1155 Commits (a2f6028a516cba8f7a95f2f286add0555fac8a67)

Author	SHA1	Message	Date
George Joseph	b7a875778a	res_musiconhold: Add kill_escalation_delay, kill_method to class ... By default, when res_musiconhold reloads or unloads, it sends a HUP signal to custom applications (and all descendants), waits 100ms, then sends a TERM signal, waits 100ms, then finally sends a KILL signal. An application which is interacting with an external device and/or spawns children of its own may not be able to exit cleanly in the default times, expecially if sent a KILL signal, or if it's children are getting signals directly from res_musiconhoild. * To allow extra time, the 'kill_escalation_delay' class option can be used to set the number of milliseconds res_musiconhold waits before escalating kill signals, with the default being the current 100ms. * To control to whom the signals are sent, the "kill_method" class option can be set to "process_group" (the default, existing behavior), which sends signals to the application and its descendants directly, or "process" which sends signals only to the application itself. Change-Id: Iff70a1a9405685a9021a68416830c0db5158603b	9 years ago
Jenkins2	33aa3907eb	Merge "pjproject_bundled: Allow passing configure options to bundled"	9 years ago
Jenkins2	e1c0e14fac	Merge "res_pjsip: Add DTMF INFO Failback mode"	9 years ago
George Joseph	f573e599c0	pjproject_bundled: Allow passing configure options to bundled ... There wasn't any good way to pass options like --host or --build down to the pjproject configure which makes cross-compiling difficult. * Added a new PJPROJECT_CONFIGURE_OPTS environment variable which can be used to pass arbitrary options to pjproject configure. * Automatically set the pjproject configure --host and --build options to match those supplied for the asterisk configure. ASTERISK-27097 #close Reported-by: Kinsey Moore Change-Id: I5fa776e110262851173002a26ffe1172e4c35b2e	9 years ago
Torrey Searle	fb7247c57c	res_pjsip: Add DTMF INFO Failback mode ... The existing auto dtmf mode reverts to inband if 4733 fails to be negotiated. This patch adds a new mode auto_info which will switch to INFO instead of inband if 4733 is not available. ASTERISK-27066 #close Change-Id: Id185b11e84afd9191a2f269e8443019047765e91	9 years ago
Alexei Gradinari	0cef7b9d4e	app_voicemail: IMAP connection control ... A new global option "imap_poll_logout" was added to specify whether need to disconnect from the IMAP server after polling of mailboxes. ASTERISK-27068 #close Closing IMAP connection after loading mailbox from voicemail.conf ASTERISK-24052 #close Change-Id: Ib7558ba04516240a32b65f42e9be64372a0ae12a	9 years ago
Alexei Gradinari	7a46309d3d	res_pjsip: New endpoint option "notify_early_inuse_ringing" ... This option was added to control whether to notify dialog-info state 'early' or 'confirmed' on Ringing when already INUSE. The value "yes" is useful for some SIP phones (Cisco SPA) to be able to indicate and pick up ringing devices. ASTERISK-26919 #close Change-Id: Ie050bc30023543c7dfb4365c5be3ce58c738c711	9 years ago
Jenkins2	abe1dd3039	Merge "pjsip: Extend 'asymmetric_rtp_codec' option to include us changing."	9 years ago
Alexei Gradinari	19da99df2f	CHANGES: correct version for a new option 'refer_blind_progress' ... Change-Id: If4817d26a8974610827624fb8a4e56d681d6bf97	9 years ago
Joshua Colp	d3e951edf5	pjsip: Extend 'asymmetric_rtp_codec' option to include us changing. ... PJSIP support in Asterisk differs from chan_sip in that it allows media to be sent as-is without transcoding provided the codecs were negotiated in the SDP. This is allowed according to the RFC. Support for this differs quite a lot though and some endpoints do not handle it well. This change extends the 'asymmetric_rtp_codec' option to also cover this case. When set to no (the default) the code behaves as chan_sip does - the best codec is selected and we will only ever send that, unless we change what we are sending if the remote side changes. When set to yes we will send media as-is without transcoding if the codec has been negotiated in the SDP. ASTERISK-26996 Change-Id: Ib1647f6902a0843e8c435946f831c2159e8d1d51	9 years ago
Joshua Colp	861984eac0	res_pjsip: Add support for returning only reachable contacts and use it. ... This introduces the ability for PJSIP code to specify filtering flags when retrieving PJSIP contacts. The first flag for use causes the query code to only retrieve contacts that are not unreachable. This change has been leveraged by both the Dial() process and the PJSIP_DIAL_CONTACTS dialplan function so they will now only attempt calls to contacts which are not unreachable. ASTERISK-26281 Change-Id: I8233b4faa21ba3db114f5a42e946e4b191446f6c	9 years ago
Jenkins2	48d047ad5a	Merge "res_pjsip: New endpoint option "refer_blind_progress""	9 years ago
Sean Bright	d847fe6585	res_agi: Allow configuration of audio format of EAGI pipe ... This change allows the format of the EAGI audio pipe to be changed by setting the dialplan variable 'EAGI_AUDIO_FORMAT' to the name of one of the loaded formats. ASTERISK-26124 #close Change-Id: I7a10fad401ad2a21c68c2e7246fa357d5cee5bbd	9 years ago
Jenkins2	57217e4cc2	Merge "res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages."	9 years ago
Alexei Gradinari	808f299808	res_pjsip: New endpoint option "refer_blind_progress" ... This option was added to turn off notifying the progress details on Blind Transfer. If this option is not set then the chan_pjsip will send NOTIFY "200 OK" immediately after "202 Accepted". Some SIP phones like Mitel/Aastra or Snom keep the line busy until receive "200 OK". ASTERISK-26333 #close Change-Id: Id606fbff2e02e967c02138457badc399144720f2	9 years ago
Joshua Colp	3c36c29c81	res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages. ... This change adds the required logic to allow the SIP Call-ID to be placed into the HEP RTCP traffic if the chan_sip module is used. In cases where the option is enabled but the channel is not either SIP or PJSIP then the code will fallback to the channel name as done previously. Based on the change on Nir's branch at: team/nirs/hep-chan-sip-support ASTERISK-26427 Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d	9 years ago
George Joseph	201346fb7d	logger: Added logger_queue_limit to the configuration options. ... All log messages go to a queue serviced by a single thread which does all the IO. This setting controls how big that queue can get (and therefore how much memory is allocated) before new messages are discarded. The default is 1000. Should something go bezerk and log tons of messages in a tight loop, this will prevent memory escalation. When the limit is reached, a WARNING is logged to that effect and messages are discarded until the queue is empty again. At that time another WARNING will be logged with the count of discarded messages. There's no "low water mark" for this queue because the logger thread empties the entire queue and processes it in 1 batch before going back and waiting on the queue again. Implementing a low water mark would mean additional locking as the thread processes each message and it's not worth it. A "test" was added to test_logger.c but since the outcome is non-deterministic, it's really just a cli command, not a unit test. Change-Id: Ib4520c95e1ca5325dbf584c7989ce391649836d1	9 years ago
Richard Mudgett	7312cbe803	res_rtp_asterisk.c: Add stun_blacklist option ... Added the stun_blacklist option to rtp.conf. Some multihomed servers have IP interfaces that cannot reach the STUN server specified by stunaddr. Blacklist those interface subnets from trying to send a STUN packet to find the external IP address. Attempting to send the STUN packet needlessly delays processing incoming and outgoing SIP INVITEs because we will wait for a response that can never come until we give up on the response. Multiple subnets may be listed. ASTERISK-26890 #close Change-Id: I3ff4f729e787f00c3e6e670fe6435acce38be342	9 years ago
Troy Bowman	fac5115c43	app_queue: Log reason for PAUSEALL/UNPAUSEALL ... We needed the reason for our reporting when agents pause/unpause all of their queues at once. This is a small, simple patch that adds a reason for PAUSEALL and UNPAUSEALL. I have been using it in production for years. ASTERISK-26920 #close Change-Id: Ifb3f0d1a0abd5194253d9794023546e1395baf3d	9 years ago
George Joseph	2fe52174de	res_pjsip_config_wizard: Add 2 new parameters to help with proxy config ... Two new parameters have been added to the pjsip config wizard. * Setting 'sends_line_with_registrations' to true will cause the wizard to skip the creation of an identify object to match incoming request to the endpoint and instead add the line and endpoint parameters to the outbound registration object. * Setting 'outbound_proxy' is a shortcut for adding individual endpoint/outbound_proxy, aor/outbound_proxy and registration/outbound_proxy parameters. Change-Id: I678e5f80765734c056620528a6d40d82736ceeb0 (cherry picked from commit a827892ff7) (cherry picked from commit 27344675be)	9 years ago
zuul	90634cc184	Merge "rtp_engine: allocate RTP dynamic payloads per session"	9 years ago
Joshua Colp	c1ab8ca74c	Merge "res_pjsip_session: Enable RFC3578 overlap dialing support."	9 years ago
Kevin Harwell	9b103e7bea	rtp_engine: allocate RTP dynamic payloads per session ... Dynamic payload types were statically defined in Asterisk. This unfortunately limited the number of dynamic payloads that could be registered. With this patch dynamic payload type numbers are now assigned dynamically and per RTP instance. However, in order to limit any issues where some clients expect the old statically defined value this patch makes it so the value Asterisk used to pre- designate is used for the dynamic assignment if available. An option, "rtp_use_dynamic", has also been added (can be set in asterisk.conf) that turns the new dynamic behavior on or off. When off it reverts back to using statically defined payload values. This option defaults to "yes" in Asterisk 15. ASTERISK-26515 #close patches: ASTERISK-26515.diff submitted by jcolp (license 5000 Change-Id: I7653465c5ebeaf968f1a1cc8f3f4f5c4321da7fc	9 years ago
Richard Begg	6b7697ed48	res_pjsip_session: Enable RFC3578 overlap dialing support. ... Support for RFC3578 overlap dialling (i.e. 484 Response to partially matched destinations) as currently provided by chan_sip is missing from res_pjsip. This patch adds a new endpoint attribute (allow_overlap) [defaults to yes] which when set to yes enables 484 responses to partial destination matches rather than the current 404. ASTERISK-26864 Change-Id: Iea444da3ee7c7d4f1fde1d01d138a3d7b0fe40f6	9 years ago
Richard Mudgett	8cb4f9cea1	CHANNEL(callid): Give dialplan access to the callid. ... * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	9 years ago
George Joseph	5013d8f5d3	res_pjsip: Symmetric transports ... A new transport parameter 'symmetric_transport' has been added. When a request from a dynamic contact comes in on a transport with this option set to 'yes', the transport name will be saved and used for subsequent outgoing requests like OPTIONS, NOTIFY and INVITE. It's saved as a contact uri parameter named 'x-ast-txp' and will display with the contact uri in CLI, AMI, and ARI output. On the outgoing request, if a transport wasn't explicitly set on the endpoint AND the request URI is not a hostname, the saved transport will be used and the 'x-ast-txp' parameter stripped from the outgoing packet. * config_transport was modified to accept and store the new parameter. * config_transport/transport_apply was updated to store the transport name in the pjsip_transport->info field using the pjsip_transport->pool on UDP transports. * A 'multihomed_on_rx_message' function was added to pjsip_message_ip_updater that, for incoming requests, retrieves the transport name from pjsip_transport->info and retrieves the transport. If transport->symmetric_transport is set, an 'x-ast-txp' uri parameter containing the transport name is added to the incoming Contact header. * An 'ast_sip_get_transport_name' function was added to res_pjsip. It takes an ast_sip_endpoint and a pjsip_sip_uri and returns a transport name if endpoint->transport is set or if there's an 'x-ast-txp' parameter on the uri and the uri host is an ipv4 or ipv6 address. Otherwise it returns NULL. * An 'ast_sip_dlg_set_transport' function was added to res_pjsip which takes an ast_sip_endpoint, a pjsip_dialog, and an optional pjsip_tpselector. It calls ast_sip_get_transport_name() and if a non-NULL is returned, sets the selector and sets the transport on the dialog. If a selector was passed in, it's updated. * res_pjsip/ast_sip_create_dialog_uac and ast_sip_create_dialog_uas were modified to call ast_sip_dlg_set_transport() instead of their original logic. * res_pjsip/create_out_of_dialog_request was modified to call ast_sip_get_transport_name() and pjsip_tx_data_set_transport() instead of its original logic. * Existing transport logic was removed from endpt_send_request since that can only be called after a create_out_of_dialog_request. * res_pjsip/ast_sip_create_rdata was converted to a wrapper around a new 'ast_sip_create_rdata_with_contact' function which allows a contact_uri to be specified in addition to the existing parameters. (See below) * res_pjsip_pubsub/internal_pjsip_evsub_send_request was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac and ast_sip_create_dialog_uas. * 'contact_uri' was added to subscription_persistence. This was necessary because although the parsed rdata contact header has the x-ast-txp parameter added (if appropriate), subscription_persistence_update stores the raw packet which doesn't have it. subscription_persistence_recreate was then updated to call ast_sip_create_rdata_with_contact with the persisted contact_uri so the recreated subscription has the correct transport info to send the NOTIFYs. * res_pjsip_session/internal_pjsip_inv_send_msg was eliminated since all it did was transport selection and that is now done in ast_sip_create_dialog_uac. * pjsip_message_ip_updater/multihomed_on_tx_message was updated to remove all traces of the x-ast-txp parameter from the outgoing headers. NOTE: This change does NOT modify the behavior of permanent contacts specified on an aor. To do so would require that the permanent contact's contact uri be updated with the x-ast-txp parameter and the aor sorcery object updated. If we need to persue this, we need to think about cloning permanent contacts into the same store as the dynamic ones on an aor load so they can be updated without disturbing the originally configured value. You CAN add the x-ast-txp parameter to a permanent contact's uri but it would be much simpler to just set endpoint->transport. Change-Id: I4ee1f51473da32ca54b877cd158523efcef9655f	9 years ago
Mark Michelson	10fa49e327	Add rtcp-mux support ... This commit adds support for RFC 5761: Multiplexing RTP Data and Control Packets on a Single Port. Specifically, it enables the feature when using chan_pjsip. A new option, "rtcp_mux" has been added to endpoint configuration in pjsip.conf. If set, then Asterisk will attempt to use rtcp-mux with whatever it communicates with. Asterisk follows the rules set forth in RFC 5761 with regards to falling back to standard RTCP behavior if the far end does not indicate support for rtcp-mux. The lion's share of the changes in this commit are in res_rtp_asterisk.c. This is because it was pretty much hard wired to have an RTP and an RTCP transport. The strategy used here is that when rtcp-mux is enabled, the current RTCP transport and its trappings (such as DTLS SSL session) are freed, and the RTCP session instead just mooches off the RTP session. This leads to a lot of specialized if statements throughout. ASTERISK-26732 #close Reported by Dan Jenkins Change-Id: If46a93ba1282418d2803e3fd7869374da8b77ab5	9 years ago
Matt Jordan	1475604eff	res_pjsip_endpoint_identifier_ip: Add an option to match requests by header ... This patch adds a new features to the endpoint identifier module, 'match_header'. When set, inbound requests are matched by a provided SIP header: value pair. This option works in conjunction with the existing 'match' configuration option, such that if any 'match*' attribute matches an inbound request, the request is associated with the specified endpoint. Since this module now identifies by more than just IP address, appropriate renaming of the module and/or variables can be done in a non-release branch. ASTERISK-26863 #close Change-Id: Icfc14835c962f92e35e67bbdb235cf0589de5453 (cherry picked from commit 30f52d79d7)	9 years ago
Joshua Colp	6113bb470b	Merge "pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel"	9 years ago
Sean Bright	bc2c66b594	pbx_spool: Set AST_OUTGOING_ATTEMPT variable on channel ... Set a variable on the channel that indicates which attempt number we are currently performing to allow for attempt-specific behavior. ASTERISK-26568 #close Reported by: Roman Shubovich Change-Id: Iacd7e8d43b0ed5b6cb021c62f41f1a1f5733dd89	9 years ago
Daniel Journo	60998371e3	app_voicemail: Cannot set fromstring on a per-mailbox basis ... * apps/app_voicemail.c fromstring field added to mailbox which will override the global fromstring if set. ASTERISK-24562 #close Change-Id: I5e90e3a1ec2b2d5340b49a0db825e4bbb158b2fe	9 years ago
Jørgen H	7922f26cb0	res_pjsip WebRTC/websockets: Fix usage of WS vs WSS. ... According to the RFC[1] WSS should only be used in the Via header for secure Websockets. * Use WSS in Via for secure transport. * Only register one transport with the WS name because it would be ambiguous. Outgoing requests may try to find the transport by name and pjproject only finds the first one registered. This may mess up unsecure websockets but the impact should be minimal. Firefox and Chrome do not support anything other than secure websockets anymore. * Added and updated some debug messages concerning websockets. * security_events.c: Relax case restriction when determining security transport type. * The res_pjsip_nat module has been updated to not touch the transport on Websocket originating messages. [1] https://tools.ietf.org/html/rfc7118 ASTERISK-26796 #close Change-Id: Ie3a0fb1a41101a4c1e49d875a8aa87b189e7ab12	9 years ago
George Joseph	22242fef5d	res_pjsip_outbound_registration: Subscribe to network change events ... Outbound registration now subscribes to network change events published by res_stun_monitor and refreshes all registrations when an event happens. The 'pjsip send (un)register' CLI commands were updated to accept '*all' as an argument to operate on all registrations. The 'PJSIP(Un)Register' AMI commands were also updated to accept '*all'. ASTERISK-26808 #close Change-Id: Iad58a9e0aa5d340477fca200bf293187a6ca5a25	9 years ago
Sean Bright	275f469a4d	app_voicemail: Allow 'Comedian Mail' branding to be overriden ... Original patch by John Covert, slight modifications by me. ASTERISK-17428 #close Reported by: John Covert Patches: app_voicemail.c.patch (license #5512) patch uploaded by John Covert Change-Id: Ic3361b0782e5a5397a19ab18eb8550923a9bd6a6	9 years ago
Sean Bright	662c9e69fa	app_record: Add option to prevent silence from being truncated ... When using Record() with the silence detection feature, the stream is written out to the given file. However, if only 'silence' is detected, this file is then truncated to the first second of the recording. This patch adds the 'u' option to Record() to override that behavior. ASTERISK-18286 #close Reported by: var Patches: app_record-1.8.7.1.diff (license #6184) patch uploaded by var Change-Id: Ia1cd163483235efe2db05e52f39054288553b957	9 years ago
George Joseph	6691606723	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f3)	10 years ago
Sebastian Gutierrez	8cc1cd5df7	app_queue: Add QueueUpdate application. ... Add an application that allows tracking outbound calls using app_queue. ASTERISK-19862 Change-Id: Ia0ab64aed934c25b2a25022adcc7c0624224346e	10 years ago
Joshua Colp	a7d856cd96	res_pjsip_endpoint_identifier_ip: Add support for SRV lookups. ... This change implements SRV support for the IP based endpoint identifier module. All possible addresses through SRV are looked up and added as matches. If no SRV records are available a fallback to normal host resolution is done. If an IP address is provided then no SRV lookup occurs. This is configured using the "srv_lookups" option on the identify section and defaults to "yes". ASTERISK-26693 Change-Id: I6b641e275bf96629320efa8b479737062aed82ac	10 years ago
Jonathan R. Rose	d96e350256	core/pbx: dialplan show - display filename/line# ... Adds the ability for extensions to be registered to include filename and line number so that dialplan show output can show the filename and line number of a config file responsible for generating a given extension. This only affects config modules that are written to use the new extension registering functions. In this patch, that only includes pbx_config, so extensions registered in extensions.conf and any included extension will be shown in this manner. Extensions registered in this manner will show the filename and line number *instead* of the registrar. ASTERISK-26658 #close Reported by: Jonathan R. Rose Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30	10 years ago
George Joseph	79b09b5f18	res_pjsip_registrar: AMI Add RegistrationInboundContactStatuses command ... The PJSIPShowRegistrationsInbound AMI command was just dumping out all AORs which was pretty useless and resource heavy since it had to get all endpoints, then all aors for each endpoint, then all contacts for each aor. PJSIPShowRegistrationInboundContactStatuses sends ContactStatusDetail events which meets the intended purpose of the other command and has significantly less overhead. Also, some additional fields that were added to Contact since the original creation of the ContactStatusDetail event have been added to the end of the event. For compatibility purposes, PJSIPShowRegistrationsInbound is left intact. ASTERISK-26644 #close Change-Id: I326f12c9ecb52bf37ba03f0748749de4da01490a	10 years ago
Joshua Colp	faf2194fab	Merge "app_originate: Add option to execute gosub prior to dial"	10 years ago
Richard Mudgett	4b3d3fc741	res_pjsip_outbound_registration.c: Filter redundant statsd reporting. ... Increasing the testsuite shutdown timeout before forcibly killing Asterisk allowed more events to be sent out. Some tests failed as a result. The tests/channels/pjsip/statsd/registrations failed because we now get the statsd events that a comment in the test configuration stated couldn't be intercepted. Unfortunately, we get a variable number of events because of internal status state transition races generating redundant statsd events. We were reporting redundant statsd PJSIP.registrations.state changes for internal state changes that equated to the same thing publicly. * Made update_client_state_status() filter out redundant statsd updates. ASTERISK-26527 Change-Id: If851c7d514bb530d9226e4941ba97dcf52000646	10 years ago
Richard Mudgett	1dfa11b65c	PJPROJECT logging: Made easier to get available logging levels. ... Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389	10 years ago
David Kerr	ddc951060a	app_originate: Add option to execute gosub prior to dial ... Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57	10 years ago
zuul	120a4999f0	Merge "Add support for building RADIUS with radcli"	10 years ago
Mark Michelson	7a8d6bc81b	Bump ARI version to 2.0.0 ... In order to not have version number overlap between different versions of Asterisk, each new major version of Asterisk will mean we also bump the ARI major version number. This particular change does NOT introduce any known breaking changes to ARI. For discussion relating to this topice, see: http://lists.digium.com/pipermail/asterisk-dev/2016-November/075964.html Change-Id: I712ee0df177a8fe1252da2bc029705268b97b665	10 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	10 years ago
Matt Jordan	a72ef38113	res/ari/resource_bridges: Add the ability to manipulate the video source ... In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	10 years ago
Sebastien Duthil	c6d755de11	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	10 years ago
Tzafrir Cohen	97a75e3829	Add support for building RADIUS with radcli ... Radcli is yet another RADIUS client library, generally compatible with freeradius and radiusclient-ng. This commit adds autoconf option for detecting it as well and changes cdr_radius and cel_radius to use its header file in that case. ASTERISK-26540 #close Change-Id: I271f0715406334874865ffbce0b354b3a2ca148f	10 years ago