4966 Commits (6cc20174c04f1235d71dc14a88b26f1615a88f11)

Author	SHA1	Message	Date
Joshua Colp	134e2f0ddc	module: Remove deprecated modules and update support levels. ... I have removed the STATIC_BUILD option immediately as it has not been maintained in many years and is non-functional. ASTERISK-27965 Change-Id: I64783d017b86dba9ee3c7bcfb97e59889a3f76d7	7 years ago
George Joseph	34f3fe9552	app_confbridge: Use the SDP 'label' attribute to correlate users ... Previously, the msid "label" attribute was used to correlate participant info but because streams could be reused, the msid wasn't being updated correctly when someone left the bridge and another joined. Now, instead of looking for the msid attribute on a channel's streams, app_confbridge sets an "SDP:LABEL" attribute on the stream which res_pjsip_sdp_rtp looks for. If it finds it, it adds a "label" attribute to the current sdp. Change-Id: I6cbaa87fb59a2e0688d956e72d2d09e4ac20d5a5	7 years ago
Robert Mordec	9d3f3a4b0a	app_confbridge: Bridge and announcers not removed if conference ends quickly ... If a conference is ended very quickly after it was created (i.e., the first user immediately hangs up) then the conference bridge and announcer channels are not removed. When a conference is created, the push_announcer() function is added to the playback queue task processor and the conference object reference is bumped. If a conference is ended while the push_announcer() function is still going then the ao2_cleanup(conference) at the end of push_announcer() will call the destructor function - destroy_conference_bridge(). The destroy_conference_bridge() function will then add the hangup_playback() task to the playback queue and will wait for it to end. Since it is already a current task of the playback queue it will wait forever. This patch makes the conference thread call push_announcer() directly. This way the conference object reference bump is not needed. Since the playback queue task processor is only used by the conference thread itself, there is no danger of trying to play announcements before the announcer is pushed to the bridge. ASTERISK-27870 #close Change-Id: I947a50fb121422d90fd1816d643a54d75185a477	7 years ago
Joshua Colp	5f517bacd0	Merge "app_confbridge: Move participant info code to confbridge_manager."	7 years ago
George Joseph	5f12e2bd07	app_confbridge: Move participant info code to confbridge_manager. ... With the participant info code in app_confbridge, we were still in the process of adding the channel to the bridge when trying to send an in-dialog MESSAGE. This caused 2 threads to grab the channel blocking flag at the same time. To mitigate this, the participant info code was moved to confbridge_manager so it runs after all channel/bridge actions have finished. Change-Id: I228806ac153074f45e0b35d5236166e92e132abd	7 years ago
Kristian F. Høgh	184b375b41	app_queue: Add option for predial handlers on caller and callee channels ... Add predial handler support to app_queue. app_dial (ASTERISK_19548) and app_originate (ASTERISK_26587) have the ability to execute predial handlers on caller and callee channels. This patch adds predial handlers to app_queue and uses the same options as Dial and Originate (b and B). The caller routine gets executed when the caller first enters the queue. The callee routine gets executed for each queue member when they are about to be called. ASTERISK-27912 Change-Id: I5acf5c32587ee008658d12e8a8049eb8fa4d0f24	7 years ago
Richard Mudgett	7d874c1af7	AMI PlayDTMF Action: Make not compete with channel's media thread. ... There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	7 years ago
Jenkins2	986cffa18e	Merge "app_confbridge: Enable sending events to participants"	7 years ago
Sam Wierema	4c7ab73468	app_mp3: remove 10 seconds of silence after mp3 playback ... This patch changes the way asterisk polls output from mpg123, instead of waiting for 10 seconds(when playing an http url) it now uses a timeout of one second and iterates 10 times using this same timeout. The main difference is that for every timeout asterisk receives it now checks if mpg123 is still running before poll again. ASTERISK-27752 Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620	7 years ago
George Joseph	e7a7506f9c	app_confbridge: Enable sending events to participants ... ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e	7 years ago
George Joseph	49196c3a7b	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response"	7 years ago
Joshua Colp	8b08a8437e	Merge "app_meetme: Fix manager event documentation for several events."	7 years ago
George Joseph	437ab41881	app_sendtext: Allow content types other than text/plain ... There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9	7 years ago
William McCall	a7f4121238	app_confbridge: Add talking indicator for ConfBridgeList AMI response ... When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	7 years ago
Richard Mudgett	6bbede84fb	app_meetme: Fix manager event documentation for several events. ... The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and MeetmeTalkRequest AMI events were documented with sending out a Usernum header when the User header was actually output. * Change the online documentation to match reality. ASTERISK-27873 ASTERISK-25261 Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39	7 years ago
Joshua Colp	f5c1e74524	Merge "app_queue: Update year Copyright and fix missing tabs in documentation"	7 years ago
Rodrigo Ramírez Norambuena	d402594f74	app_queue: Update year Copyright and fix missing tabs in documentation ... Change-Id: Ieb8faf37dc765463ee5dbca1d1343242c756b1c7	7 years ago
Joshua Colp	21dd609e77	Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database"	7 years ago
Nic Colledge	2ca3b6d9cc	app_voicemail: Fix data-type mismatch between app_voicemail and database ... Fix data-type mismatch between app_voicemail and database columns exposed by new version of MariaDB ASTERISK-27760 Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b	7 years ago
Nic Colledge	97f20fe5ed	app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail ... Correct the log warning message shown when ODBC voicemail retrieve_file is called and there is a null value in the category column. A more meaningfull message is now written at debug level. ASTERISK-27853 Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4	7 years ago
Corey Farrell	b5914d90ac	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
Corey Farrell	d855658f23	app_macro: Prevent infinite loop in find_matching_priority. ... Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will prevent crash if dialplan contains an include loop. Log an error when this occurs, at most one message per call to Macro() so we avoid logger spam. ASTERISK-26570 #close Change-Id: I6c71b76998c31434391b150de055ae9a531e31da	7 years ago
George Joseph	5dd6fe478c	Merge "app_sendtext: Enhance SendText to support Enhanced Messaging"	7 years ago
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	7 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	7 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	7 years ago
Joshua Colp	d6e1acd25e	Merge "app_confbridge / bridge_softmix: Add ability to configure REMB interval."	7 years ago
Joshua Colp	df3db2f146	Merge "app_originate: Add async option."	7 years ago
Richard Mudgett	6774913e82	app_agent_pool.c: Fix off nominal ref leak. ... Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6	7 years ago
Joshua Colp	0f6431e8e4	app_confbridge / bridge_softmix: Add ability to configure REMB interval. ... This change adds a configuration option to app_confbridge which can be used to set the interval at which we will send a combined REMB (remote estimated maximum bitrate) frame to sources of video. The bridging API has also been extended slightly to allow setting this so bridge_softmix can use it. ASTERISK-27786 Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82	7 years ago
Russell Bryant	75715b95b4	app_originate: Add async option. ... Add an option to make app_originate not wait for the created channel to answer. Change-Id: I7fc2facd77079abc6321f44e8bcd4e39298de2ae Requested-by: Frederic Steinfels <fst@highdefinition.ch> Signed-off-by: Russell Bryant <russell@russellbryant.net>	7 years ago
Joshua Colp	0bfc83ee4f	Merge "BuildSystem: Remove unused dependency on libltdl."	7 years ago
Jenkins2	5843a19797	Merge "loader: Convert reload_classes to built-in modules."	7 years ago
George Joseph	c29b5389c5	Merge "app_voicemail: Fix json blob errors"	7 years ago
George Joseph	d6a49afb56	Merge "app_dial: Enable early-media video"	7 years ago
Alexander Traud	10a978829e	BuildSystem: Remove unused dependency on libltdl. ... Asterisk does not need the development package of libltdl, because it does not use any symbol of -lltdl directly. Instead, it uses the runtime package via the shared library -lodbc. On the supported platforms, that shared library declares its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have failed. ASTERISK-27745 Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba	7 years ago
Florian Floimair	ecc846b26b	app_dial: Enable early-media video ... Certain applications (e.g. door-phone) require that also video is transmitted before a call is accepted. Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e	7 years ago
George Joseph	4d1e3fef6b	app_voicemail: Fix json blob errors ... When app_voicemail calls ast_test_suite_notify with the results of a user keypress, it formats the keypress as '%c'. If the user hung up or some other error occurrs, the result of the keypress is a non printable character. This ultimately causes json_vpack_ex to think it's being passed a non utf-8 string and return an error. * Keypress results passed to ast_test_suite_notify are now checked with isprint() and a '?' is substituted if the check fails. Change-Id: I78ee188916bbac840f3d03f40201b692347ea865	7 years ago
Corey Farrell	572a508ef2	loader: Convert reload_classes to built-in modules. ... * acl (named_acl.c) * cdr * cel * ccss * dnsmgr * dsp * enum * extconfig (config.c) * features * http * indications * logger * manager * plc * sounds * udptl These modules are now loaded at appropriate time by the module loader. Unlike loadable modules these use AST_MODULE_LOAD_FAILURE on error so the module loader will abort startup on failure of these modules. Some of these modules are still initialized or shutdown from outside the module loader. logger.c is initialized very early and shutdown very late, manager.c is initialized by the module loader but is shutdown by the Asterisk core (too much uses it without holding references). Change-Id: I371a9a45064f20026c492623ea8062d02a1ab97f	7 years ago
Jenkins2	1485719531	Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro."	7 years ago
Jenkins2	ffdd4b1c39	Merge "Complete deprecating legacy modules."	7 years ago
Jenkins2	4ef5b58230	Merge "app_osplookup: Move header defines into the app."	7 years ago
Kevin Harwell	257be6847c	Merge "voicemail: Fixed wrong voicemail message count"	7 years ago
Richard Mudgett	e6738b79b3	Complete deprecating legacy modules. ... The menuselect comment was updated to deprecate these modules but the AST_MODULE_INFO block at the end of file was missed. ASTERISK-27671 Change-Id: I63070b5c4d4f08af010c6034acd4793c1bcef839	7 years ago
Corey Farrell	c8a521b6c8	Replace direct checks of option_debug with DEBUG_ATLEAST macro. ... Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0	7 years ago
Sungtae Kim	961dd9fe52	voicemail: Fixed wrong voicemail message count ... Fixed wrong voicemail mailbox reference for Action: VoicemailUsersList. ASTERISK-27703 Change-Id: Ie6578ad80bba2bfaf34b84f0be978f59045ce6cd	7 years ago
Corey Farrell	add03e207c	app_osplookup: Move header defines into the app. ... astosp.h is leftover from when logic was split between app_osplookup and res_osp. All logic was moved into app_osplookup by 109737eb1c in 2006, but astosp.h remained. This moves the remaining defines into app_osplookup and deletes astosp.h. Change-Id: I0a6c4debd7c9543b608520b1765abfa4fab7b2fd	7 years ago
Michael Cargile	7b01236028	apps/app_amd.c: Fixed total time and silence calculations ... Between Asterisk 11 and Asterisk 13 there was a significant increase in the number of AST_FRAME_NULL frames being processed by app_amd.c's main loop. Each AST_FRAME_NULL frame was being counted as 100ms towards the total time and silence. This may have been accurate when app_amd.c was orginally added, but it is not in Asterisk 13. As such the total analysis time and silence calculations were way off effectively breaking app_amd.c * Additional debug messages were added * AST_FRAME_NULL are now ignored ASTERISK-27610 Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9	7 years ago
Corey Farrell	9f74afbdcf	Deprecate legacy modules. ... * app_fax (replaced by res_fax). * res_config_sqlite (replaced by res_config_sqlite3). * res_monitor (replaced by app_mixmonitor). This is related to ASTERISK~23657 but does not resolve that ticket. Resolving that ticket would require complete removal of res_monitor. ASTERISK-27671 #close Change-Id: I16a3edd61fc1abd4a7b2e9357693ed663f62dd49	7 years ago
Richard Mudgett	67cd90f10d	app_confbridge: ConfbridgeList event has standard channel shapshot headers. ... * Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977	7 years ago