa6e6955	bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64 On OpenSuse Leap, libjansson.a is installed in third-party/jansson/dest/lib64 instead of lib (which is where the top-level makeopts looks). This causes a link failure. * Updated jansson/Makefile to add an explicit --libdir to force the installation to third-party/jansson/dest/lib. ASTERISK-28271 Reported by: David Wilcox Change-Id: Ibf8af75e5da13562105fcc39ed898c6ef0b5a5f3	6 years ago
Kevin Harwell	9c3b4dcf80	pjsip/config_global: regcontext context not created The context specified by 'regcontext' was not being created, so when Asterisk attempted to later dynamically add an extension it would fail. This patch now creates the context if a 'regcontext' is specified. ASTERISK-28238 Change-Id: I0f36cf4ab0a93ff4b1cc5548d617ecfd45e09265	6 years ago
George Joseph	603143bd5a	media_index.c: Refactored so it doesn't cache the index Testing revealed that the cache added no benefit but that it could consume excessive memory. Two new index related functions were created: ast_sounds_get_index_for_file() and ast_media_index_update_for_file() which restrict index updating to specific sound files. The original ast_sounds_get_index() and ast_media_index_update() calls are still available but since they no longer cache the results internally, developers should re-use an index they may already have instead of calling ast_sounds_get_index() repeatedly. If information for only a single file is needed, ast_sounds_get_index_for_file() should be called instead of ast_sounds_get_index(). The media_index directory scan code was elimininated in favor of using the existing ast_file_read_dirs() function. Since there's no more cache, ast_sounds_index_init now only registers the sounds cli commands instead of generating the initial index and subscribing to stasis format register/unregister messages. "sounds" is no longer a valid target for the "module reload" command. Both the sounds cli commands and the sounds ari resources were refactored to only call ast_sounds_get_index() once per invocation and to use ast_sounds_get_index_for_file() when a specific sound file is requested. Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d	6 years ago
George Joseph	36a10a3672	Merge "codecs.conf.sample: update codec opus docs" into 16	6 years ago
George Joseph	d0087a8107	Merge "format_g726: add support for seeking" into 16	6 years ago
George Joseph	47a2b6d0f3	Merge "res_http_websocket: ensure control frames do not interfere with data" into 16	6 years ago
Kevin Harwell	f6452f9656	codecs.conf.sample: update codec opus docs The option value "sdp" for some of the settings was removed a while back, however the sample conf was not updated. This patch removes any wording with regards to the old "sdp" option value, and adjusts the defaults to what they are now. ASTERISK-28263 Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445	6 years ago
Joshua C. Colp	e994dff064	Merge "build : Fix cross-compilation errors" into 16	6 years ago
eyalhasson	c1da2e94a3	format_g726: add support for seeking Added support for the seek function in format_g726 so playback can start from anywhere. Before the fix, playback of g726 files always started from the beginning. ASTERISK-28246 Change-Id: I626235bc4642df1479050d3d06828412603a9b40	6 years ago
Joshua C. Colp	fe4c6434c1	Merge "app_voicemail: Add Mailbox Aliases" into 16	6 years ago
Joshua C. Colp	e2fb30245b	Merge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown" into 16	6 years ago
Joshua C. Colp	a40fb7fa7f	Merge "Test_cel: Fails when DONT_OPTIMIZE is off" into 16	6 years ago
Jean Aunis	d3a6714158	build : Fix cross-compilation errors Bundled pjproject and jansson must be configured with the host and build parameters provided to the configure script. Autotools do not permit to check for the existence of local header files, so the control of hrirs.h must not be done when cross-compiling. ASTERISK-28250 Change-Id: If0a76e52a87d4ab82b7d4c72d27d8759ca931880	6 years ago
Joshua C. Colp	6640ecfd39	Merge "manager_channels: Fix throwing of HangupHandler manager events" into 16	6 years ago
Jeremy Lainé	907d71b551	res_http_websocket: ensure control frames do not interfere with data Control frames (PING / PONG / CLOSE) can be received in the middle of a fragmented message. In order to ensure they do not interfere with the reassembly buffer, we exit early and do not return the payload to the caller. ASTERISK-28257 #close Change-Id: Ia5367144fe08ac6141bba3309517a48ec7f013bc	6 years ago
Joshua C. Colp	7d1736b59b	Merge "stasis / manager / ari: Better filter messages." into 16	6 years ago
Joshua C. Colp	84bed54bd7	Merge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix" into 16	6 years ago
Joshua C. Colp	fc28aa2de6	Merge "pjsip_transport_management: Shutdown transport immediately on disconnect" into 16	6 years ago
Xiemin Chen	f6cf837aed	bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix To avoid the stream name collide if there're more than one video track in one client. If client has multi video tracks, the name of ast_stream which represents each video track may be the same. Use the MSID:LABEL here because it's identifiable. ASTERISK-28196 #close Reported-by: xiemchen Change-Id: Ib62b2886e8d3a30e481d94616b0ceaeab68a870b	6 years ago
Joshua C. Colp	ef43bd0b43	Merge "res_http_websocket: respond to CLOSE opcode" into 16	6 years ago
Gerald Schnabel	735bd4d185	manager_channels: Fix throwing of HangupHandler manager events The type value extracted from stasis message data in channel_hangup_handler_cb isn't compared against the valid values "run", "pop" and "push". Thus the manager events HangupHandlerPush, HangupHandlerPop and HangupHandlerRun are never thrown. This regression was introduced by ASTERISK_21462. ASTERISK-28252 Change-Id: I9956e35e18da1873113644df1ddc3c7cd37bf524	6 years ago
Chris-Savinovich	05b79d16ab	Test_cel: Fails when DONT_OPTIMIZE is off A bug in GCC causes TEST_CEL to return failure under the following conditions: 1. TEST_FRAMEWORK on 2. DONT_OPTIMIZE off 3. Fedora and Ubuntu 4. GCC 8.2.1 5. Test name: test_cel_dial_pickup 6. There must exist a certain combination of multithreading. The bug affects arithmetic calculations when the optimization level is bigger than O1 and the -fpartial-inline flag is on. Provided these conditions, function ast_str_to_lower() fails to convert to lower case due to said function being of type force_inline. The solution is to remove the "force_inline" type declaration from function ast_str_to_lower() Change-Id: Ied32e0071f12ed9d5f3b4cdd878b2532a1c769d7	6 years ago
George Joseph	dbef559e0b	app_voicemail: Add Mailbox Aliases You can now define an "aliases" context in voicemail.conf whose entries point to actual mailboxes. These can be used anywhere the mailbox is specified. Example: [general] aliasescontext = myaliases [default] 1234 = yadayada [myaliases] 4321@devices = 1234@default Now you can use 4321@devices to refer to the 1234@default mailbox. This can be useful to provide channel drivers with constant mailbox specifications such as <extension>@devices leaving app_voicemail to control exactly which mailbox the alias points to. Now, only voicemail has to be reloaded to make changes instead of individual channel drivers which are usually more expensive to reload. Change-Id: I395b9205c91523a334fe971be0d1de4522067b04	6 years ago
Kevin Harwell	f1fb249132	res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown When a reliable transport is shutdown it's possible for the pjsip registrar resource shutdown handler to get called multiple times. If this happens and one of the threads is taking "too long" (slow database call for instance) then the others get blocked waiting to delete. Since it only takes one to delete the contact then the other threads should be able to continue on if one of the threads is currently "deleting". This patch makes it so now when a thread enters the shutdown handler it checks to see if a thread is currently already "deleting". If so, then the thread does not attempt to get the lock, and instead continues on thus avoiding the blockage. ASTERISK-28213 #close Change-Id: I7563ca596312b1dff4f3ab41483e89fe2862328a	6 years ago
George Joseph	9c11399be3	pjproject_bundled: Add patch for double free issue in timer heap Fixed #2172: Avoid double reference counter decrements in timer in the scenario of race condition between pj_timer_heap_cancel() and pj_timer_heap_poll(). Change-Id: If000e9438c83ac5084b678eb811e902c035bd2d8	6 years ago
Sean Bright	fb6e0df173	pjsip_transport_management: Shutdown transport immediately on disconnect The transport management code that checks for idle connections keeps a reference to PJSIP's transport for IDLE_TIMEOUT milliseconds (32000 by default). Because of this, if the transport is closed before this timeout, the idle checking code will keep the transport from actually being shutdown until the timeout expires. Rather than passing the AO2 object to the scheduler task, we just pass its key and look it up when it is time to potentially close the idle connection. The other transport management code handles cleaning up everything else for us. Additionally, because we use the address of the transport when generating its name, we concatenate an incrementing ID to the end of the name to guarantee uniqueness. Related to ASTERISK~28231 Change-Id: I02ee9f4073b6abca9169d30c47aa69b5e8ae9afb	6 years ago
Valentin Vidic	6506c5b1d4	channel.c: Fix segfault with Monitor(wav,file,i) If the Monitor is started with the i option the read_stream will be NULL. One code path in channel.c checks if write_stream is set but than uses read_stream instead causing a segfault. ASTERISK-28249 Change-Id: I1bae9126537be54895c7fea2d08dd9488d8cc525	6 years ago

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