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${ noResults }
4390 Commits (c43e19e8e5fc5bc133f2ce3eeafbf9ce07048bd1)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e35860f954 |
Changed some BUGBUG tags to associated JIRA issue tags.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396792 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
82ba10bb47 |
Fix feature_attended_transfer test
The feature_attended_transfer test is failing due to Asterisk not passing DTMF in the bridges created for internal attended transfers. This sets the features initialization routine to set this flag by default and adjusts the basic bridge and confbridge's use of the bridging system accordingly as per Richard's suggestion instead of adjusting this individual case. This change allows the necessary DTMF to pass through the attended transfer bridge and complete the test successfully. Review: https://reviewboard.asterisk.org/r/2759/ (closes issue ASTERISK-22222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396724 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
08991ac281 |
app_bridgewait: Inhibit local channel optimizations to the bridge.
Holding bridges can allow local channel move/swap optimization to the bridge. However, we cannot allow it for the BridgeWait holding bridge because the call will lose the channel roles and dialplan location as a result. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396695 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
6eec8a44e7 |
Update documentation for ConfBridge with some additional markup
Add some additional markup for items that needed it, e.g., replaceable tags, literal tags, etc. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396490 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
3f724fa493 |
Make bridge snapshots use prefixes.
* Changed ast_manager_build_bridge_state_string() to assume an empty prefix string just like ast_manager_build_channel_state_string(). * Created ast_manager_build_bridge_state_string_prefix() to work just like ast_manager_build_channel_state_string_prefix(). * Made BridgeMerge AMI event use To/From prefixes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396417 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
33e7b76d1d |
Hide the Surrogate channels from external consumers; kill Masquerade events
This patch does three things: 1. It provides a Surrogate channel technology with a consolidated "implementation detail flag" on the channel technology. This tells consumers of Stasis that the creation of this channel is an implementation detail in Asterisk and can be ignored (if they so choose). This consolidates the conference recorder/announcer flags as well - these flags had no additional meaning beyond "ignore this channel please". 2. It modifies allocation of a channel in two ways: (a) If a channel technology can be determined from the name, we set it directly in the allocation routine. This prevents the initial publication of the message from going out with a NULL channel technology where possible. This lets Stasis consumers get the right channel technology on the first publication. (b) It reorganizes allocation to make use of the 'finalized' property on the channel. This was already used to know that a channel had completely finished its construction in the masquerade routine; now we also use it to know whether or not the setting of certain channel properties is occurring during or post construction. The various set routines were modified accordingly as well. 3. The masquerade event is now dead, Jim. It no longer served any purpose whatsoever - if you perform a call pickup you'll get a Pickup event; if you perform an attended transfer you will still get those events; if you steal a channel to put it elsewhere you'll get the corresponding NewExten or BridgeEnter events. Review: https://reviewboard.asterisk.org/r/2740 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
200ed6a405 |
Perform Ring-No-Answer checks before processing Hangup logic
The rna() routine will raise a Stasis message involving both the caller and the agent. This doesn't work so well if we already hung up the agent channel, as the channel doesn't quite exist. Not surprisingly, this will crash. This patch properly runs the rna subroutine (performing all of the Ring-No-Answer logic) prior to hanging up the agent channel. (closes issue ASTERISK-22258) Reported by: Kiril Valchev Tested by: Kiril Valchev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396365 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
860ab29dab |
Fixed app_meetme for cache split changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c790848794 |
ARI: Add recording controls
This patch implements the controls from ARI recordings. The controls are: * DELETE /recordings/live/{recordingName} - stop recording and discard it * POST /recordings/live/{recordingName}/stop - stop recording * POST /recordings/live/{recordingName}/pause - pause recording * POST /recordings/live/{recordingName}/unpause - resume recording * POST /recordings/live/{recordingName}/mute - mute recording (record silence to the file) * POST /recordings/live/{recordingName}/unmute - unmute recording. Since this underlying functionality did not already exist, is was added to app.c by a set of control frames, similar to how playback control works. The pause/mute control frames are toggles, even though the ARI controls are idempotent, to be consistent with the playback control frames. (closes issue ASTERISK-22181) Review: https://reviewboard.asterisk.org/r/2697/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396331 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
ccdfe67bf2 |
Check result of ast_var_assign() calls for memory allocation failure.
We try to keep the system running even when all available memory is spent. Review: https://reviewboard.asterisk.org/r/2734/ ........ Merged revisions 396279 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 396287 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
f8622e7c5c |
Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels.
This commit is smaller than the initial review placed on review board. This is because a change to allow for channel drivers to access parking functionality externally was committed and invalidated quite a few of the changes initially made. (closes issue ASTERISK-22039) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2717 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
38236e54a8 |
Remove dead code from features.c; refactor pickup code into pickup.c
This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
c8a91b5b01 |
Add queue member paused hints
This patch adds the ability in Queue to raise a hint when a member's paused state changes. The hint uses the form 'Queue:{queue_name}_pause_{member_name}', where {queue_name} and {member_name} are the name of the queue and the name of the member to subscribe to, respectively. For example: exten => 8501,hint,Queue:sales_pause_mark. Members will show as In Use when paused. Note that the format of the queue pause hint was changed slightly from what is on the issue to accomodate suggestion on the code review. Review: https://reviewboard.asterisk.org/r/2254 (closes issue ASTERISK-20842) Reported by: Philippe Lindheimer patches: qpause-10-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-11-378206.diff uploaded by Philippe Lindheimer (license 5519) qpause-trunk-378206.diff uploaded by Philippe Lindheimer (license 5519) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
03090a88ba |
Fix documentation replication issues
This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
e1b959ccbb |
Split caching out from the stasis_caching_topic.
In working with res_stasis, I discovered a significant limitation to the current structure of stasis_caching_topics: you cannot subscribe to cache updates for a single channel/bridge/endpoint/etc. To address this, this patch splits the cache away from the stasis_caching_topic, making it a first class object. The stasis_cache object is shared amongst individual stasis_caching_topics that are created per channel/endpoint/etc. These are still forwarded to global whatever_all_cached topics, so their use from most of the code does not change. In making these changes, I noticed that we frequently used a similar pattern for bridges, endpoints and channels: single_topic ----------------> all_topic ^ | single_topic_cached ----+----> all_topic_cached | +----> cache This pattern was extracted as the 'Stasis Caching Pattern', defined in stasis_caching_pattern.h. This avoids a lot of duplicate code between the different domain objects. Since the cache is now disassociated from its upstream caching topics, this also necessitated a change to how the 'guaranteed' flag worked for retrieving from a cache. The code for handling the caching guarantee was extracted into a 'stasis_topic_wait' function, which works for any stasis_topic. (closes issue ASTERISK-22002) Review: https://reviewboard.asterisk.org/r/2672/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
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 |
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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 |
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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 |