32193 Commits (18f5f5fc99a8106c1856d13aeeeeafa2b13ef033)

 

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Author	SHA1	Message	Date
Friendly Automation	99186c4b15	Merge "pjsip/config_global: regcontext context not created"	7 years ago
George Joseph	00d7b6df49	Merge "res_stasis: Auto-create context and extens on Stasis app launch."	7 years ago
George Joseph	4787b47da0	Merge "Added ARI resource /ari/asterisk/ping"	7 years ago
Sean Bright	f174eb4ac1	sounds: Sort 'core show sounds' output ... Change-Id: Ib39052a745040f75eb635f15a042da15b20e22ab	7 years ago
Joshua C. Colp	f0737352ee	Merge "bundled-jansson: On OpenSuse Leap libjansson.a was placed in lib64"	7 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	7 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	7 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	7 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	7 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	7 years ago
George Joseph	6118b65a5a	Merge "codecs.conf.sample: update codec opus docs"	7 years ago
George Joseph	42d63b69b0	Merge "format_g726: add support for seeking"	7 years ago
George Joseph	423a51ca11	Merge "res_http_websocket: ensure control frames do not interfere with data"	7 years ago
Kevin Harwell	0bcaadc037	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	7 years ago
eyalhasson	aede739778	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	7 years ago
Joshua C. Colp	83fa4510fc	Merge "build : Fix cross-compilation errors"	7 years ago
Joshua C. Colp	1797cd2071	Merge "app_voicemail: Add Mailbox Aliases"	7 years ago
Joshua C. Colp	5e3d7ffdca	Merge "res_pjsip_registrar: mitigate blocked threads on reliable transport shutdown"	7 years ago
Joshua C. Colp	fc8db07208	Merge "Test_cel: Fails when DONT_OPTIMIZE is off"	7 years ago
Friendly Automation	389b2ab39b	Merge "manager_channels: Fix throwing of HangupHandler manager events"	7 years ago
Jeremy Lainé	69e9fd63e1	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	7 years ago
Jean Aunis	d9fae4a824	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	7 years ago
Joshua C. Colp	884aaa5f72	Merge "stasis / manager / ari: Better filter messages."	7 years ago
Joshua C. Colp	3655d304af	Merge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix"	7 years ago
Joshua C. Colp	4e41be0a59	Merge "pjsip_transport_management: Shutdown transport immediately on disconnect"	7 years ago
Joshua C. Colp	ca5ca2e861	Merge "res_http_websocket: respond to CLOSE opcode"	7 years ago
Gerald Schnabel	f9ca0afb39	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	7 years ago
Chris-Savinovich	1c8378bbc9	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	7 years ago
George Joseph	c6980e32ae	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	7 years ago
Kevin Harwell	b82d2856b4	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	7 years ago
George Joseph	deffb8a6e0	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	7 years ago
Xiemin Chen	a526676836	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	7 years ago
Joshua C. Colp	6f5bc854ab	Merge "channel.c: Fix segfault with Monitor(wav,file,i)"	7 years ago
Jeremy Lainé	0b8867f7d6	res_http_websocket: respond to CLOSE opcode ... This ensures that Asterisk responds properly to frames received from a client with opcode 8 (CLOSE) by echoing back the status code in its own CLOSE frame. Handling of the CLOSE opcode is moved up with the rest of the opcodes so that unmasking gets applied. The payload is no longer returned to the caller, but neither ARI nor the chan_sip nor pjsip made use of the payload, which is a good thing since it was masked. ASTERISK-28231 #close Change-Id: Icb1b60205fc77ee970ddc91d1f545671781344cf	7 years ago
Sean Bright	20f672539e	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	7 years ago
Valentin Vidic	17f76d27cc	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	7 years ago
Joshua C. Colp	1323730f6c	stasis / manager / ari: Better filter messages. ... Previously both AMI and ARI used a default route on their stasis message router to handle some of the messages for publishing out their respective connection. This caused messages to be given to their subscription that could not be formatted into AMI or JSON. This change adds an API call to the stasis message router which allows a default route to be set as well as formatters that the default route is expecting. This allows both AMI and ARI to specify that their default route only wants messages of their given formatter. By doing so stasis can more intelligently filter at publishing time so that they do not receive messages which will not be turned into AMI or JSON. ASTERISK-28244 Change-Id: I65272819a53ce99f869181d1d370da559a7d1703	7 years ago
Sean Bright	58b55f2a30	sched: Make sched_settime() return void because it cannot fail ... Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5	7 years ago
Sean Bright	2b8602e8cf	res_pjsip_transport_websocket: Don't assert on 0 length payloads ... When --enable-dev-mode is used, pjsip_tpmgr_receive_packet() will assert if passed a payload length of 0, so treat empty frames as if we didn't receive them. Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48	7 years ago
Joshua C. Colp	b19210214f	Merge "res_pjsip: add option to enable ContactStatus event when contact is updated"	7 years ago
Joshua C. Colp	c2e02645b4	Merge "stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure."	7 years ago
Joshua C. Colp	a3f1f82272	Merge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled."	7 years ago
Joshua C. Colp	2bbcc76f79	Merge "RTP: reset DTMF last seqno/timestamp on RTP renegotiation"	7 years ago
Joshua C. Colp	b4523ef334	Merge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail"	7 years ago
mohitdhiman	d60ee2eeae	stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure. ... During Bridging of two channels if masquerade operation is performed on a channel (clone channel) which was created with endpoint details (ast_channel_alloc_with_endpoint()) and the original channel which is created without endpoint details (ast_channel_alloc()) then both the channels must exchange their endpoint details or else after masquerade when clone channel is being destroyed the endpoint cleanup callbacks will be destroyed too and after call completion unique_id of original channel will still be there in ast_endpoint structure's channel_ids container. ASTERISK-28197 Change-Id: I97ce73da390af20fd082fb09d722a6fe9cb2f39d	7 years ago
Alexei Gradinari	f0546d1d87	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	7 years ago
Joshua Colp	18e206381a	res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled. ... For video streams it was possible for the abs-send-time information to be placed into RTP streams even if not negotiated. Depending on the endpoint in use this could cause video to not flow. We now only enable abs-send-time for negotiation if WebRTC is enabled. ASTERISK-28230 Change-Id: I0eb682302f8da3a4ea3c42e839208d55f825ed0c	7 years ago
Diederik de Groot	7bd30905fd	RAII: Change order or variables in clang version ... This prevents use-after-scope issues when unwinding the stack, which happens in reverse order. The varname variable needs to remain alive for the destruction to be able to access it. Issue was found using clang + address-sanitizer. ASTERISK-28232 #close Change-Id: I00811c34ae910836a5fb6d22304528aef92624db	7 years ago
Alexei Gradinari	f662a26ea0	RTP: reset DTMF last seqno/timestamp on RTP renegotiation ... The remote side may start a new stream when renegotiating RTP. Need to reset the DTMF last sequence number and the timestamp of the last END packet on RTP renegotiation. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA5XX and codec g722. On SIP session update the SPA50X resets stream and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: Ic72b4497e74d801b27a635559c1cf29c16c95254	7 years ago
Joshua C. Colp	28dbb06e8c	Merge "ast_coredumper: Refactor the pid determination process"	7 years ago