asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	cc59e21409	media_cache.c: Various CLI improvements * Use ast_cli_completion_add() to improve performance when large number of cached items are present. * Only complete one URI for commands that only accept a single URI. * Change command documentation to wrap at 80 characters to improve readability. Change-Id: Iedb0a2c3541e49561bc231dca2dcc0ebd8612902	6 years ago
Friendly Automation	e9b9141d09	Merge "manager.c: Prevent the Originate action from running the Originate app" into 17	6 years ago
Ben Ford	665a94cb76	chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	6 years ago
George Joseph	6b1ba58967	manager.c: Prevent the Originate action from running the Originate app If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	6 years ago
George Joseph	6aa9ee1661	Merge "chan_dahdi: PRI span status may stay "Down, Active" after a short alarm" into 17	6 years ago
Friendly Automation	715e4d0e72	Merge "app_senddtmf: Add receive mode to AMI Action PlayDTMF" into 17	6 years ago
Kevin Harwell	b1c9176b3f	Merge "app_amd: Fixed timeout issue" into 17	6 years ago
Friendly Automation	ecdc5405f9	Merge "func_curl.c: Support custom http headers" into 17	6 years ago
George Joseph	8a02454ec6	Merge "res_pjsip_registrar: Fix uninitlized variable warning" into 17	6 years ago
George Joseph	2a92e6b576	CI: Fix missing script block in jenkinsfiles Change-Id: I9f44a3d5085ea7880fad1a3883a4820907e29ea3 (cherry picked from commit `95213b01d2`)	6 years ago
George Joseph	f0d1ce50af	CI: Fix missing script block in jenkinsfiles Change-Id: Ib4b6e4887695f230ea7a5b0c879b29fc5a13be4f (cherry picked from commit `d60f23ecbd`) (cherry picked from commit `ce8a23fdf9`)	6 years ago
George Joseph	46dceab33f	CI: Increase clone depth and do better cleanup The original clone depth of 10 was causing the need to rebase changes whose parent was older than the 10 commits. The clone depth has been increased to 100. Workspace cleanup was only happening for successful builds which wasn't enough to keep the 8G workspace in-memory drives on the docker slaves from filling up. Now the workspaces are cleaned up after every build regardless of success/failure. If you need to preserve builds temporarily, you can log into Jenkins/Manage Jenkins/Configure System and change the CLEANUP_WS_* environment variable for the job type you're troubleshooting to "FALSE". Change-Id: I0d7366e87cea714e5dbc9488caf718802fce75ca	6 years ago
Friendly Automation	43a7e44fbe	Merge "serializer: set high/low alert levels on whole pool" into 17	6 years ago
Sean Bright	76ef36fafc	res_pjsip_registrar: Fix uninitlized variable warning Fixes: error: ‘domain_name’ may be used uninitialized in this function Found with gcc (Ubuntu 9.2.1-9ubuntu2) 9.2.1 20191008 Change-Id: I44413b49ea1205aa25538142161deb73883c79e8	6 years ago
Michael Cargile	e23b2856d0	app_amd: Fixed timeout issue ASTERISK_28143 attempted to fix an issue where calls with no audio would never timeout. It did so by adding AST_FRAME_NULL as a frame type to process in its calculations. Unfortunately these frames seem to show up at irregular time intervals. This resulted in app_amd returning prematurely most of the time. * Removed AST_FRAME_NULL from the calculations * Added a check to see how much time has actually passed since app_amd began ASTERISK-28608 Change-Id: I642a21b02d389b17e40ccd5357754b034c3daa42	6 years ago
Frederic LE FOLL	d3dd4c5459	chan_dahdi: PRI span status may stay "Down, Active" after a short alarm Upon a short PRI disconnection, libpri may maintain Q.921 layer 'up' and may thus not send PRI_EVENT_DCHAN_DOWN / PRI_EVENT_DCHAN_UP events. If pri_event_alarm() clears DCHAN_UP status bit upon alarm detection and no Q.921 reconnection sequence occurs, chan_dahdi will keep seeing span status "Down" at the end of alarm. This patch modifies pri_event_alarm() in order to keep DCHAN_UP bit unchanged. libpri will send a PRI_EVENT_DCHAN_DOWN event if it detects a disconnection of Q.921 layer and this will clear DCHAN_UP if required. ASTERISK-28615 Change-Id: Ibe27df4971fd4c82cc6850020bce4a8b2692c996	6 years ago
lvl	6345a00228	app_senddtmf: Add receive mode to AMI Action PlayDTMF ASTERISK-28614 Change-Id: I183501297ae1dc294ae56b34acac9b0343eb2664	6 years ago
Friendly Automation	f1f28aa9e3	Merge "parking: Fix case where we can't get the parker." into 17	6 years ago
George Joseph	16066ce5fc	Merge "various files - fix some alerts raised by lgtm code analysis" into 17	6 years ago
Friendly Automation	d75099b845	Merge "func_env: Prevent FILE() from reading garbage at end-of-file" into 17	6 years ago
Friendly Automation	cb58d4cc2d	Merge "res_rtp_asterisk: Always return provided DTLS packet length." into 17	6 years ago
Alexei Gradinari	649733612d	serializer: set high/low alert levels on whole pool The current code sets alert levels starting from index 1. Need to set on whole pool starting from index 0. Change-Id: I5decbb43160954fb9a512f04302637fc666b6f5d	6 years ago
Joshua Colp	0c486e7edf	res_rtp_asterisk: Always return provided DTLS packet length. OpenSSL can not tolerate if the packet sent out does not match the length that it provided to the sender. This change lies and says that each time the full packet was sent. If a problem does occur then a retransmission will occur as appropriate. ASTERISK-28576 Change-Id: Id42455b15c9dc4eb987c8c023ece6fbf3c22a449	6 years ago
Sean Bright	92bb381d5d	func_env: Prevent FILE() from reading garbage at end-of-file If the last line of a file does not have a terminating EOL sequence, we potentially add garbage to the value returned from the FILE() function. There is no overflow potential here as we are reading from a buffer of a known size, we are just reading too much of it. ASTERISK-26481 #close Change-Id: I50dd4fcf416fb3c83150040a1a79a59d9eb1ae01	6 years ago
Kevin Harwell	8b4610acfe	bridge_softmix: clear hold when joining a softmix bridge MOH continues to play to a channel if that channel was on hold prior to entering a softmix bridge. MOH will not stop even if the original "holder" attempts an unhold. For the most part a softmix bridge ignores holds, so a participating channel shouldn't join while on hold. This patch checks to see if the channel joining the softmix bridge is currently on hold. If so then it indicates an unhold. ASTERISK-28618 Change-Id: I66ccd4efc80f5b4c3dd68186b379eb442916392b	6 years ago
Kevin Harwell	8bc6fa0fbd	various files - fix some alerts raised by lgtm code analysis This patch fixes several issues reported by the lgtm code analysis tool: https://lgtm.com/projects/g/asterisk/asterisk Not all reported issues were addressed in this patch. This patch mostly fixes confirmed reported errors, potential problematic code points, and a few other "low hanging" warnings or recommendations found in core supported modules. These include, but are not limited to the following: * innapropriate stack allocation in loops * buffer overflows * variable declaration "hiding" another variable declaration * comparisons results that are always the same * ambiguously signed bit-field members * missing header guards Change-Id: Id4a881686605d26c94ab5409bc70fcc21efacc25	6 years ago
Martin Tomec	d579ec9cdf	func_curl.c: Support custom http headers When user wants to send json data, the default Content-Type header is incorect (application/x-www-form-urlencoded). This patch allows to set any custom headers so the Content-Type header can be overriden. User can set multiple headers by multiple calls of curlopt(). This approach is not consistent with other parameters, but is more readable in dialplan than one call with multiple headers. ASTERISK-28613 Change-Id: I4dd68c3f4e25362ef941d73a3861f58348dcfbf9	6 years ago
Joshua Colp	de433cdcaf	parking: Fix case where we can't get the parker. ASTERISK-28616 Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d	6 years ago
Joshua Colp	d638d9c6c6	parking: Use channel snapshot instead of channel. There exists a scenario where a thread can hold a lock on the channels container while trying to lock a bridge. At the same time another thread can hold the lock for said bridge while attempting to retrieve a channel. This causes a deadlock. This change fixes this scenario by retrieving a channel snapshot instead of a channel, as information present in the snapshot is all that is needed. ASTERISK-28616 Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2	6 years ago
Kevin Harwell	3084a6c617	Merge "res_pjsip_session: initialize pending's topology to endpoint's" into 17	6 years ago
Kevin Harwell	ea3daa94c8	res_pjsip_session: initialize pending's topology to endpoint's Found during some testing, there is a race condition between selecting an appropriate bridge type for a call versus the applying of media on the callee's session. In some instances a native bridge type would have been chosen, but due to the callee's media not yet being established at bridge compatibility check time the simple bridge type is picked instead. When using chan_pjsip this initiates a topology change event. The topologies are then compared for the two sessions. However, when the topology was created for the caller its streams are initialized to "inactive". This topology is then used as a base when creating the callee's topology, and streams. Soon after the caller's topology's stream(s) get updated based on the sdp (get set to sendrecv in the failing scenario). Now when the topology change event is raised, and the two topologies are compared, the comparison fails due to a stream state mismatch (sendrecv vs inactive). And since they differ a reinvite is sent out (to the caller in this case). This patch makes it such that when the caller's topology is initially created it gets created based on its configured endpoint's media topology. When the endpoint's topology is created its stream's state(s) are initialized to sendrecv instead of inactive. Subsequently, now when the callee's topology is created its topology streams are now initialized to sendrecv. Thus when the topology change event occurs due to the mentioned scenario the stream states match for the given sessions, and the reinvite is not sent unless due to some other valid mismatch. Note, this patch only changes one pending media state's creation point. It's possible other places could be changed, however for now it was deemed best to only alter what's here. Change-Id: I6ba3a6a75f64824a1b963044c37acbe951c389c7	6 years ago
George Joseph	7202624b3b	stasis: Don't hold app_registry and session locks unnecessarily resource_events:stasis_app_message_handler() was locking the session, then attempting to determine if the app had debug enabled which locked the app_registry container. res_stasis:__stasis_app_register was locking the app_registry container then calling app_update which caused app_handler (which locks the session) to run. The result was a deadlock. * Updated resource_events:stasis_app_message_handler() to determine if debug was set (which locks the app_registry) before obtaining the session lock. * Updated res_stasis:__stasis_app_register to release the app_registry container lock before calling app_update (which locks the sesison). ASTERISK-28423 Reported by Ross Beer Change-Id: I58c69d08cb372852a63933608e4d6c3e456247b4	6 years ago
George Joseph	36b28c98dd	Build: Fix compile issues with seldom used modules The following modules needed tweaks for API changes. addons/cdr_mysql.c addons/chan_ooh323.c apps/app_meetme.c ASTERISK-28604 Change-Id: Ib40e513ae55b5114be035cdc929abb6a8ce2d06d	6 years ago
Joshua Colp	eea2d499f4	res_pjsip_outbound_registration: Extend documentation for "max_retries". If the "max_retries" option is set to 0 then upon failure no further attemps are made, so explicitly document the behavior. ASTERISK-28602 Change-Id: I1e30daae9dd6c49ce18744164214d3def505acbf	6 years ago
Joshua Colp	43ea6e21db	Merge "res_calendar: Resolve memory leak on calendar destruction" into 17	6 years ago
Sean Bright	b3792e1288	res_calendar: Resolve memory leak on calendar destruction Calling ne_uri_parse allocates memory that needs to be freed with a corresponding call to ne_uri_free. ASTERISK-28572 #close Change-Id: I8a6834da27000a6807d89cb7a157b2a88fcb5e61	6 years ago
Joshua Colp	e37d546109	res_ari_events: Add module reference when a WebSocket is open. This change ensures that the module isn't unloaded when a WebSocket is open. Previously it was possible to unload the module manually or during shutdown which could cause a crash when any active WebSockets were terminated. ASTERISK-28585 Change-Id: I85c71ab112f99875b586419a34c08c8b34c14c5c	6 years ago
George Joseph	1b4502ec5d	Merge "ExternalMedia: Change return object from ExternalMedia to Channel" into 17	6 years ago
Friendly Automation	8b0661df75	Merge "utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN" into 17	6 years ago
Joshua Colp	0cfe9186b0	Merge "app_voicemail.c: Support multiple file formats for forwarded messages." into 17	6 years ago
Friendly Automation	500e5f488f	Merge "Crash during "pjsip show channelstats" execution" into 17	6 years ago
Sean Bright	b3c56c7fa5	utils.h: Set lower bound for thread stack size to PTHREAD_STACK_MIN ASTERISK-28590 #close Change-Id: I51abce00c04d0a06550bda5205580705185b9c1c	6 years ago
Joshua Colp	4d67cd10ba	Merge "res_rtp_asterisk: Remove a log message that slipped in." into 17	6 years ago
George Joseph	2d665091a3	ExternalMedia: Change return object from ExternalMedia to Channel When we created the External Media addition to ARI we created an ExternalMedia object to be returned from the channels/externalMedia REST endpoint. This object contained the channel object that was created plus local_address and local_port attributes (which are also in the Channel variables). At the time, we thought that creating an ExternalMedia object would give us more flexibility in the future but as we created the sample speech to text application, we discovered that it doesn't work so well with ARI client libraries that a) don't have the ExternalMedia object defined and/or b) can't promote the embedded channel structure to a first-class Channel object. This change causes the channels/externalMedia REST endpoint to return a Channel object (like channels/create and channels/originate) instead of the ExternalMedia object. Change-Id: If280094debd35102cf21e0a31a5e0846fec14af9	6 years ago
Salah Ahmed	40acd7d198	Crash during "pjsip show channelstats" execution During execution "pjsip show channelstats" cli command by an external module asterisk crashed. It seems this is a separate thread running to fetch and print rtp stats. The crash happened on the ao2_lock method, just before it going to read the rtp stats on a rtp instance. According to gdb backtrace log, it seems the session media was already cleaned up at that moment. ASTERISK-28578 Change-Id: I3e05980dd4694577be6d39be2c21a5736bae3c6f	6 years ago
Joshua Colp	b8ae799ca9	res_rtp_asterisk: Remove a log message that slipped in. This was only supposed to be for testing, so now it can be removed. Change-Id: I3dfc2e776e70b3196aeed5688372ea80c0214b59	6 years ago
Sean Bright	4bc1c170cd	README-SERIOUSLY.bestpractices.md: Speling correetions. ASTERISK-28586 #close Change-Id: I43dc4e8bd9dc685b17695b215a5360314074734f	6 years ago
Joshua Colp	b590e8f23e	Merge "test_res_rtp: Enable FIR and REMB nominal tests." into 17	6 years ago
cmaj	aa0973f868	app_voicemail.c: Support multiple file formats for forwarded messages. If you specify multiple formats in voicemail.conf, eg. "format = gsm\|wav" and are using realtime ODBC backend, only the first format gets stored in the database. So when you forward a message later on, there is a bug generating the email, related to the stored format (GSM) being different than the desired email format (WAV) specified for the user. Sox can handle this, but Asterisk needs to tell sox exactly what to do. ASTERISK-22192 Change-Id: I7321e7f7e7c58adbf41dd4fd7191c887b9b2eafd	6 years ago
George Joseph	3c712391c6	Merge "res_pjsip_mwi: potential double unref, and potential unwanted double link" into 17	6 years ago

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