4365 Commits (a6329a3acf0109d5b364476d212cd0aee4c20dfb)

Author	SHA1	Message	Date
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
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
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
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
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
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
Matthew Jordan	f54b1f2788	Provide error message for QUEUE_MEMBER when member is not in queue ... When QUEUE_MEMBER is used and the member specified is not in the queue, Asterisk provides an ERROR message that indicates that the option specified is not valid. This patch now properly displays an ERROR message that the member is not in the queue if an interface is specified. (closes issue ASTERISK-21980) Reported by: Avraam David ........ Merged revisions 394345 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Russell Bryant	0bfe2d4cc4	astobj2-ify the SLA code ... The SLA code within app_meetme was written before asotbj2 had been merged into Asterisk. Worse, support for reloads did not exist at first and was added later as a bolt-on feature. I knew at the time that reloading was not safe at all while SLA was in use, so the reload would be queued up to execute when the system was idle. Unfortunately, this approach was still prone to errors beyond the fact that this was the only place in Asterisk where configuration was not reloaded instantly when requested. This patch converts various SLA objects to be reference counted objects using astobj2. This allows reloads to be processed while the system is in use. The code ensures that the objects will not disappear while one of the other threads is using them. However, they will be immediately removed from the global trunk and station containers so no new calls will use them if removed from configuration. Review: https://reviewboard.asterisk.org/r/2581/ ........ Merged revisions 393928 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 393929 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	02f55a36a0	Revert accidental overcommit. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b4e9a3fc2f	Add BUGBUG note for ASTERISK-22009 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	a75fd32212	ARI - channel recording support ... This patch is the first step in adding recording support to the Asterisk REST Interface. Recordings are stored in /var/spool/recording. Since recordings may be destructive (overwriting existing files), the API rejects attempts to escape the recording directory (avoiding issues if someone attempts to record to ../../lib/sounds/greeting, for example). (closes issue ASTERISK-21594) (closes issue ASTERISK-21581) Review: https://reviewboard.asterisk.org/r/2612/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a3c91e955a	MixMonitor: Minor code cleanup. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393500 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	e4a5ae0376	MixMonitor: Make start_mixmonitor_callback() options parameter NULL tolerant. ... * Removed some unnecessary code in start_mixmonitor_callback(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393496 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	f6cdd9a12c	MixMonitor: Don't use ast_strdupa() in a loop. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b47152121c	MixMonitor: Update XML documentation and CLI "mixmonitor {start|stop|list}" help. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	8ab7724352	MixMonitor: Fix refleak in manager_stop_mixmonitor() if could not stop monitoring. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6a65c4d072	MixMonitor: Remove some unnecessary channel locking. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	631fad018f	Fix MixMonitor b option. ... The option had not been converted to use the replacement for ast_bridged_channel(). One touch mixmonitor now records files again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f306dbd841	bridge_features: Support One touch Monitor/MixMonitor ... In addition to porting those features, they now enjoy greater feature parity with one another. Specifically, AutoMixMon now has a start and stop message that can be specified with TOUCH_MIXMONITOR_MESSAGE_START and TOUCH_MIXMONITOR_MESSAGE_STOP. (closes issue ASTERISK-21553) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2620/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	909ee4bfb9	Refactor extraneous channel events ... This change removes JitterBufStats, ChannelReload, and ChannelUpdate and refactors the following events to travel over Stasis-Core: * LocalBridge * DAHDIChannel * AlarmClear * SpanAlarmClear * Alarm * SpanAlarm * DNDState * MCID * SIPQualifyPeerDone * SessionTimeout Review: https://reviewboard.asterisk.org/r/2627/ (closes issue ASTERISK-21476) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a022379107	Fix incorrect calls to ast_bridge_impart(). ... There was a misunderstanding about ast_bridge_impart()'s handling of the imparted channel's reference. The channel reference is passed by the caller unless ast_bridge_impart() returns an error. * Fixed a memory leak in conf_announce_channel_push() if the impart failed. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a1e219ef51	CEL refactoring cleanup ... This change removes AST_CEL_BRIDGE_UPDATE since it should no longer be used because masquerade situations are now accounted for in other ways. This also refactors usage of AST_CEL_FORWARD to be produced by a Dial message which has been extended with a "forward" field. (closes issue ASTERISK-21566) Review: https://reviewboard.asterisk.org/r/2635/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392829 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	b5f18d1677	Fix menuselect display for stasis modules. ... The menuselect parser is very simple. It looks for AST_MODULE_INFO and uses any quoted string on that line as the module summary display. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392777 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	cd40e179a9	Fix potential bridge hook resource leak if the hook install fails. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework ... This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bfdff342b4	app_mixmonitor: Fix crashes caused by unloading app_mixmonitor ... Unloading app_mixmonitor while active mixmonitors were running would cause a segfault. This patch fixes that by making it impossible to unload app_mixmonitor while mixmonitors are active. Review: https://reviewboard.asterisk.org/r/2624/ ........ Merged revisions 391778 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391794 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0e2a9d07ac	app_confbridge: Fix memory leak on reload. ... The config framework options should not be registered multiple times. Instead the configuration just needs to be reprocessed by the config framework. ........ Merged revisions 391700 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c2e29abcbf	Add announce-to-first-user option for app_queue ... In r386792, the ability to play prompts to the first caller in a call queue was added. While this is arguably a bug fix for those who expect the first caller to continue receiving prompts while the agent is dialed, it has the side effect of preventing the first caller from hearing the agent immediately upon bridging. This may not be a problem for those who really want this option, but for those who didn't care whether or not the first caller in queue heard their position, it was an issue. This patch disables the ability for the first caller in the queue to hear prompts and adds a new option, announce-to-first-user, to queues.conf. Those who the behavior can enable it by setting this value to True. Note that if we ever implement the ability to have the prompts be stopped upon bridging, this option can be removed. (closes issue ASTERISK-21782) Reported by: Remi Quezada ........ Merged revisions 391215 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 391241 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391245 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	a2d02edca5	Make app_queue AMI events more consistent. Give Join/Leave more useful names. ... This also removes the eventwhencalled and eventmemberstatus configuration options. These events can just be filtered via manager.conf blacklists. (closes issue ASTERISK-21469) Review: https://reviewboard.asterisk.org/r/2586/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	bec2d79484	app_meetme: Refactor manager events to use stasis ... (closes issue ASTERISK-21467) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2564/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	2dc8a06006	Refactor the features configuration scheme. ... Features configuration is handled in its own API in features_config.h and features_config.c. This way, features configuration is accessible to anything that needs it. In addition, features configuration has been altered to be more channel-oriented. Most callers of features API code will be supplying a channel so that the individual channel's settings will be acquired rather than the global setting. Missing from this commit is XML documentation for the features configuration. That will be handled in a separate commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5c554dc470	* Fix a couple missed hook installs that need AST_BRIDGE_HOOK_REMOVE_ON_PULL. ... * Rename some hook flag parameters to remove_flags. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	1458a20e47	Refactor code and fix a reference leak ... Refactor some channel blob publishing code to use ast_channel_publish_blob now that it is available and fix a JSON reference leak that was occurring during varset publishing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	39d5e40cd5	Remove remnant of snapshot blob JSON types ... Remove usage of the once-mandatory snapshot blob type field, refactor confbridge stasis messages accordingly, and remove ast_bridge_blob_json_type(). Review: https://reviewboard.asterisk.org/r/2575/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390250 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6851801a5e	Resolve a merge conflict ... When ast_channel_cached_blob_create was merged, ast_channel_blob_create_from_cache was partially removed in an unresolved merge conflict. This restores ast_channel_blob_create_from_cache and refactors usage of ast_channel_cached_blob_create (requires an ast_channel) to use ast_channel_blob_create_from_cache (requires a channel uniqueid) instead. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389974 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	fac3839e68	Adds support for a core attended transfer function plus adds some hiding of masquerades. ... The attended transfer API call can complete the attended transfer in a number of ways depending on the current bridged states of the channels involved. The hiding of masquerades is done in some bridging-related functions, such as the manager Bridge action and the Bridge dialplan application. In addition, call pickup was edited to "move" a channel rather than masquerade it. Review: https://reviewboard.asterisk.org/r/2511 (closes issue ASTERISK-21334) Reported by Matt Jordan (closes issue Asterisk-21336) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	06be8463b6	Migrate a large number of AMI events over to Stasis-Core ... This patch moves a number of AMI events over to the Stasis-Core message bus. This includes: * ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload * All Voicemail/MWI related events In addition, it adds some Stasis-Core and AMI support for generic AMI messages, refactors the message router in AMI to use a single router with topic forwarding for the topics that AMI cares about, and refactors MWI message types and topics to be more name compliant. Review: https://reviewboard.asterisk.org/r/2532 (closes issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	b6aac885be	Add dial events to app_queue and app_followme. ... Also fixes an issue in app_dial, where the channels were swapped on dial events. (closes issue ASTERISK-21551) (closes issue ASTERISK-21550) Review: https://reviewboard.asterisk.org/r/2549/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389492 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	908ac3507a	Conditional out more app_queue logging that needs to be reworked. ... Fixes crash because app_queue was unconditionally freeing a datastore that was still on a channel. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	afb1d96068	Raise the ConfBridgeMute/Unmute events when a CLI or AMI action triggers the change ... New in 12 are the ConfBridgeMute/Unmute events, which are triggered when a user changes their mute/unmute state. This was typically triggered when a user hit a DTMF key that triggered the mute/unmute menu handler. Forgotten in this is when an AMI action or CLI command triggers the mute/unmute. This patch now raises the events in those situations as well. (closes issue ASTERISK-21802) Reported by: Birger "WIMPy" Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1e1cc2dee	Fixed some extra field assertion when the event WebSocket is connected ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	eb06c505f4	Add documentation for record_file_append ... When this option was added, it was noted in CHANGES, but was missing the XML documentation that this patch adds. (closes issue ASTERISK-21780) Patch-by: Brad Latus (snuffy) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389202 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b97c71bb11	Fix shutdown assertions in stasis-core ... In r388005, macros were introduced to consistently define message types. This added an assert if a message type was used either before it was initialized or after it had been cleaned up. It turns out that this assertion fires during shutdown. This actually exposed a hidden shutdown ordering problem. Since unsubscribing is asynchronous, it's possible that the message types used by the subscription could be freed before the final message of the subscription was processed. This patch adds stasis_subscription_join(), which blocks until the last message has been processed by the subscription. Since joining was most commonly done right after an unsubscribe, a stasis_unsubscribe_and_join() convenience function was also added. Similar functions were also added to the stasis_caching_topic and stasis_message_router, since they wrap subscriptions and have similar problems. Other code in trunk was refactored to join() where appropriate, or at least verify that the subscription was complete before being destroyed. Review: https://reviewboard.asterisk.org/r/2540 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d04f1fd60a	Publish the outbound channel's application/data when dialing ... This patch does two things: * It fixes a bug where the outbound channel's application/data set by the dialing API/app_dial is not communicated until the channel is hung up. If that happens, AMI would incorrectly send a NewExten event immediately after a Hangup. This isn't really AMI's fault, as the dialing APIs never communicated the 'helpful' app/data on the outbound channel until it was hungup. * It makes public sending a stasis message about a change in channel state. This is useful enough that - for now at least - it should be public. If operations on a channel go to being more coarse-grained, this function could be made private again. Review: https://reviewboard.asterisk.org/r/2548 Note that this problem was found and reported by Matt DiMeo. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	d8d1def22c	Fix VM snapshot handling for combined INBOX. ... The snapshot API contains an option that allow for combining of new and old messages within a single snapshot. New messages, however, include options beyond just 'INBOX' - it also includes the Urgent folder. A previous patch that combined INBOX and Urgent accidentally impacted snapshots that attempted to gain messages from just the Old folder. This patch fixes the snapshot gathering such that the API returns the appropriate messages for the folder selected, with and without the combine option. This should make it more clear about what's happening. Review: https://reviewboard.asterisk.org/r/2539/ ........ Merged revisions 388816 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388818 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e8f4ac6c61	Break res_stasis into smaller files. ... When implementing playback for stasis-http, the monolithicedness of res_stasis really started to get in my way. This patch breaks the major components of res_stasis.c into individual files. * res/stasis/app.c - Stasis application tracking * res/stasis/control.c - Channel control objects * res/stasis/command.c - Channel command object This refactoring also allows res_stasis applications to be loaded as independent modules, such as the new res_stasis_answer module. The bulk of this patch is simply moving code from one file to another, adjusting names and adding accessors as necessary. Review: https://reviewboard.asterisk.org/r/2530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4666079b05	Address unload order issues for res_stasis* modules ... I've noticed when doing a graceful shutdown that the res_stasis_http.so module gets unloaded before the modules that use it, which causes some asserts during their unload. While r386928 was a quick hack to get it to not assert and die, this patch increases the use counts on res_stasis.so and res_stasis_http.so properly. It's a bigger change than I expected, hence the review instead of just committing it. Review: https://reviewboard.asterisk.org/r/2489/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	7ce05bfb9b	Add channel events for res_stasis apps ... This change adds a framework in res_stasis for handling events from channel topics. JSON event generation and validation code is created from event documentation in rest-api/api-docs/events.json to assist in JSON event generation, ensure consistency, and ensure that accurate documentation is available for ALL events that are received by res_stasis applications. The userevent application has been refactored along with the code that handles userevent channel blob events to pass the headers as key/value pairs in the JSON blob. As a side-effect, app_userevent now handles duplicate keys by overwriting the previous value. Review: https://reviewboard.asterisk.org/r/2428/ (closes issue ASTERISK-21180) Patch-By: Kinsey Moore <kmoore@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388275 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c81ff6102f	Don't expect to pack three tuples when you only have two ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago