30901 Commits (6531f8b2ea50c0e9ead60f29f355912df907c43b)

 

All Branches   

Search

Author	SHA1	Message	Date
Joshua C. Colp	128b90a437	Merge "res_http_websocket: respond to CLOSE opcode" into 13	6 years ago
Gerald Schnabel	8791dda672	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	9b85882f04	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	d0eccac3ff	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	2e1cbcde7d	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
Joshua C. Colp	e66d4d1d2a	Merge "pjproject_bundled: Add patch for double free issue in timer heap" into 13	6 years ago
George Joseph	8414ca191e	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
Jeremy Lainé	da50162fa4	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	6 years ago
Sean Bright	9cd6161083	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	efde1ab138	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
Joshua C. Colp	42922af7b8	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	6 years ago
Sean Bright	28da2948d1	sched: Make sched_settime() return void because it cannot fail ... Change-Id: I66b8b2b2778f186919d73ae9bf592104b8fb1cd5	6 years ago
Sean Bright	d87fc1c591	res_pjsip_transport_websocket: Don't assert on 0 length payloads ... If an external pjproject is used and built with assertions enabled, we will assert if passed a payload length of 0, so treat empty frames as if we didn't receive them. Change-Id: I9c5fdccd89cc8d2f3ed7e3ee405ef0fc78178f48	6 years ago
Friendly Automation	5254ebf09f	Merge "res_pjsip: add option to disable ContactStatus event when contact is updated" into 13	6 years ago
Joshua C. Colp	94f9c681d4	Merge "stasis/endpoint: Fix memory leak of channel_ids in ast_endpoint structure." into 13	6 years ago
Friendly Automation	c7a7638236	Merge "RTP: reset DTMF last seqno/timestamp on RTP renegotiation" into 13	6 years ago
Friendly Automation	7de0b919c9	Merge "app_voicemail: Fix Channel variable VM_MESSAGEFILE for "urgent" voicemail" into 13	6 years ago
Alexei Gradinari	4a8564cafa	res_pjsip: add option to disable ContactStatus event when contact is updated ... This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to have the same performance benefits as version 16. By default old behavior, i.e. the ContactStatus event will be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	6 years ago
mohitdhiman	84a7b4d356	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	6 years ago
Diederik de Groot	790626ec82	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	6 years ago
Joshua C. Colp	d309f1d9f7	Merge "ast_coredumper: Refactor the pid determination process" into 13	7 years ago
George Joseph	e3439d3011	Merge "stasis: Fix ABI between DEVMODE and non-DEVMODE." into 13	7 years ago
Bryan Boatright	250812df09	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
Richard Mudgett	b26ec935ec	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	7f2a953691	Merge "backtrace.c: Fix casting pointer to/from integral type." into 13	7 years ago
Corey Farrell	240f339938	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	96f8b19b1f	Revert "stasis_cache: Stop caching stasis subscription change messages" ... This reverts commit 94a4eea7f6. This commit caused issues with polling when combined with the revert commit "Revert "app_voicemail: Remove need to subscribe to stasis" ASTERISK-28222 Reported by: abelbeck Change-Id: I0aaab5218a282706dfc6fcbbaeead228d599b99b	7 years ago
George Joseph	a39dd50436	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
Alexei Gradinari	f9face3612	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
Richard Mudgett	a551539048	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	5dc38c2fa9	Merge "app_voicemail: Don't delete mailbox state unless mailbox is deleted" into 13	7 years ago
George Joseph	1036959a49	Merge "res_rtp_asterisk: Remove some unused structure fields." into 13	7 years ago
George Joseph	e137086163	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	ead94ad621	Merge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set" into 13	7 years ago
Sean Bright	a2336adfb9	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
Friendly Automation	4cfc25d4e6	Merge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels" into 13	7 years ago
Sean Bright	560747dca4	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
Friendly Automation	c01a770fc0	Merge "confbridge: announce to the marked users when they join an empty conference" into 13	7 years ago
Sean Bright	92cc094d4e	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	70a2f51786	Merge "utils: Don't set or clear flags that don't need setting or clearing" into 13	7 years ago
Friendly Automation	ac5c4316ef	Merge "stasis: Add statistics gathering in developer mode." into 13	7 years ago
Friendly Automation	212f2a3aa3	Merge "Use non-blocking socket() and pipe() wrappers" into 13	7 years ago
Joshua C. Colp	e5605232df	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
Friendly Automation	a03202e551	Merge "stasis: Allow filtering by formatter" into 13	7 years ago
Joshua C. Colp	e557a72566	Merge "build: Update config.guess and config.sub" into 13	7 years ago
George Joseph	6109126e2d	Merge "pjproject_bundled: check whether UPDATE is supported on outgoing calls" into 13	7 years ago
Friendly Automation	53bda83f5e	Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"" into 13	7 years ago
Sean Bright	b84fd37ef0	Use non-blocking socket() and pipe() wrappers ... Change-Id: I050ceffe5a133d5add2dab46687209813d58f597	7 years ago
George Joseph	232a56d0af	Merge "CI: Various updates to buildAsterisk.sh" into 13	7 years ago
Sean Bright	3f3035680b	utils: Don't set or clear flags that don't need setting or clearing ... Change-Id: I0e7fb507ac09b15e45e1ff8501ecfca67afa5217	7 years ago