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Author	SHA1	Message	Date
Chris-Savinovich	b437656c2e	"confbridge show profile bridge" does not output "sfu" when video_mode is sfu ... Fixes a bug on the "confbridge show profile bridge" cli command that showed "video_mode=no video" when video_mode was set to "sfu" ASTERISK-27418 #close Change-Id: I481e3172c7f872664c7ac7809879d541c9f031e9	8 years ago
Corey Farrell	179ae87cf4	Build System: Add missing ASTMM_LIBC to flex output. ... Redirect libc allocation functions to use Asterisk functions for main/ast_expr2f.c and res/ael/ael_lex.c. This will resolve errors produced by astmm.h when these files are regenerated, though other issues still remain. ASTERISK~27813 Change-Id: I7263e9e4217a17bde4ffaa2087a8f8aeb2a8588c	8 years ago
George Joseph	af39255052	Merge "bridge_softmix / app_confbridge: Add support for REMB combining."	8 years ago
Jenkins2	0bea1da3ef	Merge "utils/pval: Add -lBlocksRuntime for compiler clang conditionally."	8 years ago
Jenkins2	e63d40aa78	Merge "utils: Add ast_assert_return"	8 years ago
Sean Bright	80e6952013	format_pcm: Correct behavior of fseek and ftell for G.722 ... There are twice as many samples in the same number of bytes, so redefine some of the G.722 format functions in terms of their PCM counterparts. Change-Id: I6a8c7352624b930a5f2d9e4857f75283fa5dd9f9	8 years ago
Alexander Anikin	95e8450194	chan_ooh323: introduce localras config parameter ... Introduce localras parameter that specify source IP for connecting to Gatekeeper. Useful for multihome configurations. ASTERISK-25129 #close Reported by: Dmitry Melekhov Tested by: Dmitry Melekhov Change-Id: I0b604b01793f3e02a776502659e07cd3fc7e3097	8 years ago
Alexander Anikin	446320f1d4	chan_ooh323: Fix cppcheck warnings ... Fix cppcheck warnings about redundant conditions and possible null pointer usage ASTERISK-27793 #close Reported by: Ilya Shipitsin Tested by: Ilya Shipitsin Change-Id: I0b31933b062a23331dbac9a82b8bcfe345f406f6	8 years ago
Joshua Colp	8de3fa2b56	bridge_softmix / app_confbridge: Add support for REMB combining. ... This change adds the ability for multiple REMB reports in bridge_softmix to be combined according to a configured behavior into a single report. This single report is sent back to the sender of video, which adjusts the encoding bitrate to be at or below the bitrate of the report. The available behaviors are: lowest, highest, and average. Lowest uses the lowest received bitrate. Highest uses the highest received bitrate. Average goes through the received bitrates adding them to the previous average and creates a new average. Other behaviors can be added in the future and the existing average one may be adjusted, but this provides the foundation to do so. Support for configuring which behavior to use has been added to app_confbridge. ASTERISK-27804 Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66	8 years ago
Jenkins2	3255a286b3	Merge "res_musiconhold: Don't restart MOH from beginning after announcement."	8 years ago
George Joseph	f79a372941	streams: Add string metadata capability ... Replaces the never used opaque data array. Updated stream tests to include get/set metadata and stream clone with metadata. Added stream metadata dump to "core show channel" Change-Id: Id7473aa4b374d7ab53046c20e321037ba9a56863	8 years ago
George Joseph	f7e7ce6ba2	utils: Add ast_assert_return ... Similar to pjproject's PJ_ASSERT_RETURN macro, this one will do the following... If the assert passes... NoOp If the assert fails and AST_DEVMODE is defined, execute ast_assert() then, if DO_CRASH isn't set, return from the calling function with the supplied value. If the assert fails and AST_DEVMODE is not defined, return from the calling function with the supplied value. The macro will execute a return without a value if one isn't suppled. Change-Id: I0003844affeab550d5ff5bca7aa7cf8a559b873e	8 years ago
George Joseph	8135558bab	app_sendtext: Enhance SendText to support Enhanced Messaging ... SendText now accepts new channel variables that can be used to override the To and From display names and set the Content-Type of a message. Since you can now set Content-Type, other text/* content types are now valid. Change-Id: I648b4574478119f95de09d9f08e9595831b02830	8 years ago
George Joseph	4fb7967c73	bridge_softmix: Forward TEXT frames ... Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
Alexander Traud	8a1ffb050b	utils/pval: Add -lBlocksRuntime for compiler clang conditionally. ... ASTERISK-27809 Change-Id: I930b364a33d54cc08dedfcd5bb45f7e83242f134	8 years ago
Alexander Traud	3d9345e3ae	chan_vpb: Avoid GNU old-style field designator extension. ... clang 6.0 warned about this. Beside that, this change removes the used variable 'desc'. ASTERISK-27808 Change-Id: Ia26bdcc0a562c058151814511cfcf70ecafa595b	8 years ago
Ben Ford	f5d5083ea7	res_rtp_asterisk: Add support for receiving and handling NACK requests. ... Adds the ability to receive and handle incoming NACK requests if retransmissions are enabled. If retransmissions are enabled, a data buffer is allocated that stores packets being sent. If a NACK request is received, the packet requested for retransmission is sent if it is still in the buffer. In the same request, if any of the following 16 packets are marked as not received, those will be sent as well if available, as outlined in RFC4585. Also changes RTCP RR and SR to use media source SSRC instead of packet source SSRC when determining which instance to use for RTCP reports. For more information, refer to the wiki page: https://wiki.asterisk.org/wiki/display/AST/WebRTC+User+Experience+Improvements ASTERISK-27806 #close Change-Id: I7f7f124af3b9d5d2fd9cffc6ba8cb48a6fff06ec	8 years ago
Richard Mudgett	d50d637764	stringfields: Collect extended stringfields into the stringfield section. ... Use of extended stringfields is a temporary mechanism to avoid ABI breakage in released branches without resorting to more inconvienient methods. * Collect existing extended stringfields into the parent stringfield section of the struct. Change-Id: I8d46d037801b4518837c3ea4b6df95ceadc9436b	8 years ago
George Joseph	38dae51b78	Merge "res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations."	8 years ago
Jenkins2	267e007c28	Merge "pjsip_scheduler.c: Add ability to trace scheduled tasks."	8 years ago
Jenkins2	1f6fc78a2e	Merge "pjsip_scheduler.c: Fix some corner cases."	8 years ago
Ben Ford	4aeec6100f	res_musiconhold: Don't restart MOH from beginning after announcement. ... This reverts a problem introduced by the fix for ASTERISK_24329. Now, when an announcement is played while waiting in a queue, music on hold will not restart from the beginning of the sound file and will instead pick up where it left off. However, the incorrect behavior in ASTERISK_24329 is now present again; if an announcement X seconds long is played when music on hold starts, music on hold will start X seconds into the file. ASTERISK-27774 #close Reported by: lvl Change-Id: I86b2885ee7063268f9b9747eddb788336ade989b	8 years ago
Richard Mudgett	3bb6cf43b5	pjsip_scheduler.c: Add ability to trace scheduled tasks. ... When a scheduled task is created you can pass in the AST_SIP_SCHED_TASK_TRACK flag. This new flag causes scheduling events to be logged. Change-Id: I91967eb3d5a220915ce86881a28af772f9a7f56b	8 years ago
Richard Mudgett	237d341bbd	res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations. ... ast_sip_push_task_synchronous() did not necessarily execute the passed in task under the specified serializer. If the current thread is any registered pjsip thread then it would execute the task immediately instead of under the specified serializer. Reentrancy issues could result if the task does not execute with the right serializer. The original reason ast_sip_push_task_synchronous() checked to see if the current thread was a registered pjsip thread was because of a deadlock with masquerades and the channel technology's fixup callback (ASTERISK_22936). A subsequent masquerade deadlock fix (ASTERISK_24356) involving call pickups avoided the original deadlock situation entirely. The PJSIP channel technology's fixup callback no longer needed to call ast_sip_push_task_synchronous(). However, there are a few places where this unexpected behavior is still required to avoid deadlocks. The pjsip monitor thread executes callbacks that do calls to ast_sip_push_task_synchronous() that would deadlock if the task were actually pushed to the specified serializer. I ran into one dealing with the pubsub subscriptions where an ao2 destructor called ast_sip_push_task_synchronous(). * Split ast_sip_push_task_synchronous() into ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer(). ast_sip_push_task_wait_servant() has the old behavior of ast_sip_push_task_synchronous(). ast_sip_push_task_wait_serializer() has the new behavior where the task is always executed by the specified serializer or a picked serializer if one is not passed in. Both functions behave the same if the current thread is not a SIP servant. * Redirected ast_sip_push_task_synchronous() to ast_sip_push_task_wait_servant() to preserve API for released branches. ASTERISK_26806 Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3	8 years ago
Richard Mudgett	c2f85e881d	pjsip_scheduler.c: Fix some corner cases. ... * Fix the periodic interval wander because it may take significant time between the sched thread queueing the task in the serializer and the serializer actually executing the task. The time it takes to actually execute the task was already taken into account. * Pass a schtd ref to the serializer when we queue a scheduled task on the serializer. We don't want it going away on us while it is in the serializer queue. * Skip the scheduled task if the task was canceled between queueing the task to the serializer and the serializer actually executing the task. * Reorder struct ast_sip_sched_task to avoid unnecessary padding. Removed task_id and added next_periodic. * Hold a ref to the passed in serializer so the serializer cannot go away on the scheduled task. ASTERISK_26806 Change-Id: I6c8046b75f6953792c8c30e55b836a4291143f24	8 years ago
Richard Mudgett	96c4a57edf	pjsip_scheduler.c: Sort "pjsip show scheduled_tasks" output. ... * A side benefit is that the scheduled tasks are not completely blocked while the CLI command executes. * Adjusted the "Task Name" column width to have more room for longer names. Change-Id: Iec64aa463ee8b10eef90120e00c38b1fb444087e	8 years ago
Jenkins2	3c5d76863b	Merge "res_pjsip_notify.c: enable in-dialog NOTIFY"	8 years ago
Jenkins2	7777326244	Merge "pjsip_scheduler.c: Fix ao2 usage errors."	8 years ago
Jenkins2	2652ae1471	Merge "Build System: Strip '-std=c99' from CFLAGS provided by libraries."	8 years ago
Evandro Cesar Arruda	429c758e48	cdr_mysql: Compile error because MYSQL_PORT definition is missing ... If it is not defined, it will add MYSQL_PORT definition. After some research on MySQL/MariaDB development tree, I couldn't find any reference to MYSQL_PORT definition in include files. ASTERISK-27782 #close Change-Id: Ieee56c836fc2e8bd021c456145bba04c6068bb77	8 years ago
Chris-Savinovich	0747ac893b	res_pjsip_session: Rewrite o= with external_media_address. ... It now appends the external IP address on the o= line of the SDP packet. The decision was made to write the numeric IP address as opposed to the RFC that states the FQDN should be used if and when available. We believe the usage of literal IP address will help avoid potential problems. ASTERISK-27614 #close Change-Id: I84f3360f3606b8c4e8d161edb228799ec0b8a302	8 years ago
Nathan Bruning	1cd704de36	res_pjsip_notify.c: enable in-dialog NOTIFY ... This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29	8 years ago
Jenkins2	fabfe701bb	Merge "res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge"	8 years ago
George Joseph	8af759c088	Merge "chan_sip.c: Fix INVITE with replaces channel ref leak."	8 years ago
Richard Mudgett	7157dcf83b	pjsip_scheduler.c: Fix ao2 usage errors. ... * Removed several invalid uses of OBJ_NOLOCK. These uses resulted in the 'tasks' container being accessed without a lock in a multi-threaded environment. A recipe for crashes. * Removed needlessly obtaining schtd object references. If the caller providing you a pointer to an object doesn't have a valid reference then you cannot safely get one from it. * Getting a ref to 'tasks' when you aren't copying the pointer into another location is useless. The 'tasks' container pointer is global. * Removed many unnecessary uses of RAII_VAR. * Make ast_sip_schedule_task() name parameter const. ASTERISK_26806 Change-Id: I5c62488e651314e2a1dbc01f5b078a15512d73db	8 years ago
Jenkins2	2a6072a9c4	Merge "pjsip / res_rtp_asterisk: Add support for sending REMB"	8 years ago
Joshua Colp	2e60196265	Merge "res_rtp_asterisk: Fix minimum block word length for REMB."	8 years ago
Joshua Colp	d6e1acd25e	Merge "app_confbridge / bridge_softmix: Add ability to configure REMB interval."	8 years ago
Joshua Colp	0c56e3d3eb	Merge "Build System: Fixes for configure script."	8 years ago
Joshua Colp	df3db2f146	Merge "app_originate: Add async option."	8 years ago
Jenkins2	070428415a	Merge "res_rtp_asterisk: Queue video update on picture loss indication."	8 years ago
Corey Farrell	879e592baf	Build System: Enable python3 compatibility. ... * Consistently use spaces in rest-api-templates/asterisk_processor.py. * Exclude third-party from docs/full-en_US.xml. * Add docs/full-en_US.xml to .gitignore. * Use list() to convert python3 view. * Use python3 print function. * Replace cmp() with equivalent equation. * Replace reference to out of scope subtype variable with name parameter. * Use unescaping triple bracket notation in mustache templates where needed. This causes behavior of Python2 to be maintained when using Python3. * Fix references to has_websocket / is_websocket in res_ari_resource.c.mustache. * Update calculation of has_websocket to use any(). * Use unicode mode for writing output file in transform.py. * Replace 'from swagger_model import *' with explicit import of required symbols. I have not tested spandspflow2pcap.py or voicemailpwcheck.py, only the print syntax has been fixed. Change-Id: If5c5b556a2800d41a3e2cfef080ac2e151178c33	8 years ago
Richard Mudgett	0c03eab962	res_pjsip_refer/chan_sip: Fix INVITE with replaces transfer to ConfBridge ... There is a problem when an INVITE-with-Replaces transfer targets a channel in a ConfBridge. The transfer will unconditionally swap out the ConfBridge channel. Unfortunately, the ConfBridge state will not be aware of this change. Unexpected behavior will happen as a result since ConfBridge channels currently can only be replaced by a masquerade and not normal bridge channel moves. * We just need to pretend that the channel isn't in a bridge (like other transfer methods already do) so the transfer channel will masquerade into the ConfBridge channel. Change-Id: I209beb0e748fa4f4b92a576f36afa8f495ba4c82	8 years ago
Joshua Colp	c7bd554094	pjsip / res_rtp_asterisk: Add support for sending REMB ... This change allows chan_pjsip to be given an AST_FRAME_RTCP containing REMB feedback and pass it to res_rtp_asterisk. Once res_rtp_asterisk receives the frame a REMB RTCP feedback packet is constructed with the appropriate contents and sent to the remote endpoint. ASTERISK-27776 Change-Id: Ic53f821c1560d8924907ad82c4d9c0bc322b38cd	8 years ago
Jenkins2	72a8e2106e	Merge "res_pjsip: Update authenticate_qualify documentation."	8 years ago
Joshua Colp	39016e3582	res_rtp_asterisk: Fix minimum block word length for REMB. ... The minimum block word length is actually 4, not 5. Change-Id: I878542218225aed72c72bdf1b856fc822cd2d649	8 years ago
Joshua Colp	8a602f18db	res_rtp_asterisk: Queue video update on picture loss indication. ... The previous payload specific feedback handling was very single minded in that it just assumed everything should trigger a video update. This was changed but the handling of picture loss indication was not added. The result was that video may not flow. This change adds it explicitly in. Change-Id: I1894be02e39ee10a0af841b5a1dca5f0ec7d60b6	8 years ago
Richard Mudgett	d72a2966da	chan_sip.c: Fix INVITE with replaces channel ref leak. ... Given the below call scenario: A -> Ast1 -> B C <- Ast2 <- B 1) A calls B through Ast1 2) B calls C through Ast2 3) B transfers A to C When party B transfers A to C, B sends a REFER to Ast1 causing Ast1 to send an INVITE with replaces to Ast2. Ast2 then leaks a channel ref of the channel between Ast1 and Ast2. Channel ref leaks are easily seen in the CLI "core show channels" output. The leaked channels appear in the output but you can do nothing with them and they never go away unless you restart Asterisk. * Properly account for the channel refs when imparting a channel into a bridge when handling an INVITE with replaces in handle_invite_replaces(). The ast_bridge_impart() function steals a channel ref but the code didn't account for how many refs were held by the code at the time and which ref was stolen. * Eliminated RAII_VAR in handle_invite_replaces(). ASTERISK-27740 Change-Id: I7edbed774314b55acf0067b2762bfe984ecaa9a4	8 years ago
Richard Mudgett	71a67a98c4	res_pjsip: Update authenticate_qualify documentation. ... Change-Id: I3811de0014b1ffe96d4a3b49cddd5d4ca02ee5d4	8 years ago
Richard Mudgett	6774913e82	app_agent_pool.c: Fix off nominal ref leak. ... Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6	8 years ago