asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	16885ffda5	Expose the chan_pjsip implementation pvt and session in a defined manner. This allows modules outside of chan_pjsip itself to get the session given only an Asterisk channel. Review: https://reviewboard.asterisk.org/r/2674/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395102 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	b4c2eecca6	Fix unbalanced lock when serializing CDR variables I'm only surprised that this didn't cause larger problems. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0ac2c093e2	Remove some BUGBUG notes that have been handled. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	8aac17a3ee	Make the CEL blind transfer test pass consistently git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	715d894d48	Update copyright year to 2013 in asterisk.c; some whitespace fixes (closes issue ASTERISK-22179) Reported by: Malcolm Davenport ........ Merged revisions 395032 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 395033 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395034 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e6fb7c9764	Blocked revisions 395020 ........ Add an upgrade note for libuuid dependency; remove note in CHANGES This patch notes that libuuid is now a dependency for res_rtp_asterisk; this was introduced in between 11.4.0 and 11.5.0 to resolve a dependency for pjproject, which res_rtp_asterisk uses for ICE/STUN/TURN support. It also removes a conflicting note from CHANGES. While support for playing prompts to the first participant was added for app_queue, it was disabled by default and an option added to enable it. That was properly noted in the UPGRADE.txt file. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	88e344d2df	Clean up documentation This patch cleans up documentation in func_channel for the following items: * rtpsource * secure_signaling * secure_media * various OOH323 parameters (closes issue ASTERISK-20969) Reported by: snuffy patches: func_chan-update.diff uploaded by snuffy (License 5024) ........ Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394981 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bb955e37fb	Provide proper ring tone in indications.conf for Malaysia The ring tone provided in the sample indications.conf was incorrect. This patch modifies the sample ring tone to be what it should: ring = 425/400,0/200,425/400,0/2000 This brings it in line with the tone definition in DAHDI 2.7.0. (zonedata.c) (closes issue ASTERISK-21997) Reported by: Filip Jenicek patches: malaysia_ring.patch uploaded by phill (License 6277) ........ Merged revisions 394940 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394941 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	54803338b4	Always install safe_asterisk; add configuration file support This patch modifies the behavior of safe_asterisk in two ways: (1) It modifies the Asterisk Makefile such that safe_asterisk is always installed on a 'make install'. This was done as bugfixes in the safe_asterisk script were not applied in previous version of Asterisk without first removing the old version of the script. (2) In order to keep a newly installed version of safe_asterisk from impacting local modifications, a new config file - safe_asterisk.conf.sample - has been provided. Settings that were previously modified in safe_asterisk can be set there instead. (closes issue ASTERISK-21965) Reported by: Jeremy Kister patches: safe_asterisk.patch uploaded by jkister (License 6232) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bdb1c6bfb0	Tolerate presence of RFC2965 Cookie2 header by ignoring it This patch modifies parsing of cookies in Asterisk's http server by doing an explicit comparison of the "Cookie" header instead of looking at the first 6 characters to determine if the header is a cookie header. This avoids parsing "Cookie2" headers and overwriting the previously parsed "Cookie" header. Note that we probably should be appending the cookies in each "Cookie" header to the parsed results; however, while clients can send multiple cookie headers they never really do. While this patch doesn't improve Asterisk's behavior in that regard, it shouldn't make it any worse either. Note that the solution in this patch was pointed out on the issue by the issue reporter, Stuart Henderson. (closes issue ASTERISK-21789) Reported by: Stuart Henderson Tested by: mjordan, Stuart Henderson ........ Merged revisions 394899 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394900 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	44fe8fcb91	Update PostgreSQL realtime scripts with schema for queue_log table This patch updates the realtime SQL scripts with an entry that will create the queue_log table. This brings the PostgreSQL scripts inline with the MySQL scripts, with respect to what tables they will create. (closes issue ASTERISK-21021) Reported by: Eugene patches: queue_log.sql uploaded by varnav (license 6360) ........ Merged revisions 394896 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394897 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394898 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e502d0095b	Add additional control frame types to the IAX2 parser for debug messages This patch adds some of the more recent control frame types to the IAX2 parser. When IAX2 debugging is enabled, it will now show more of the control frame types. (closes issue ASTERISK-22120) Reported by: Birger "WIMPy" Harzenetter patches: iaxcmds.diff uploaded by wimpy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	75e83bdbab	Document connectedline parameter for chan_iax2 The connectedline parameter for a chan_iax2 peer was undocumented. This patch documents the options in the sample configuration file. (closes issue ASTERISK-21953) Reported by: Birger "WIMPy" Harzenetter ........ Merged revisions 394886 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394890 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3a5b68f07c	Allow setting allowmultiplelogin on an account basis This patch modifies manager to allow the allowmultiplelogin setting to be set on an account by account basis. When set in the general context, it will act as the default for the defined accounts. Setting it in the account will override the general setting. (closes issue ASTERISK-21324) Reported by: vldmr patches: asterisk-manager-per-user-allowmultiplelogin.patch uploaded by vldmr (License 6487) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394881 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	c3b8939be8	Add CEL local optimization record type This adds a new CEL event type, AST_CEL_LOCAL_OPTIMIZE, to represent local channel optimizations. Local channel optimizations were one of several things conveyed by the now defunct BRIDGE_UPDATE event type. This also adds a unit test to test generation of this new CEL event. Review: https://reviewboard.asterisk.org/r/2676/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	684c83b29b	Add transfer support to CEL This adds CEL support for blind and attended transfers and call pickup. During the course of adding this functionality I noticed that CONF_ENTER, CONF_EXIT, and BRIDGE_TO_CONF events are particularly useless without a bridge identifier, so I added that as well. This adds tests for blind transfers, several types of attended transfers, and call pickup. The extra field in CEL records now consists of a JSON blob whose fields are defined on a per-event basis. Review: https://reviewboard.asterisk.org/r/2658/ (closes issue ASTERISK-21565) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394858 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4e15046cf6	Regroup the ao2 search_flags. Moved the OBJ_POINTER, OBJ_KEY, and OBJ_PARTIAL_KEY flags together into a field and renamed them to OBJ_SEARCH_OBJECT, OBJ_SEARCH_KEY, and OBJ_SEARCH_PARTIAL_KEY respectively. The values were selected to keep existing code compiling and working until the codebase can be changed to stop using these values as bit flags and use them as an enum field. The old names are defined to the new names for backward compatibility. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	643fb1ed14	Minor optimizations. * Made ast_audiohook_detach_list() and ast_audiohook_write_list_empty() NULL tolerant. * Made ast_audiohook_detach_list() return void since it is a destructor. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2838683742	Extract a repeated test into ast_channel_has_audio_frame_or_monitor(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a6329a3acf	ARI: MOH start and stop for a channel (issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2680/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	17c546173f	ARI: Bridge Playback, Bridge Record Adds a new channel driver for creating channels for specific purposes in bridges, primarily to act as either recorders or announcers. Adds ARI commands for playing announcements to ever participant in a bridge as well as for recording a bridge. This patch also includes some documentation/reponse fixes to related ARI models such as playback controls. (closes issue ASTERISK-21592) Reported by: Matt Jordan (closes issue ASTERISK-21593) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2670/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	5a8f32703c	Filter channels used as internal mechanisms This adds new flags to the channel tech properties that flag it as different types of implementation detail used exclusively to provide a feature. Examples of channels that would have these flags include the announcement and recording channels used by confbridge which are the only two marked as such by this patch. Review: https://reviewboard.asterisk.org/r/2633/ (closes issue ASTERISK-21873) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	0b83761f9a	Fix crash when using temporary peers Temporary peers do not have an associated Stasis endpoint and quite a bit of code in chan_sip assumes that all peers have a Stasis endpoint. All endpoint accesses in chan_sip are now wrapped in an endpoint NULL-check. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	900714beb9	Convert CCSS manager events to stasis. (closes issue ASTERISK-21473) Review: https://reviewboard.asterisk.org/r/2682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	345f033a19	Made audiohooks, framehooks, and monitor prevent local channel optimization. Audiohooks, framehooks, and monitor represent state on a local channel that will go away if it is optimized out. (closes issue ASTERISK-21954) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2685/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e5d3deb299	Fixup doxygen on ast_hangup(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394776 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c47787feab	Add a bunch of options from sip.conf to res_sip.conf For a complete list of the options added, see the review linked at the bottom of this commit message. (closes issue ASTERISK-21506) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2671 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394759 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	3c86832f9f	Fixed null dereference when WebSocket subprotocol isn't specified git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394744 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	81f36bee0f	bridge_holding/app_bridgewait: Add new entertainment options This patch adds more entertainment options to holding bridges and the bridge_wait application. Also, holding bridges will now use music on hold as the default entertainment option instead of none. The parameters for app_bridgewait have changed to (role, options) from the previous (options) and the options themselves have changed as well (entertainment options are now contained in an enumerator, role specification is handled by the role parameter, etc) (closes issue ASTERISK-21923) Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/2679/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394731 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	c1a7567d24	ARI: Add support for suppressing media streams. Also convert res_mutestream to use the core feature behind this. (closes issue ASTERISK-21618) Review: https://reviewboard.asterisk.org/r/2652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3a2a12ca1a	Tweak debug statements This patch does two things: 1. It moves the debug statement that shows the HTTP sub-protocols being compared after the string length calculation such that it shows the correct string length in the output 2. It adds some additional debug that displays when it matches on a sub-protocol and when it fails git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	de68dabb97	Fix caching topic shutdown assertions The recent changes to update stasis_cache_topics directly from the publisher thread uncovered a race condition, which was causing asserts in the /stasis/core tests. If the caching topic's subscription is the last reference to the caching topic, it will destroy the caching topic after the final message has been processed. When dispatching to a different thread, this usually gave the unsubscribe enough time to finish before destruction happened. Now, however, it consistently destroys before unsubscription is complete. This patch adds an extra reference to the caching topic, to hold it for the duration of the unsubscription. This patch also removes an extra unref that was happening when the final message was received by the caching topic. It was put there because of an extra ref that was put into the caching topic's constructor. Both have been removed, which makes the destructor a bit less confusing. Review: https://reviewboard.asterisk.org/r/2675/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394686 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	6d9909887e	Properly indicate failure to open an audio stream in res_agi If there is an error streaming an audio file, the current return status makes it difficult for an AGI script to determine that there was an error with the audio file. This patches changes the result to return -1 and the function returns RESULT_FAILURE instead of RESULT_SUCCESS. From looking at other parts of res_agi, this would appear to be the proper way to handle an error. (closes issue ASTERISK-21903) Reported by: Ariel Wainer Tested by: Ariel Wainer Patches: asterisk-21903-return-stream-res_1.8.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2625/ ........ Merged revisions 394640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394641 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	40ce5e0d18	Change ast_hangup() to return void and be NULL safe. Since ast_hangup() is effectively a channel destructor, it should be a void function. * Make the few silly callers checking the return value no longer do so. Only the CDR and CEL unit tests checked the return value. * Make all callers take advantage of the NULL safe change and remove the NULL check before the call. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394623 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	da1902cdc0	Remove some completed and no longer relevant BUGBUG notes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	4f29b97020	app_confbridge: Eliminate a reference leak for confbridge announcer channels git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	109f527eb9	Left over spacing issues of review 726. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	b10bb8fa7b	handle DAHDI_EVENT_REMOVED on a pri D-Channel When a DAHDI device is removed at run-time it sends the event DAHDI_EVENT_REMOVED on each channel. This is intended to signal the userspace program to close the respective file handle, as the driver of the device will need all of them closed to properly clean-up. This event has long since been handled in chan_dahdi (chan_zap at the time). However the event that is sent on a D-Channel of a "PRI" (ISDN) span simply gets ignored. This commit adds handling for closing the file descriptor (and shutting down the span, while we're at it). It also adds a CLI command 'pri destroy span <N>' to destroy the span and its DAHDI channels. Review: https://reviewboard.asterisk.org/r/726/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	19d8f8c8e4	Add 'kick all' capability to ConfBridge CLI command This patch adds the ability to kick all users out of a conference from the ConfBridge kick CLI command. It is invoked by passing 'all' as the channel parameter to the CLI command, i.e., "confbridge kick <conf> all". Note that this patch was modified slightly to conform to trunk. (closes issue ASTERISK-21827) Reported by: dorianlogan patches: kickall-patch_v2.diff uploaded by dorianlogan (License 6504) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394531 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1623347817	Re-order handlers in CEL to ensure that HANGUP events happen after APP_END When a channel is hungup, both an APP_END event and a HANGUP event can be fired. To ensure that HANGUP events occur after APP_END events, the method callbacks for the APP_END event should be processed prior to the callbacks for the HANGUP event. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1b75afdb8	Debug logging to help with WebSocket connection problems git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3f9be54d91	chan_gulp: Fix gulp_indicate() handling of AST_CONTROL_PVT_CAUSE_CODE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	6bdd453168	Prevent crash from trying to end a session in an invalid way. This ensures that code that was only meant to be run on a reinvite failure only runs on a reinvite failure. (closes issue ASTERISK-22061) reported by Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394473 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6ba25dd3f2	Remove some dead code dealing with old bridging method. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9732112261	Simplify bridge_simple chan join code. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394470 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	957f863dba	Re-order cleanup This patch attempts to fix some possible race conditions in shutdown of the CDR engine. It: * Adds a cleanup handler to only unsubscribe and join on stasis messages during graceful shutdown. The cleanup handler should execute before the regular atexit handler, as we want to unsubscribe for any further messages before dispatching the CDRs. * The CDRs are now locked when we dispatch them on shutdown. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	80dd0229f1	Fixed null dereference when WebSocket protocol is omitted git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d43b17a872	Replace chan_agent with app_agent_pool. The ill conceived chan_agent is no more. It is now replaced by app_agent_pool. Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) Sample extensions.conf [login] ; Sample agent 1001 login ; Set COLP for in between calls so the agent does not see the last caller COLP. exten => 1001,1,Set(CONNECTEDLINE(all)="Agent Waiting" <1001>) ; Give the agent DTMF transfer and disconnect features when connected to a caller. same => n,Set(CHANNEL(dtmf-features)=TX) same => n,AgentLogin(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() [caller] ; Sample caller direct connect to agent 1001 exten => 800,1,AgentRequest(1001) same => n,NoOp(AGENT_STATUS is ${AGENT_STATUS}) same => n,Hangup() ; Sample caller going through a Queue to agent 1001 exten => 900,1,Queue(agent_q) same => n,Hangup() Sample queues.conf [agent_q] member => Local/800@caller,,SuperAgent,Agent:1001 Under the hood operation overview: 1) Logged in agents wait for callers in an agents holding bridge. 2) Caller requests an agent using AgentRequest() 3) A basic bridge is created, the agent is notified, and caller joins the basic bridge to wait for the agent. 4) The agent is either automatically connected to the caller or must ack the call to connect. 5) The agent is moved from the agents holding bridge to the basic bridge. 6) The agent and caller talk. 7) The connection is ended by either party. 8) The agent goes back to the agents holding bridge. To avoid some locking issues with the agent holding bridge, I needed to make some changes to the after bridge callback support. The after bridge callback is now a list of requested callbacks with the last to be added the only active callback. The after bridge callback for failed callbacks will always happen in the channel thread when the channel leaves the bridging system or is destroyed. (closes issue ASTERISK-21554) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2657/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d43bbfb74e	Remove misleading documentation for channel snapshot creation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	de31a362db	Document the ari.conf allowed_origins setting git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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