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Author	SHA1	Message	Date
Friendly Automation	9d6161ee6a	Merge "res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen"	6 years ago
Friendly Automation	b408bfdd06	Merge "res_mwi_devstate.c: New module to allow presence subs to VM boxes"	6 years ago
George Joseph	e0fc663295	CI: Update jenkinsfiles with new Gerrit URLs ... The recent upgrade of Gerrit to 2.16 elimiated referencing a repository in a way the jenkinsfiles were relying on so the URL references were changed to a more consistent and supported format. Change-Id: I2e8e3f213b9a96bb1b27665eca4a9a24bc49820e (cherry picked from commit 5ce084579f)	6 years ago
Joshua C. Colp	f48a9c2520	Merge "rest-api-templates/asterisk_processor - replace http line breaks with line feed"	6 years ago
George Joseph	9ee76cf070	res_mwi_devstate.c: New module to allow presence subs to VM boxes ... This module allows presence subscriptions to voicemail boxes. This allows common BLF keys to act as voicemail waiting indicators. ASTERISK-28301 Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd	6 years ago
Torrey Searle	360f543677	res/res_rtp_asterisk: smoother can cause wrong timestamps if dtmf happen ... Delivery timeval in the smoother object will fall behind while a DTMF is being generated. This can eventually lead to invalid rtp timestamps. To prevent this from happening the smoother needs to be reset after every DTMF to keep the timing up to date. ASTERISK-28303 #close Change-Id: Iaba3f7b428ebd72a4caa90e13b829ab4f088310f	6 years ago
Friendly Automation	12c359ba5d	Merge "http.c: Support separated HTTP request"	6 years ago
Joshua C. Colp	18f92cb6f3	Merge "taskprocessor: Enable subsystems and overload by subsystem"	6 years ago
Friendly Automation	b37283e694	Merge "app_queue: Enable set the wrapuptime from AddQueueMember application"	6 years ago
Joshua C. Colp	5d08b4c463	Merge "res_ari_applications: Fix incorrect call to ao2_lock."	6 years ago
Joshua C. Colp	5f29d3b581	Merge "Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY"	6 years ago
Kevin Harwell	574128dec6	rest-api-templates/asterisk_processor - replace http line breaks with line feed ... Including line breaks (<br>, <br/>, <br />) in certain parts of the rest-api json definition (e.g. summary, notes) displays them correctly in swagger. However, when the field gets converted to the wiki format those breaks get escaped and show up in the text as the actual string literal "<br>" etc... This patch makes it so when converting to the wiki format it replaces all line break values (<br>, etc...) with line feeds ('\n'). Change-Id: Ie1c9faa0d1c5d622804cc0a21ce769095b08aa3d	6 years ago
Joshua C. Colp	e687cf214d	res_ari_applications: Fix incorrect call to ao2_lock. ... When listing the applications the apps lock was incorrectly locked twice instead of being locked and then unlocked. ASTERISK-28302 Change-Id: If7d064592a9e88c0f1049214c50e02be6dabf79e	6 years ago
Joshua Colp	e6b67b2a5d	res_pjsip_sdp_rtp: Allow only single ssrc attribute. ... When processing SSRC attributes we were iterating through all of them, even though we only need to know the remote SSRC once. This was problematic because some browsers group SSRCs together on a stream, and due to our negotiation only end up using the first one. Since we set the second one as the remote SSRC we would drop the received media from them instead of allowing it through. In the future this may be extended to allow SSRC groups and to use information from the attributes. Change-Id: I4dc87087dbe56a83aa65f0f897bbd4ca75ec1270	6 years ago
Joshua C. Colp	88f77b3983	Merge "stasis: Store subscriber uniqueids with topic statistics."	6 years ago
Joshua C. Colp	cf1b64aaf0	Merge "res_pjsip_session Added rtcp stats result vector into the session"	6 years ago
Sungtae Kim	b4ccaad671	http.c: Support separated HTTP request ... Currently, the Asterisk does not support seperated HTTP request. This patch make the Asterisk enables to wait lest part of HTTP request. Also increases acceptable HTTP body length to 40k to support more larger request. ASTERISK-28236 Change-Id: I48a401aa64a21c3b37bf3cb4e0486d64b7dd8aa1	6 years ago
George Joseph	bc8dead610	Core: Increase AST_PBX_MAX_STACK to 512 if not LOW_MEMORY ... The current settings AST_PBX_MAX_STACK is 128 entries which is too low for some FreePBX installations with complex parking arrangements. Increased to 512 if LOW_MEMORY is not defined. ASTERISK-28300 Change-Id: I7c4b540bc92e6642df0f3da639b003f7da8b1299	6 years ago
Joshua C. Colp	a286f546f1	stasis: Store subscriber uniqueids with topic statistics. ... This change provides an easier mechanism to determine which subscribers are subscribed to a topic. Using this you can inspect the specific subscribers for further details. Change-Id: I8deea21703cd5c5357b85593b46c3eaf24e18c0c	6 years ago
George Joseph	c2adeb9dc2	taskprocessor: Enable subsystems and overload by subsystem ... To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	6 years ago
Kevin Harwell	8681fc9db7	ARI event type filtering ... Event type filtering is now enabled, and configurable per application. An app is now able to specify which events are sent to the application by configuring an allowed and/or disallowed list(s). This can be done by issuing the following: PUT /applications/{applicationName}/eventFilter And then enumerating the allowed/disallowed event types as a body parameter. ASTERISK-28106 Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b	6 years ago
George Joseph	0eb27e1970	Merge "json.c/strings.c - Add a couple of utility functions"	6 years ago
Friendly Automation	de64949f6a	Merge "chan_pjsip: Changed to continued after invalid media for pjsip show channelstats"	6 years ago
George Joseph	e20ddfde46	Merge "CI: Use tmpfs option to Docker instead of mount."	6 years ago
Joshua Colp	f4c9a351d8	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	8f1b3edde8	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
Rodrigo Ramírez Norambuena	ce0523a57e	app_queue: Enable set the wrapuptime from AddQueueMember application ... This change add ability to set the wrapuptime per-member using the AddQueueMember application. The feature to set wrapuptime per member was include in the issue ASTERISK-27483 for static member by configuration file and was not added to set from AddQueueMember. ASTERISK-28055 #close Change-Id: I7c7ee4a6f804922cd7c42cb02eea26eb3806c6cf	6 years ago
Torrey Searle	8ea9608efb	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
sungtae kim	fb651756c7	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	7e1d881d89	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
George Joseph	16084df36f	Merge "ci: Rerun unit tests when non-code changes occur."	6 years ago
Friendly Automation	d4a78063eb	Merge "res_odbc: Add basic query logging."	6 years ago
Joshua Colp	c2ea9c90a2	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	61a8f79a29	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	54a912b26d	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	68bb6ef6cb	Merge "sounds: Sort 'core show sounds' output"	6 years ago
Sungtae Kim	5a2a7d65b5	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	e2bbab17b3	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
Friendly Automation	99186c4b15	Merge "pjsip/config_global: regcontext context not created"	6 years ago
George Joseph	00d7b6df49	Merge "res_stasis: Auto-create context and extens on Stasis app launch."	6 years ago
George Joseph	4787b47da0	Merge "Added ARI resource /ari/asterisk/ping"	6 years ago
Sean Bright	f174eb4ac1	sounds: Sort 'core show sounds' output ... Change-Id: Ib39052a745040f75eb635f15a042da15b20e22ab	6 years ago
Joshua C. Colp	f0737352ee	Merge "bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64"	6 years ago
Ben Ford	3f9c5fba95	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
George Joseph	ac2d302c2c	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
sungtae kim	ac90968afd	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
Kevin Harwell	f668db9ba0	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	7071e9d64c	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	6118b65a5a	Merge "codecs.conf.sample: update codec opus docs"	6 years ago
George Joseph	42d63b69b0	Merge "format_g726: add support for seeking"	6 years ago