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${ noResults }
4375 Commits (aa3da100e8b91f3d15fb485cc753f5738f81899a)
Author | SHA1 | Message | Date |
---|---|---|---|
|
c017d5e6a3 |
Remove the unsafe bridge parameter from ast_bridge_hook_callback's.
Most hook callbacks did not need the bridge parameter. The pointer value could become invalid if the channel is moved to another bridge while it is executing. * Fixed some issues in feature_attended_transfer() as a result. * Reduce the bridge inhibit count in attended_transfer_properties_shutdown() after it has restored the bridge channel hooks. * Removed basic bridge requirement on feature_blind_transfer(). It does not require the basic bridge like feature_attended_transfer(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395574 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
50aba6be36 |
Improved feature limits interval hook implementaion.
* Fixed feature limits to not use special members of struct ast_bridge_features. * Fixed memory leak in off nominal paths of bridge_builtin_set_limits(). * Fixed off nominal path in ast_bridge_features_limits_construct() freeing unallocated memory if it was not called by bridge_builtin_set_limits(). * Made bridge_builtin_interval_features.so unloadable. * Simplified parking's use of its duration interval hook. * Made BridgeWait S option not depend upon another module being loaded. (closes issue ASTERISK-22107) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2701/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395559 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9a46c1d019 |
Add name argument to BridgeWait() so multiple holding bridges may be used
Changes arguments for BridgeWait from BridgeWait(role, options) to BridgeWait(bridge_name, role, options). Now multiple holding bridges may be created and referenced by this application. (closes issue ASTERISK-21922) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2642/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395509 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
eec3150a8a |
Remove some unnecessary parentheses.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
cafc115896 |
A great big renaming patch
This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
9d8a5ceb02 |
Move after bridge callbacks into their own file
One more major refactoring to go. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
07d2694f72 |
Refactor ast_bridge_features struct.
* Reduced the number of hook containers to just dtmf_hooks, interval_hooks, and other_hooks. As a result, several functions dealing with the different hook containers could be combined. * Extended the generic hook struct for DTMF and interval hooks instead of using a variant record. * Merged the special talk detector hook into the other_hooks container. * Replaced ast_bridge_features_set_talk_detector() with ast_bridge_talk_detector_hook(). (issue ASTERISK-22107) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395322 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
1d1650f572 |
Update bridge_channel refactorings; export bridge_ symbol
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395295 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
d91dc6d1a8 |
Perform the initial renaming of the Bridging API
This patch does the following: * It pulls out bridge_channel and puts it into its own translation unit * It adds public and protected headers for bridging_channel. Protected functions are appropriate only for the Bridging API and sub-classes of a bridge. (issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395253 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
02f55a36a0 |
Revert accidental overcommit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393632 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
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 |
|
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 |
|
a3c91e955a |
MixMonitor: Minor code cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |
|
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 |
13 years ago |