31789 Commits (7d1736b59ba8e85cb3203a046317b08165586ea2)

 

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Author	SHA1	Message	Date
Joshua C. Colp	7d1736b59b	Merge "stasis / manager / ari: Better filter messages." into 16	7 years ago
Joshua C. Colp	84bed54bd7	Merge "bridge_softmix: Use MSID:LABEL metadata as the cloned stream's appendix" into 16	7 years ago
Joshua C. Colp	fc28aa2de6	Merge "pjsip_transport_management: Shutdown transport immediately on disconnect" into 16	7 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	7 years ago
Joshua C. Colp	ef43bd0b43	Merge "res_http_websocket: respond to CLOSE opcode" into 16	7 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	7 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	7 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	7 years ago
Joshua C. Colp	fcd07c34fb	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	011e46d5a6	sched: Make sched_settime() return void because it cannot fail ... Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5	7 years ago
Jeremy Lainé	21a1feece2	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	44a862fb57	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	9e7e150a13	Merge "res_pjsip: add option to enable ContactStatus event when contact is updated" into 16	7 years ago
Friendly Automation	d76896a30b	Merge "stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure." into 16	7 years ago
Joshua C. Colp	11d63c9e4f	Merge "res_pjsip_sdp_rtp: Only enable abs-send-time when WebRTC is enabled." into 16	7 years ago
Joshua C. Colp	32934180f9	Merge "RTP: reset DTMF last seqno/timestamp on RTP renegotiation" into 16	7 years ago
Joshua C. Colp	72fdba63a8	Merge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail" into 16	7 years ago
mohitdhiman	4b24da607e	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: Ied0451f378a3f2a36acc8c0984959a69895efa17	7 years ago
Diederik de Groot	d2c182b6ab	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	7f22c9f4b7	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	c6271155fb	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
Alexei Gradinari	c0e57e458b	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	701cd2ee58	Merge "ast_coredumper: Refactor the pid determination process" into 16	7 years ago
Friendly Automation	140ed330ac	Merge "stasis: Fix ABI between DEVMODE and non-DEVMODE." into 16	7 years ago
Joshua C. Colp	1e9da4dba6	Merge "stasic.c: Fix printf format type mismatches with arguments." into 16	7 years ago
Joshua C. Colp	92ee1bdfbb	Merge "app_queue: Fix crash when using 'b' option on non-ringall queue." into 16	7 years ago
Bryan Boatright	92298434bd	app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail ... If a voicemail is marked "urgent" then the VM_MESSAGEFILE channel variable is not updated correctly since urgent messages are in a different directory. The fix is to update the channel variable when the path to the urgent message is created. ASTERISK-28225 Change-Id: I8efbace06e6122ea0793f7bdb073d4378e8274ca	7 years ago
Joshua Colp	2d9482695d	app_queue: Fix crash when using 'b' option on non-ringall queue. ... When using the 'b' option to Queue with a queue that was not configured for ring all a crash would occur as the wrong pointer would be used. ASTERISK-28218 Change-Id: If1390f64e321047dff24fd2410c95dde74904980	7 years ago
Richard Mudgett	f196078705	stasic.c: Fix printf format type mismatches with arguments. ... An int64_t is not likely the same size as a long. * Changed the int64_t values in the statistics structs to longs so casting is not necessary when generating the formatted CLI output. The offending members did not need to be int64_t anyway as they were only set by an int type variable which was already truncating bits. * Reordered the statistics structs to reduce potential padding bytes. Change-Id: Ic090a070e9dc4ca650ebdb9c01ed50a581289962	7 years ago
George Joseph	4ff64bfc2c	Merge "backtrace.c: Fix casting pointer to/from integral type." into 16	7 years ago
Corey Farrell	44a7faca21	stasis: Fix ABI between DEVMODE and non-DEVMODE. ... Create compatibility stubs for maximum ABI compatibility. ASTERISK-28212 #close Change-Id: I872c04842ab6b61e9dd6d37e4166bc619aa20626	7 years ago
George Joseph	ff2ed4eeee	Revert "stasis_cache: Stop caching stasis subscription change messages" ... This commit caused issues with polling when combined with the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis" This reverts commit 17d6d9e1e7. ASTERISK-28222 Reported by: abelbeck Change-Id: Ib6a16cbe2c0b74c76c64264f525ab647959c1483	7 years ago
George Joseph	3efe5061d5	ast_coredumper: Refactor the pid determination process ... In order to get a dump of the running process, we need to find the pid of the main asterisk process. This can be tricky if there are also instances of "asterisk -r" running or if an alternate location for asterisk.conf was specified on the command line with the -C option that also specified an alternation location for the pid file. So now... 1. We find the asterisk executable with "which" or the --asterisk-bin command line option. 2. If there's only 1 process with an executable path that matches, we use that pid. If not... 3. We try "<asterisk-bin> -rx 'core show settings'" and parse the output to find the pidfile, then read that for the pid. If that didn't work... 4. We get a list of all the pids matching <asterisk-bin> and look in /proc/<pid>/cmdline for a -C argument and retry the "core show settings" using the same -C option. We can't parse the output of "ps" to get the -C path because it may contain spaces. The contents of /proc/<pid>/cmdline are delimited by NULLs. For BSDs we may have to mount /proc first. :( ASTERISK-28221 Reported by: Andrew Nagy Change-Id: I8aa1f3f912f949df2b5348908803c636bde1d57c	7 years ago
Richard Mudgett	59717b5e85	backtrace.c: Fix casting pointer to/from integral type. ... The backtrace library bfd.h include file does not get the sizes of pointers and ints right on some platforms. On my old test box the size of bfd_vma is 8 while the size of a pointer is 4. gcc on the box complains of the integer casting to/from pointers size mismatch. * uintptr_t to the rescue by doing an appropriate two stage cast. Change-Id: Icb2621583f50c8728de08a3c824d95fe53cc45d0	7 years ago
Joshua C. Colp	08e6cb6480	Merge "res/res_ari: Add additional hangup reasons" into 16	7 years ago
Friendly Automation	0110532bf9	Merge "app_voicemail: Don't delete mailbox state unless mailbox is deleted" into 16	7 years ago
George Joseph	4cbe04de77	Merge "res_pjsip: Patch for res_pjsip_* module load/reload crash" into 16	7 years ago
George Joseph	955129e9a4	Merge "res_rtp_asterisk: Remove some unused structure fields." into 16	7 years ago
George Joseph	aebb822d1f	app_voicemail: Don't delete mailbox state unless mailbox is deleted ... The free_user function was automatically deleting the stasis mailbox state but this only makes sense when the mailbox is actually deleted, not just the structure freed. This was causing issues where leave_voicemail would publish the mwi message to stasis and delete the state before the message could be processed by res_pjsip_mwi. * Removed the delete of state from free_user(). * Created a new free_user_final() function that both frees the data structure and deletes the state. This function is only called during module load/unload where it's appropriate to delete the state. ASTERISK-28215 Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd	7 years ago
Friendly Automation	9b57199a7a	Merge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set" into 16	7 years ago
Sean Bright	970805180e	res_rtp_asterisk: Remove some unused structure fields. ... All of the fields that were removed were no longer referenced except for 'lastrxts' and 'rxseqno' which were only ever written to. Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c	7 years ago
Joshua C. Colp	9e159914dc	Merge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels" into 16	7 years ago
Sean Bright	f60afac587	res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set ... The profile-iop octet (the 2nd) of profile-level-id can be zero according to RFC 6184 Section 8.1. So we ignore its value when deciding to include profile-level-id in the outgoing SDP. ASTERISK-27959 #close Reported by: David Kuehling Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73	7 years ago
Joshua C. Colp	84d693a689	Merge "utils: Don't set or clear flags that don't need setting or clearing" into 16	7 years ago
Joshua C. Colp	b924594104	Merge "confbridge: announce to the marked users when they join an empty conference" into 16	7 years ago
Sean Bright	640aac768b	bridge_builtin_features.c: Set auto(mix)mon variables on both channels ... This is how features behaved up through Asterisk 11, but was changed when the new bridging framework was implemented in Asterisk 12. Reported by rrittgarn in #asterisk. Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b	7 years ago
Friendly Automation	67e91d012b	Merge "stasis: Add statistics gathering in developer mode." into 16	7 years ago
Joshua C. Colp	36db878adc	Merge "Use non-blocking socket() and pipe() wrappers" into 16	7 years ago
Alexei Gradinari	2610379605	confbridge: announce to the marked users when they join an empty conference ... Currently the file sound_only_person is not played when a marked user (with announce_only_user=yes) joins an empty conference. This patch fixes it. ASTERISK-28201 #close Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4	7 years ago
Joshua C. Colp	68ec7d93e8	stasis: Add statistics gathering in developer mode. ... This change adds statistics gathering to Stasis topics, subscriptions, and message types. These can be viewed using CLI commands and provide insight into how Stasis is used and how long certain operations take to execute. These are only available when Asterisk is compiled in developer mode and do not have any impact under normal operation. ASTERISK-28117 Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f	7 years ago